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diff --git a/1.2-netsec/doc/CODING-GUIDELINES b/1.2-netsec/doc/CODING-GUIDELINES deleted file mode 100644 index c3c62bfdd..000000000 --- a/1.2-netsec/doc/CODING-GUIDELINES +++ /dev/null @@ -1,510 +0,0 @@ -== Asterisk Patch/Coding Guidelines == --------------------------------------- - -We are looking forward to your contributions to Asterisk - the -Open Source PBX! As Asterisk is a large and in some parts very -time-sensitive application, the code base needs to conform to -a common set of coding rules so that many developers can enhance -and maintain the code. Code also needs to be reviewed and tested -so that it works and follows the general architecture and guide- -lines, and is well documented. - -Asterisk is published under a dual-licensing scheme by Digium. -To be accepted into the codebase, all non-trivial changes must be -disclaimed to Digium or placed in the public domain. For more information -see http://bugs.digium.com - -Patches should be in the form of a unified (-u) diff, made from the directory -above the top-level Asterisk source directory. For example: - -- the base code you are working from is in ~/work/asterisk-base -- the changes are in ~/work/asterisk-new - -~/work$ diff -urN asterisk-base asterisk-new - -If you are changing within a fresh SVN working copy, you can create -a patch with -$ svn diff <mycodefile>.c - -* General rules ---------------- - -- All code, filenames, function names and comments must be in ENGLISH. - -- Don't annotate your changes with comments like "/* JMG 4/20/04 */"; - Comments should explain what the code does, not when something was changed - or who changed it. If you have done a larger contribution, make sure - that you are added to the CREDITS file. - -- Don't make unnecessary whitespace changes throughout the code. - If you make changes, submit them to the tracker as separate patches - that only include whitespace and formatting changes. - -- Don't use C++ type (//) comments. - -- Try to match the existing formatting of the file you are working on. - -- Use spaces instead of tabs when aligning in-line comments or #defines (this makes - your comments aligned even if the code is viewed with another tabsize) - -* Declaration of functions and variables ----------------------------------------- - -- Do not declare variables mid-block (e.g. like recent GNU compilers support) - since it is harder to read and not portable to GCC 2.95 and others. - -- Functions and variables that are not intended to be used outside the module - must be declared static. - -- When reading integer numeric input with scanf (or variants), do _NOT_ use '%i' - unless you specifically want to allow non-base-10 input; '%d' is always a better - choice, since it will not silently turn numbers with leading zeros into base-8. - -- Strings that are coming from input should not be used as a first argument to - a formatted *printf function. - -* Use the internal API ----------------------- - -- Make sure you are aware of the string and data handling functions that exist - within Asterisk to enhance portability and in some cases to produce more - secure and thread-safe code. Check utils.c/utils.h for these. - - -* Code formatting ------------------ - -Roughly, Asterisk code formatting guidelines are generally equivalent to the -following: - -# indent -i4 -ts4 -br -brs -cdw -cli0 -ce -nbfda -npcs -nprs -npsl -saf -sai -saw foo.c - -this means in verbose: - -i4: indent level 4 - -ts4: tab size 4 - -br: braces on if line - -brs: braces on struct decl line - -cdw: cuddle do while - -cli0: case indentation 0 - -ce: cuddle else - -nbfda: dont break function decl args - -npcs: no space after function call names - -nprs: no space after parentheses - -npsl: dont break procedure type - -saf: space after for - -sai: space after if - -saw: space after while - -Function calls and arguments should be spaced in a consistent way across -the codebase. - GOOD: foo(arg1, arg2); - GOOD: foo(arg1,arg2); /* Acceptable but not preferred */ - BAD: foo (arg1, arg2); - BAD: foo( arg1, arg2 ); - BAD: foo(arg1, arg2,arg3); - -Don't treat keywords (if, while, do, return) as if they were functions; -leave space between the keyword and the expression used (if any). For 'return', -don't even put parentheses around the expression, since they are not -required. - -There is no shortage of whitespace characters :-) Use them when they make -the code easier to read. For example: - - for (str=foo;str;str=str->next) - -is harder to read than - - for (str = foo; str; str = str->next) - -Following are examples of how code should be formatted. - -- Functions: -int foo(int a, char *s) -{ - return 0; -} - -- If statements: -if (foo) { - bar(); -} else { - blah(); -} - -- Case statements: -switch (foo) { -case BAR: - blah(); - break; -case OTHER: - other(); - break; -} - -- No nested statements without braces, e.g.: - -for (x = 0; x < 5; x++) - if (foo) - if (bar) - baz(); - -instead do: -for (x = 0; x < 5; x++) { - if (foo) { - if (bar) - baz(); - } -} - -- Don't build code like this: - -if (foo) { - /* .... 50 lines of code ... */ -} else { - result = 0; - return; -} - -Instead, try to minimize the number of lines of code that need to be -indented, by only indenting the shortest case of the 'if' -statement, like so: - -if !(foo) { - result = 0; - return; -} - -.... 50 lines of code .... - -When this technique is used properly, it makes functions much easier to read -and follow, especially those with more than one or two 'setup' operations -that must succeed for the rest of the function to be able to execute. - -- Labels/goto are acceptable -Proper use of this technique may occasionally result in the need for a -label/goto combination so that error/failure conditions can exit the -function while still performing proper cleanup. This is not a bad thing! -Use of goto in this situation is encouraged, since it removes the need -for excess code indenting without requiring duplication of cleanup code. - -- Never use an uninitialized variable -Make sure you never use an uninitialized variable. The compiler will -usually warn you if you do so. However, do not go too far the other way, -and needlessly initialize variables that do not require it. If the first -time you use a variable in a function is to store a value there, then -initializing it at declaration is pointless, and will generate extra -object code and data in the resulting binary with no purpose. When in doubt, -trust the compiler to tell you when you need to initialize a variable; -if it does not warn you, initialization is not needed. - -- Do not cast 'void *' -Do not explicitly cast 'void *' into any other type, nor should you cast any -other type into 'void *'. Implicit casts to/from 'void *' are explicitly -allowed by the C specification. This means the results of malloc(), calloc(), -alloca(), and similar functions do not _ever_ need to be cast to a specific -type, and when you are passing a pointer to (for example) a callback function -that accepts a 'void *' you do not need to cast into that type. - -* Variable naming ------------------ - -- Global variables -Name global variables (or local variables when you have a lot of them or -are in a long function) something that will make sense to aliens who -find your code in 100 years. All variable names should be in lower -case, except when following external APIs or specifications that normally -use upper- or mixed-case variable names; in that situation, it is -preferable to follow the external API/specification for ease of -understanding. - -Make some indication in the name of global variables which represent -options that they are in fact intended to be global. - e.g.: static char global_something[80] - -- Don't use un-necessary typedef's -Don't use 'typedef' just to shorten the amount of typing; there is no substantial -benefit in this: - -struct foo { - int bar; -}; -typedef foo_t struct foo; - -In fact, don't use 'variable type' suffixes at all; it's much preferable to -just type 'struct foo' rather than 'foo_s'. - -- Use enums instead of #define where possible -Use enums rather than long lists of #define-d numeric constants when possible; -this allows structure members, local variables and function arguments to -be declared as using the enum's type. For example: - -enum option { - OPT_FOO = 1 - OPT_BAR = 2 - OPT_BAZ = 4 -}; - -static enum option global_option; - -static handle_option(const enum option opt) -{ - ... -} - -Note: The compiler will _not_ force you to pass an entry from the enum -as an argument to this function; this recommendation serves only to make -the code clearer and somewhat self-documenting. In addition, when using -switch/case blocks that switch on enum values, the compiler will warn -you if you forget to handle one or more of the enum values, which can be -handy. - -* String handling ------------------ - -Don't use strncpy for copying whole strings; it does not guarantee that the -output buffer will be null-terminated. Use ast_copy_string instead, which -is also slightly more efficient (and allows passing the actual buffer -size, which makes the code clearer). - -Don't use ast_copy_string (or any length-limited copy function) for copying -fixed (known at compile time) strings into buffers, if the buffer is something -that has been allocated in the function doing the copying. In that case, you -know at the time you are writing the code whether the buffer is large enough -for the fixed string or not, and if it's not, your code won't work anyway! -Use strcpy() for this operation, or directly set the first two characters -of the buffer if you are just trying to store a one-character string in the -buffer. If you are trying to 'empty' the buffer, just store a single -NULL character ('\0') in the first byte of the buffer; nothing else is -needed, and any other method is wasteful. - -In addition, if the previous operations in the function have already -determined that the buffer in use is adequately sized to hold the string -you wish to put into it (even if you did not allocate the buffer yourself), -use a direct strcpy(), as it can be inlined and optimized to simple -processor operations, unlike ast_copy_string(). - -* Use of functions ------------------- - -When making applications, always ast_strdupa(data) to a local pointer if -you intend to parse the incoming data string. - - if (data) - mydata = ast_strdupa(data); - - -- Separating arguments to dialplan applications and functions -Use ast_app_separate_args() to separate the arguments to your application -once you have made a local copy of the string. - -- Parsing strings with strsep -Use strsep() for parsing strings when possible; there is no worry about -'re-entrancy' as with strtok(), and even though it modifies the original -string (which the man page warns about), in many cases that is exactly -what you want! - -- Create generic code! -If you do the same or a similar operation more than one time, make it a -function or macro. - -Make sure you are not duplicating any functionality already found in an -API call somewhere. If you are duplicating functionality found in -another static function, consider the value of creating a new API call -which can be shared. - -* Handling of pointers and allocations --------------------------------------- - -- Dereference or localize pointers -Always dereference or localize pointers to things that are not yours like -channel members in a channel that is not associated with the current -thread and for which you do not have a lock. - channame = ast_strdupa(otherchan->name); - -- Use const on pointer arguments if possible -Use const on pointer arguments which your function will not be modifying, as this -allows the compiler to make certain optimizations. In general, use 'const' -on any argument that you have no direct intention of modifying, as it can -catch logic/typing errors in your code when you use the argument variable -in a way that you did not intend. - -- Do not create your own linked list code - reuse! -As a common example of this point, make an effort to use the lockable -linked-list macros found in include/asterisk/linkedlists.h. They are -efficient, easy to use and provide every operation that should be -necessary for managing a singly-linked list (if something is missing, -let us know!). Just because you see other open-coded list implementations -in the source tree is no reason to continue making new copies of -that code... There are also a number of common string manipulation -and timeval manipulation functions in asterisk/strings.h and asterisk/time.h; -use them when possible. - -- Avoid needless allocations! -Avoid needless malloc(), strdup() calls. If you only need the value in -the scope of your function try ast_strdupa() or declare structs on the -stack and pass a pointer to them. However, be careful to _never_ call -alloca(), ast_strdupa() or similar functions in the argument list -of a function you are calling; this can cause very strange stack -arrangements and produce unexpected behavior. - --Allocations for structures -When allocating/zeroing memory for a structure, try to use code like this: - -struct foo *tmp; - -... - -tmp = malloc(sizeof(*tmp)); -if (tmp) - memset(tmp, 0, sizeof(*tmp)); - -This eliminates duplication of the 'struct foo' identifier, which makes the -code easier to read and also ensures that if it is copy-and-pasted it won't -require as much editing. In fact, you can even use: - -struct foo *tmp; - -... - -tmp = calloc(1, sizeof(*tmp)); - -This will allocate and zero the memory in a single operation. - -* CLI Commands --------------- - -New CLI commands should be named using the module's name, followed by a verb -and then any parameters that the command needs. For example: - -*CLI> iax2 show peer <peername> - -not - -*CLI> show iax2 peer <peername> - -* New dialplan applications/functions -------------------------------------- - -There are two methods of adding functionality to the Asterisk -dialplan: applications and functions. Applications (found generally in -the apps/ directory) should be collections of code that interact with -a channel and/or user in some significant way. Functions (which can be -provided by any type of module) are used when the provided -functionality is simple... getting/retrieving a value, for -example. Functions should also be used when the operation is in no way -related to a channel (a computation or string operation, for example). - -Applications are registered and invoked using the -ast_register_application function; see the apps/app_skel.c file for an -example. - -Functions are registered using 'struct ast_custom_function' -structures and the ast_custom_function_register function. - -* Doxygen API Documentation Guidelines --------------------------------------- - -When writing Asterisk API documentation the following format should be -followed. Do not use the javadoc style. - -/*! - * \brief Do interesting stuff. - * \param thing1 interesting parameter 1. - * \param thing2 interesting parameter 2. - * - * This function does some interesting stuff. - * - * \return zero on success, -1 on error. - */ -int ast_interesting_stuff(int thing1, int thing2) -{ - return 0; -} - -Notice the use of the \param, \brief, and \return constructs. These should be -used to describe the corresponding pieces of the function being documented. -Also notice the blank line after the last \param directive. All doxygen -comments must be in one /*! */ block. If the function or struct does not need -an extended description it can be left out. - -Please make sure to review the doxygen manual and make liberal use of the \a, -\code, \c, \b, \note, \li and \e modifiers as appropriate. - -When documenting a 'static' function or an internal structure in a module, -use the \internal modifier to ensure that the resulting documentation -explicitly says 'for internal use only'. - -Structures should be documented as follows. - -/*! - * \brief A very interesting structure. - */ -struct interesting_struct -{ - /*! \brief A data member. */ - int member1; - - int member2; /*!< \brief Another data member. */ -} - -Note that /*! */ blocks document the construct immediately following them -unless they are written, /*!< */, in which case they document the construct -preceding them. - -* Finishing up before you submit your code ------------------------------------------- - -- Look at the code once more -When you achieve your desired functionalty, make another few refactor -passes over the code to optimize it. - -- Read the patch -Before submitting a patch, *read* the actual patch file to be sure that -all the changes you expect to be there are, and that there are no -surprising changes you did not expect. During your development, that -part of Asterisk may have changed, so make sure you compare with the -latest SVN. - -- Listen to advice -If you are asked to make changes to your patch, there is a good chance -the changes will introduce bugs, check it even more at this stage. -Also remember that the bug marshal or co-developer that adds comments -is only human, they may be in error :-) - -- Optimize, optimize, optimize -If you are going to reuse a computed value, save it in a variable -instead of recomputing it over and over. This can prevent you from -making a mistake in subsequent computations, making it easier to correct -if the formula has an error and may or may not help optimization but -will at least help readability. - -Just an example (so don't over analyze it, that'd be a shame): - -const char *prefix = "pre"; -const char *postfix = "post"; -char *newname; -char *name = "data"; - -if (name && (newname = alloca(strlen(name) + strlen(prefix) + strlen(postfix) + 3))) - snprintf(newname, strlen(name) + strlen(prefix) + strlen(postfix) + 3, "%s/%s/%s", prefix, name, postfix); - -...vs this alternative: - -const char *prefix = "pre"; -const char *postfix = "post"; -char *newname; -char *name = "data"; -int len = 0; - -if (name && (len = strlen(name) + strlen(prefix) + strlen(postfix) + 3) && (newname = alloca(len))) - snprintf(newname, len, "%s/%s/%s", prefix, name, postfix); - - ------------------------------------------------ -Welcome to the Asterisk development community! -Meet you on the asterisk-dev mailing list. -Subscribe at http://lists.digium.com! - -Mark Spencer, Kevin P. Fleming and -the Asterisk.org Development Team diff --git a/1.2-netsec/doc/PEERING b/1.2-netsec/doc/PEERING deleted file mode 100644 index c1f0b3831..000000000 --- a/1.2-netsec/doc/PEERING +++ /dev/null @@ -1,499 +0,0 @@ - DIGIUM GENERAL PEERING AGREEMENT (TM) - Version 1.0.0, September 2004 - Copyright (C) 2004 Digium, Inc. - 150 West Park Loop Suite 100, Huntsville, AL 35806 USA - - Everyone is permitted to copy and distribute complete verbatim copies - of this General Peering Agreement provided it is not modified in any - manner. - - ------------------------------------------------------ - - DIGIUM GENERAL PEERING AGREEMENT - - PREAMBLE - - For most of the history of telecommunications, the power of being able -to locate and communicate with another person in a system, be it across -a hall or around the world, has always centered around a centralized -authority -- from a local PBX administrator to regional and national -RBOCs, generally requiring fees, taxes or regulation. By contrast, -DUNDi is a technology developed to provide users the freedom to -communicate with each other without the necessity of any centralized -authority. This General Peering Agreement ("GPA") is used by individual -parties (each, a "Participant") to allow them to build the E164 trust -group for the DUNDi protocol. - - To protect the usefulness of the E164 trust group for those who use -it, while keeping the system wholly decentralized, it is necessary to -replace many of the responsibilities generally afforded to a company or -government agency, with a set of responsibilities implemented by the -parties who use the system, themselves. It is the goal of this document -to provide all the protections necessary to keep the DUNDi E164 trust -group useful and reliable. - - The Participants wish to protect competition, promote innovation and -value added services and make this service valuable both commercially -and non-commercially. To that end, this GPA provides special terms and -conditions outlining some permissible and non-permissible revenue -sources. - - This GPA is independent of any software license or other license -agreement for a program or technology employing the DUNDi protocol. For -example, the implementation of DUNDi used by Asterisk is covered under a -separate license. Each Participant is responsible for compliance with -any licenses or other agreements governing use of such program or -technology that they use to peer. - - You do not have to execute this GPA to use a program or technology -employing the DUNDi protocol, however if you do not execute this GPA, -you will not be able to peer using DUNDi and the E164 context with -anyone who is a member of the trust group by virtue of their having -executed this GPA with another member. - -The parties to this GPA agree as follows: - - 0. DEFINITIONS. As used herein, certain terms shall be defined as -follows: - - (a) The term "DUNDi" means the DUNDi protocol as published by - Digium, Inc. or its successor in interest with respect to the - DUNDi protocol specification. - - (b) The terms "E.164" and "E164" mean ITU-T specification E.164 as - published by the International Telecommunications Union (ITU) in - May, 1997. - - (c) The term "Service" refers to any communication facility (e.g., - telephone, fax, modem, etc.), identified by an E.164-compatible - number, and assigned by the appropriate authority in that - jurisdiction. - - (d) The term "Egress Gateway" refers an Internet facility that - provides a communications path to a Service or Services that may - not be directly addressable via the Internet. - - (e) The term "Route" refers to an Internet address, policies, and - other characteristics defined by the DUNDi protocol and - associated with the Service, or the Egress Gateway which - provides access to the specified Service. - - (f) The term "Propagate" means to accept or transmit Service and/or - Egress Gateway Routes only using the DUNDi protocol and the - DUNDi context "e164" without regard to case, and does not apply - to the exchange of information using any other protocol or - context. - - (g) The term "Peering System" means the network of systems that - Propagate Routes. - - (h) The term "Subscriber" means the owner of, or someone who - contracts to receive, the services identified by an E.164 - number. - - (i) The term "Authorizing Individual" means the Subscriber to a - number who has authorized a Participant to provide Routes - regarding their services via this Peering System. - - (j) The term "Route Authority" refers to a Participant that provides - an original source of said Route within the Peering System. - Routes are propagated from the Route Authorities through the - Peering System and may be cached at intermediate points. There - may be multiple Route Authorities for any Service. - - (k) The term "Participant" (introduced above) refers to any member - of the Peering System. - - (l) The term "Service Provider" refers to the carrier (e.g., - exchange carrier, Internet Telephony Service Provider, or other - reseller) that provides communication facilities for a - particular Service to a Subscriber, Customer or other End User. - - (m) The term "Weight" refers to a numeric quality assigned to a - Route as per the DUNDi protocol specification. The current - Weight definitions are shown in Exhibit A. - - 1. PEERING. The undersigned Participants agree to Propagate Routes -with each other and any other member of the Peering System and further -agree not to Propagate DUNDi Routes with a third party unless they have -first have executed this GPA (in its unmodified form) with such third -party. The Participants further agree only to Propagate Routes with -Participants whom they reasonably believe to be honoring the terms of -the GPA. Participants may not insert, remove, amend, or otherwise -modify any of the terms of the GPA. - - 2. ACCEPTABLE USE POLICY. The DUNDi protocol contains information -that reflect a Subscriber's or Egress Gateway's decisions to receive -calls. In addition to the terms and conditions set forth in this GPA, -the Participants agree to honor the intent of restrictions encoded in -the DUNDi protocol. To that end, Participants agree to the following: - - (a) A Participant may not utilize or permit the utilization of - Routes for which the Subscriber or Egress Gateway provider has - indicated that they do not wish to receive "Unsolicited Calls" - for the purpose of making an unsolicited phone call on behalf of - any party or organization. - - (b) A Participant may not utilize or permit the utilization of - Routes which have indicated that they do not wish to receive - "Unsolicited Commercial Calls" for the purpose of making an - unsolicited phone call on behalf of a commercial organization. - - (c) A Participant may never utilize or permit the utilization of any - DUNDi route for the purpose of making harassing phone calls. - - (d) A Party may not utilize or permit the utilization of DUNDi - provided Routes for any systematic or random calling of numbers - (e.g., for the purpose of locating facsimile, modem services, or - systematic telemarketing). - - (e) Initial control signaling for all communication sessions that - utilize Routes obtained from the Peering System must be sent - from a member of the Peering System to the Service or Egress - Gateway identified in the selected Route. For example, 'SIP - INVITES' and IAX2 "NEW" commands must be sent from the - requesting DUNDi node to the terminating Service. - - (f) A Participant may not disclose any specific Route, Service or - Participant contact information obtained from the Peering System - to any party outside of the Peering System except as a - by-product of facilitating communication in accordance with - section 2e (e.g., phone books or other databases may not be - published, but the Internet addresses of the Egress Gateway or - Service does not need to be obfuscated.) - - (g) The DUNDi Protocol requires that each Participant include valid - contact information about itself (including information about - nodes connected to each Participant). Participants may use or - disclose the contact information only to ensure enforcement of - legal furtherance of this Agreement. - - 3. ROUTES. The Participants shall only propagate valid Routes, as -defined herein, through the Peering System, regardless of the original -source. The Participants may only provide Routes as set forth below, -and then only if such Participant has no good faith reason to believe -such Route to be invalid or unauthorized. - - (a) A Participant may provide Routes if each Route has as its - original source another member of the Peering System who has - duly executed the GPA and such Routes are provided in accordance - with this Agreement; provided that the Routes are not modified - (e.g., with regards to existence, destination, technology or - Weight); or - - (b) A Participant may provide Routes for Services with any Weight - for which it is the Subscriber; or - - (c) A Participant may provide Routes for those Services whose - Subscriber has authorized the Participant to do so, provided - that the Participant is able to confirm that the Authorizing - Individual is the Subscriber through: - - i. a written statement of ownership from the Authorizing - Individual, which the Participant believes in good faith - to be accurate (e.g., a phone bill with the name of the - Authorizing Individual and the number in question); or - - ii. the Participant's own direct personal knowledge that the - Authorizing Individual is the Subscriber. - - (d) A Participant may provide Routes for Services, with Weight in - accordance with the Current DUNDi Specification, if it can in - good faith provide an Egress Gateway to that Service on the - traditional telephone network without cost to the calling party. - - 4. REVOCATION. A Participant must provide a free, easily accessible -mechanism by which a Subscriber may revoke permission to act as a Route -Authority for his Service. A Participant must stop acting as a Route -Authority for that Service within 7 days after: - - (a) receipt of a revocation request; - - (b) receiving other notice that the Service is no longer valid; or - - (c) determination that the Subscriber's information is no longer - accurate (including that the Subscriber is no longer the service - owner or the service owner's authorized delegate). - - 5. SERVICE FEES. A Participant may charge a fee to act as a Route -Authority for a Service, with any Weight, provided that no Participant -may charge a fee to propagate the Route received through the Peering -System. - - 6. TOLL SERVICES. No Participant may provide Routes for any Services -that require payment from the calling party or their customer for -communication with the Service. Nothing in this section shall prohibit -a Participant from providing routes for Services where the calling party -may later enter into a financial transaction with the called party -(e.g., a Participant may provide Routes for calling cards services). - - 7. QUALITY. A Participant may not intentionally impair communication -using a Route provided to the Peering System (e.g. by adding delay, -advertisements, reduced quality). If for any reason a Participant is -unable to deliver a call via a Route provided to the Peering System, -that Participant shall return out-of-band Network Congestion -notification (e.g. "503 Service Unavailable" with SIP protocol or -"CONGESTION" with IAX protocol). - - 8. PROTOCOL COMPLIANCE. Participants agree to Propagate Routes in -strict compliance with current DUNDi protocol specifications. - - 9. ADMINISTRATIVE FEES. A Participant may charge (but is not required -to charge) another Participant a reasonable fee to cover administrative -expenses incurred in the execution of this Agreement. A Participant may -not charge any fee to continue the relationship or to provide Routes to -another Participant in the Peering System. - - 10. CALLER IDENTIFICATION. A Participant will make a good faith effort -to ensure the accuracy and appropriate nature of any caller -identification that it transmits via any Route obtained from the Peering -System. Caller identification shall at least be provided as a valid -E.164 number. - - 11. COMPLIANCE WITH LAWS. The Participants are solely responsible for -determining to what extent, if any, the obligations set forth in this -GPA conflict with any laws or regulations their region. A Participant -may not provide any service or otherwise use DUNDi under this GPA if -doing so is prohibited by law or regulation, or if any law or regulation -imposes requirements on the Participant that are inconsistent with the -terms of this GPA or the Acceptable Use Policy. - - 12. WARRANTY. EACH PARTICIPANT WARRANTS TO THE OTHER PARTICIPANTS THAT -IT MADE, AND WILL CONTINUE TO MAKE, A GOOD FAITH EFFORT TO AUTHENTICATE -OTHERS IN THE PEERING SYSTEM AND TO PROVIDE ACCURATE INFORMATION IN -ACCORDANCE WITH THE TERMS OF THIS GPA. THIS WARRANTY IS MADE BETWEEN -THE PARTICIPANTS, AND THE PARTICIPANTS MAY NOT EXTEND THIS WARRANTY TO -ANY NON-PARTICIPANT INCLUDING END-USERS. - - 13. DISCLAIMER OF WARRANTIES. THE PARTICIPANTS UNDERSTAND AND AGREE -THAT ANY SERVICE PROVIDED AS A RESULT OF THIS GPA IS "AS IS." EXCEPT FOR -THOSE WARRANTIES OTHERWISE EXPRESSLY SET FORTH HEREIN, THE PARTICIPANTS -DISCLAIM ANY REPRESENTATIONS OR WARRANTIES OF ANY KIND OR NATURE, -EXPRESS OR IMPLIED, AS TO THE CONDITION, VALUE OR QUALITIES OF THE -SERVICES PROVIDED HEREUNDER, AND SPECIFICALLY DISCLAIM ANY -REPRESENTATION OR WARRANTY OF MERCHANTABILITY, SUITABILITY OR FITNESS -FOR A PARTICULAR PURPOSE OR AS TO THE CONDITION OR WORKMANSHIP THEREOF, -OR THE ABSENCE OF ANY DEFECTS THEREIN, WHETHER LATENT OR PATENT, -INCLUDING ANY WARRANTIES ARISING FROM A COURSE OF DEALING, USAGE OR -TRADE PRACTICE. EXCEPT AS EXPRESSLY PROVIDED HEREIN, THE PARTICIPANTS -EXPRESSLY DISCLAIM ANY REPRESENTATIONS OR WARRANTIES THAT THE PEERING -SERVICE WILL BE CONTINUOUS, UNINTERRUPTED OR ERROR-FREE, THAT ANY DATA -SHARED OR OTHERWISE MADE AVAILABLE WILL BE ACCURATE OR COMPLETE OR -OTHERWISE COMPLETELY SECURE FROM UNAUTHORIZED ACCESS. - - 14. LIMITATION OF LIABILITIES. NO PARTICIPANT SHALL BE LIABLE TO ANY -OTHER PARTICIPANT FOR INCIDENTAL, INDIRECT, CONSEQUENTIAL, SPECIAL, -PUNITIVE OR EXEMPLARY DAMAGES OF ANY KIND (INCLUDING LOST REVENUES OR -PROFITS, LOSS OF BUSINESS OR LOSS OF DATA) IN ANY WAY RELATED TO THIS -GPA, WHETHER IN CONTRACT OR IN TORT, REGARDLESS OF WHETHER SUCH -PARTICIPANT WAS ADVISED OF THE POSSIBILITY THEREOF. - - 15. END-USER AGREEMENTS. The Participants may independently enter -into agreements with end-users to provide certain services (e.g., fees -to a Subscriber to originate Routes for that Service). To the extent -that provision of these services employs the Peering System, the Parties -will include in their agreements with their end-users terms and -conditions consistent with the terms of this GPA with respect to the -exclusion of warranties, limitation of liability and Acceptable Use -Policy. In no event may a Participant extend the warranty described in -Section 12 in this GPA to any end-users. - - 16. INDEMNIFICATION. Each Participant agrees to defend, indemnify and -hold harmless the other Participant or third-party beneficiaries to this -GPA (including their affiliates, successors, assigns, agents and -representatives and their respective officers, directors and employees) -from and against any and all actions, suits, proceedings, -investigations, demands, claims, judgments, liabilities, obligations, -liens, losses, damages, expenses (including, without limitation, -attorneys' fees) and any other fees arising out of or relating to (i) -personal injury or property damage caused by that Participant, its -employees, agents, servants, or other representatives; (ii) any act or -omission by the Participant, its employees, agents, servants or other -representatives, including, but not limited to, unauthorized -representations or warranties made by the Participant; or (iii) any -breach by the Participant of any of the terms or conditions of this GPA. - - 17. THIRD PARTY BENEFICIARIES. This GPA is intended to benefit those -Participants who have executed the GPA and who are in the Peering -System. It is the intent of the Parties to this GPA to give to those -Participants who are in the Peering System standing to bring any -necessary legal action to enforce the terms of this GPA. - - 18. TERMINATION. Any Participant may terminate this GPA at any time, -with or without cause. A Participant that terminates must immediately -cease to Propagate. - - 19. CHOICE OF LAW. This GPA and the rights and duties of the Parties -hereto shall be construed and determined in accordance with the internal -laws of the State of New York, United States of America, without regard -to its conflict of laws principles and without application of the United -Nations Convention on Contracts for the International Sale of Goods. - - 20. DISPUTE RESOLUTION. Unless otherwise agreed in writing, the -exclusive procedure for handling disputes shall be as set forth herein. -Notwithstanding such procedures, any Participant may, at any time, seek -injunctive relief in addition to the process described below. - - (a) Prior to mediation or arbitration the disputing Participants - shall seek informal resolution of disputes. The process shall be - initiated with written notice of one Participant to the other - describing the dispute with reasonable particularity followed - with a written response within ten (10) days of receipt of - notice. Each Participant shall promptly designate an executive - with requisite authority to resolve the dispute. The informal - procedure shall commence within ten (10) days of the date of - response. All reasonable requests for non-privileged information - reasonably related to the dispute shall be honored. If the - dispute is not resolved within thirty (30) days of commencement - of the procedure either Participant may proceed to mediation or - arbitration pursuant to the rules set forth in (b) or (c) below. - - (b) If the dispute has not been resolved pursuant to (a) above or, - if the disputing Participants fail to commence informal dispute - resolution pursuant to (a) above, either Participant may, in - writing and within twenty (20) days of the response date noted - in (a) above, ask the other Participant to participate in a one - (1) day mediation with an impartial mediator, and the other - Participant shall do so. Each Participant will bear its own - expenses and an equal share of the fees of the mediator. If the - mediation is not successful the Participants may proceed with - arbitration pursuant to (c) below. - - (c) If the dispute has not been resolved pursuant to (a) or (b) - above, the dispute shall be promptly referred, no later than one - (1) year from the date of original notice and subject to - applicable statute of limitations, to binding arbitration in - accordance with the UNCITRAL Arbitration Rules in effect on the - date of this contract. The appointing authority shall be the - International Centre for Dispute Resolution. The case shall be - administered by the International Centre for Dispute Resolution - under its Procedures for Cases under the UNCITRAL Arbitration - Rules. Each Participant shall bear its own expenses and shall - share equally in fees of the arbitrator. All arbitrators shall - have substantial experience in information technology and/or in - the telecommunications business and shall be selected by the - disputing participants in accordance with UNCITRAL Arbitration - Rules. If any arbitrator, once selected is unable or unwilling - to continue for any reason, replacement shall be filled via the - process described above and a re-hearing shall be conducted. The - disputing Participants will provide each other with all - requested documents and records reasonably related to the - dispute in a manner that will minimize the expense and - inconvenience of both parties. Discovery will not include - depositions or interrogatories except as the arbitrators - expressly allow upon a showing of need. If disputes arise - concerning discovery requests, the arbitrators shall have sole - and complete discretion to resolve the disputes. The parties and - arbitrator shall be guided in resolving discovery disputes by - the Federal Rules of Civil Procedure. The Participants agree - that time of the essence principles shall guide the hearing and - that the arbitrator shall have the right and authority to issue - monetary sanctions in the event of unreasonable delay. The - arbitrator shall deliver a written opinion setting forth - findings of fact and the rationale for the award within thirty - (30) days following conclusion of the hearing. The award of the - arbitrator, which may include legal and equitable relief, but - which may not include punitive damages, will be final and - binding upon the disputing Participants, and judgment may be - entered upon it in accordance with applicable law in any court - having jurisdiction thereof. In addition to award the - arbitrator shall have the discretion to award the prevailing - Participant all or part of its attorneys' fees and costs, - including fees associated with arbitrator, if the arbitrator - determines that the positions taken by the other Participant on - material issues of the dispute were without substantial - foundation. Any conflict between the UNCITRAL Arbitration Rules - and the provisions of this GPA shall be controlled by this GPA. - - 21. INTEGRATED AGREEMENT. This GPA, constitutes the complete -integrated agreement between the parties concerning the subject matter -hereof. All prior and contemporaneous agreements, understandings, -negotiations or representations, whether oral or in writing, relating to -the subject matter of this GPA are superseded and canceled in their -entirety. - - 22. WAIVER. No waiver of any of the provisions of this GPA shall be -deemed or shall constitute a waiver of any other provision of this GPA, -whether or not similar, nor shall such waiver constitute a continuing -waiver unless otherwise expressly so provided in writing. The failure -of either party to enforce at any time any of the provisions of this -GPA, or the failure to require at any time performance by either party -of any of the provisions of this GPA, shall in no way be construed to be -a present or future waiver of such provisions, nor in any way affect the -ability of a Participant to enforce each and every such provision -thereafter. - - 23. INDEPENDENT CONTRACTORS. Nothing in this GPA shall make the -Parties partners, joint venturers, or otherwise associated in or with -the business of the other. Parties are, and shall always remain, -independent contractors. No Participant shall be liable for any debts, -accounts, obligations, or other liabilities of the other Participant, -its agents or employees. No party is authorized to incur debts or other -obligations of any kind on the part of or as agent for the other. This -GPA is not a franchise agreement and does not create a franchise -relationship between the parties, and if any provision of this GPA is -deemed to create a franchise between the parties, then this GPA shall -automatically terminate. - - 24. CAPTIONS AND HEADINGS. The captions and headings used in this GPA -are used for convenience only and are not to be given any legal effect. - - 25. EXECUTION. This GPA may be executed in counterparts, each of which -so executed will be deemed to be an original and such counterparts -together will constitute one and the same Agreement. The Parties shall -transmit to each other a signed copy of the GPA by any means that -faithfully reproduces the GPA along with the Signature. For purposes of -this GPA, the term "signature" shall include digital signatures as -defined by the jurisdiction of the Participant signing the GPA. - - Exhibit A - -Weight Range Requirements - -0-99 May only be used under authorization of Owner - -100-199 May only be used by the Owner's service - provider, regardless of authorization. - -200-299 Reserved -- do not use for e164 context. - -300-399 May only be used by the owner of the code under - which the Owner's number is a part of. - -400-499 May be used by any entity providing access via - direct connectivity to the Public Switched - Telephone Network. - -500-599 May be used by any entity providing access via - indirect connectivity to the Public Switched - Telephone Network (e.g. Via another VoIP - provider) - -600- Reserved-- do not use for e164 context. - - Participant Participant - -Company: - -Address: - -Email: - - - _________________________ _________________________ - Authorized Signature Authorized Signature - -Name: - - -END OF GENERAL PEERING AGREEMENT - ------------------------------------------------- - -How to Peer using this GPA If you wish to exchange routing information -with parties using the e164 DUNDi context, all you must do is execute -this GPA with any member of the Peering System and you will become a -member of the Peering System and be able to make Routes available in -accordance with this GPA. - -DUNDi, IAX, Asterisk and GPA are trademarks of Digium, Inc. diff --git a/1.2-netsec/doc/README.ael b/1.2-netsec/doc/README.ael deleted file mode 100644 index e2014a4e1..000000000 --- a/1.2-netsec/doc/README.ael +++ /dev/null @@ -1,273 +0,0 @@ -The Asterisk Extension Language -=================================== - -Over time, people have been pushing to add features to extensions.conf to make -it more like a programming language. AEL is intended to provide an actual -programming language that can be used to write an Asterisk dialplan. - -Getting Started -------------------------- -The AEL parser (pbx_ael.so) is completely separate from the module -that parses extensions.conf (pbx_config.so). To use AEL, the only thing that -has to be done is the module pbx_ael.so must be loaded by Asterisk. This will -be done automatically if using 'autoload=yes' in /etc/asterisk/modules.conf. -When the module is loaded, it will look for 'extensions.ael' in /etc/asterisk/. -Both extensions.conf and extensions.ael can be used in conjunction with each -other if that is what is desired. Some users may want to keep extensions.conf -for the features that are configured in the 'general' section of -extensions.conf. - - -Reloading extensions.ael -------------------------- -To reload extensions.ael, the following command can be issued at the CLI. - - *CLI> reload pbx_ael.so - - -Contexts -------------------------- -Contexts in AEL represent a set of extensions in the same way that they do -in extensions.conf. - -context default { - -}; - - -Extensions -------------------------- -To specify an extension in a context, the following syntax is used. If more -than one application is be called in an extension, they can be listed in order -inside of a block. - -context default { - 1234 => Playback(tt-monkeys); - 8000 => { - NoOp(one); - NoOp(two); - NoOp(three); - }; - _5XXX => NoOp(it's a pattern!); -}; - - -Includes -------------------------- -Contexts can be included in other contexts. All included contexts are listed -within a single block. - -context default { - includes { - local; - longdistance; - international; - }; -}; - - -Dialplan Switches -------------------------- -Switches are listed in their own block within a context. - -context default { - switches { - DUNDi/e164; - IAX2/box5; - }; - eswitches { - IAX2/context@${CURSERVER}; - }; -}; - - -Ignorepat -------------------------- -ignorepat can be used to instruct channel drivers to not cancel dialtone upon -receipt of a particular pattern. The most commonly used example is '9'. - -context outgoing { - ignorepat => 9; -}; - - -Variables -------------------------- -Variables in Asterisk do not have a type, so to define a variable, it just has -to be specified with a value. - -Global variables are set in their own block. - -globals { - CONSOLE=Console/dsp; - TRUNK=Zap/g2; -}; - -Variables can be set within extensions as well. - -context foo { - 555 => { - x=5; - y=blah; - NoOp(x is ${x} and y is ${y} !); - }; -}; - -Writing to a dialplan function is treated the same as writing to a variable. - -context blah { - s => { - CALLERID(name)=ChickenMan; - NoOp(My name is ${CALLERID(name)} !); - }; -}; - - -Loops -------------------------- -AEL has implementations of 'for' and 'while' loops. - -context loops { - 1 => { - for (x=0; ${x} < 3; x=${x} + 1) { - Verbose(x is ${x} !); - }; - }; - 2 => { - y=10; - while (${y} >= 0) { - Verbose(y is ${y} !); - y=${y}-1; - }; - }; -}; - - -Conditionals -------------------------- -AEL supports if and switch statements. Note that if you have an else -clause, you MUST place braces around the non-else portion of the if -statement. - -context conditional { - _8XXX => { - Dial(SIP/${EXTEN}); - if (${DIALSTATUS} = "BUSY") { - Voicemail(${EXTEN}|b); - } else - Voicemail(${EXTEN}|u); - }; - _777X => { - switch (${EXTEN}) { - case 7771: - NoOp(You called 7771!); - break; - case 7772: - NoOp(You called 7772!); - break; - case 7773: - NoOp(You called 7773!); - // fall through - default: - NoOp(In the default clause!); - }; - }; -}; - - -goto and labels -------------------------- -This is an example of how to do a goto in AEL. - -context gotoexample { - s => { -begin: - NoOp(Infinite Loop! yay!); - Wait(1); - goto begin; - }; -}; - - -Macros -------------------------- -A macro is defined in its own block like this. The arguments to the macro are -specified with the name of the macro. They are then reffered to by that same -name. A catch block can be specified to catch special extensions. - -macro std-exten( ext , dev ) { - Dial(${dev}/${ext},20); - switch(${DIALSTATUS) { - case BUSY: - Voicemail(b${ext}); - break; - default: - Voicemail(u${ext}); - }; - catch a { - VoiceMailMain(${ext}); - return; - }; -}; - -A macro is then called by preceeding the macro name with an ampersand. - -context example { - _5XXX => &std-exten(${EXTEN}, "IAX2"); -}; - - -Examples ------------------------- - -context demo { - s => { - Wait(1); - Answer(); - TIMEOUT(digit)=5; - TIMEOUT(response)=10; -restart: - Background(demo-congrats); -instructions: - for (x=0; ${x} < 3; x=${x} + 1) { - Background(demo-instruct); - WaitExten(); - }; - }; - 2 => { - Background(demo-moreinfo); - goto s|instructions; - }; - 3 => { - LANGUAGE()=fr; - goto s|restart; - }; - 500 => { - Playback(demo-abouttotry); - Dial(IAX2/guest@misery.digium.com); - Playback(demo-nogo); - goto s|instructions; - }; - 600 => { - Playback(demo-echotest); - Echo(); - Playback(demo-echodone); - goto s|instructions; - }; - # => { -hangup: - Playback(demo-thanks); - Hangup(); - }; - t => goto #|hangup; - i => Playback(invalid); -}; - - -Syntax Note ------------------------- -Please note that all opening {'s are on the same line as the keyword. For -the time being, that syntax is mandatory. We are looking at ways to allow -other syntax in the future for flexibility, but for now, that is the way -you must write AEL clauses. - diff --git a/1.2-netsec/doc/README.asterisk.conf b/1.2-netsec/doc/README.asterisk.conf deleted file mode 100644 index 4e210aacd..000000000 --- a/1.2-netsec/doc/README.asterisk.conf +++ /dev/null @@ -1,76 +0,0 @@ -Asterisk Main Configuration File ------------------------------------------------------ -Below is a sample of the main Asterisk configuration file, -asterisk.conf. Note that this file is _not_ provided in -sample form, because the Makefile creates it when needed -and does not touch it when it already exists. - ---------------- - -[directories] -; Make sure these directoriess have the right permissions if not -; running Asterisk as root - -; Where the configuration files (except for this one) are located -astetcdir => /etc/asterisk - -; Where the Asterisk loadable modules are located -astmoddir => /usr/lib/asterisk/modules - -; Where additional 'library' elements (scripts, etc.) are located -astvarlibdir => /var/lib/asterisk - -; Where AGI scripts/programs are located -astagidir => /var/lib/asterisk/agi-bin - -; Where spool directories are located -; Voicemail, monitor, dictation and other apps will create files here -; and outgoing call files (used with pbx_spool) must be placed here -astspooldir => /var/spool/asterisk - -; Where the Asterisk process ID (pid) file should be created -astrundir => /var/run/asterisk - -; Where the Asterisk log files should be created -astlogdir => /var/log/asterisk - - -[options] -;Under "options" you can enter configuration options -;that you also can set with command line options - -verbose = 0 ; Verbosity level for logging (-v) -debug = 3 ; Debug: "No" or value (1-4) -nofork=yes | no ; Background execution disabled (-f) -console= yes | no ; Console mode (-c) -highpriority = yes | no ; Execute with high priority (-p) -initcrypto = yes | no ; Initialize crypto at startup (-i) -nocolor = yes | no ; Disable ANSI colors (-n) -dumpcore = yes | no ; Dump core on failure (-g) -quiet = yes | no ; Run quietly (-q) -timestamp = yes | no ; Force timestamping on log entries to console (-T) -runuser = asterisk ; User to run asterisk as (-U) NOTE: will require changes to - ; directory and device permisions -rungroup = asterisk ; Group to run asterisk as (-G) - -;These options have no command line equivalent -cache_record_files = yes | no ; Cache record() files in another directory until completion -record_cache_dir = <dir> -transcode_via_sln = yes | no ; Build transcode paths via SLINEAR -transmit_silence_during_record = yes | no ; send SLINEAR silence while channel is being recorded -maxload = 1.0 ; The maximum load average we accept calls for -maxcalls = 255 ; The maximum number of concurrent calls you want to allow -execincludes = yes | no ; Allow #exec entries in configuration files -dontwarn = yes | no ; Don't over-inform the Asterisk sysadm, he's a guru - -[files] -; Changing the following lines may compromise your security -; Asterisk.ctl is the pipe that is used to connect the remote CLI -; (asterisk -r) to Asterisk. Changing these settings change the -; permissions and ownership of this file. -; The file is created when Asterisk starts, in the "astrundir" above. - -;astctlpermissions = 0660 -;astctlowner = root -;astctlgroup = asterisk -;astctl = asterisk.ctl diff --git a/1.2-netsec/doc/README.backtrace b/1.2-netsec/doc/README.backtrace deleted file mode 100644 index 64d30a357..000000000 --- a/1.2-netsec/doc/README.backtrace +++ /dev/null @@ -1,187 +0,0 @@ -This document is to provide information on how to obtain the -backtraces required on the asterisk bug tracker, available at -http://bugs.digium.com. The information is required by developers to -help fix problem with bugs of any kind. Backtraces provide information -about what was wrong when a program crashed; in our case, -Asterisk. There are two kind of backtraces (aka 'bt'), which are -useful: bt and bt full. - -First of all, when you start Asterisk, you MUST start it with option --g (this tells Asterisk to produce a core file if it crashes). - -If you start Asterisk with the safe_asterisk script, it automatically -starts using the option -g. - -If you're not sure if Asterisk is running with the -g option, type the -following command in your shell: - -debian:/tmp# ps aux | grep asterisk -root 17832 0.0 1.2 2348 788 pts/1 S Aug12 0:00 /bin/sh /usr/sbin/safe_asterisk -root 26686 0.0 2.8 15544 1744 pts/1 S Aug13 0:02 asterisk -vvvg -c -[...] - -The interesting information is located in the last column. - -Second, your copy of Asterisk must have been built without -optimization or the backtrace will be (nearly) unusable. This can be -done by using 'make dont-optimize' intead of 'make install' to build -and install the Asterisk binary and modules. - -After Asterisk crashes, a core file will be "dumped" in your /tmp/ -directory. To make sure it's really there, you can just type the -following command in your shell: - -debian:/tmp# ls -l /tmp/core.* --rw------- 1 root root 10592256 Aug 12 19:40 /tmp/core.26252 --rw------- 1 root root 9924608 Aug 12 20:12 /tmp/core.26340 --rw------- 1 root root 10862592 Aug 12 20:14 /tmp/core.26374 --rw------- 1 root root 9105408 Aug 12 20:19 /tmp/core.26426 --rw------- 1 root root 9441280 Aug 12 20:20 /tmp/core.26462 --rw------- 1 root root 8331264 Aug 13 00:32 /tmp/core.26647 -debian:/tmp# - -Now that we've verified the core file has been written to disk, the -final part is to extract 'bt' from the core file. Core files are -pretty big, don't be scared, it's normal. - -*** NOTE: Don't attach core files on the bug tracker, we only need the bt and bt full. *** - -For extraction, we use a really nice tool, called gdb. To verify that -you have gdb installed on your system: - -debian:/tmp# gdb -v -GNU gdb 6.3-debian -Copyright 2004 Free Software Foundation, Inc. -GDB is free software, covered by the GNU General Public License, and you are -welcome to change it and/or distribute copies of it under certain conditions. -Type "show copying" to see the conditions. -There is absolutely no warranty for GDB. Type "show warranty" for details. -This GDB was configured as "i386-linux". -debian:/tmp# - -Which is great, we can continue. If you don't have gdb installed, go install gdb. - -Now load the core file in gdb, as follows: - -debian:/tmp# gdb -se "asterisk" -c /tmp/core.26252 -[...] -(You would see a lot of output here.) -[...] -Reading symbols from /usr/lib/asterisk/modules/app_externalivr.so...done. -Loaded symbols for /usr/lib/asterisk/modules/app_externalivr.so -#0 0x29b45d7e in ?? () -(gdb) - -Now at the gdb prompt, type: bt -You would see output similar to: -(gdb) bt -#0 0x29b45d7e in ?? () -#1 0x08180bf8 in ?? () -#2 0xbcdffa58 in ?? () -#3 0x08180bf8 in ?? () -#4 0xbcdffa60 in ?? () -#5 0x08180bf8 in ?? () -#6 0x180bf894 in ?? () -#7 0x0bf80008 in ?? () -#8 0x180b0818 in ?? () -#9 0x08068008 in ast_stopstream (tmp=0x40758d38) at file.c:180 -#10 0x000000a0 in ?? () -#11 0x000000a0 in ?? () -#12 0x00000000 in ?? () -#13 0x407513c3 in confcall_careful_stream (conf=0x8180bf8, filename=0x8181de8 "Zap/pseudo-1324221520") at app_meetme.c:262 -#14 0x40751332 in streamconfthread (args=0x8180bf8) at app_meetme.c:1965 -#15 0xbcdffbe0 in ?? () -#16 0x40028e51 in pthread_start_thread () from /lib/libpthread.so.0 -#17 0x401ec92a in clone () from /lib/libc.so.6 -(gdb) - - -The bt's output is the information that we need on the bug tracker. - -Now do a bt full as follows: -(gdb) bt full -#0 0x29b45d7e in ?? () -No symbol table info available. -#1 0x08180bf8 in ?? () -No symbol table info available. -#2 0xbcdffa58 in ?? () -No symbol table info available. -#3 0x08180bf8 in ?? () -No symbol table info available. -#4 0xbcdffa60 in ?? () -No symbol table info available. -#5 0x08180bf8 in ?? () -No symbol table info available. -#6 0x180bf894 in ?? () -No symbol table info available. -#7 0x0bf80008 in ?? () -No symbol table info available. -#8 0x180b0818 in ?? () -No symbol table info available. -#9 0x08068008 in ast_stopstream (tmp=0x40758d38) at file.c:180 -No locals. -#10 0x000000a0 in ?? () -No symbol table info available. -#11 0x000000a0 in ?? () -No symbol table info available. -#12 0x00000000 in ?? () -No symbol table info available. -#13 0x407513c3 in confcall_careful_stream (conf=0x8180bf8, filename=0x8181de8 "Zap/pseudo-1324221520") at app_meetme.c:262 - f = (struct ast_frame *) 0x8180bf8 - trans = (struct ast_trans_pvt *) 0x0 -#14 0x40751332 in streamconfthread (args=0x8180bf8) at app_meetme.c:1965 -No locals. -#15 0xbcdffbe0 in ?? () -No symbol table info available. -#16 0x40028e51 in pthread_start_thread () from /lib/libpthread.so.0 -No symbol table info available. -#17 0x401ec92a in clone () from /lib/libc.so.6 -No symbol table info available. -(gdb) - -We also need gdb's output. That output gives more details compared to -the simple "bt". So we recommend that you use bt full instead of bt. -But, if you could include both, we appreciate that. - -The final "extraction" would the to know all traces by all -threads. Even if asterisk runs on the same thread for each calls, it -could have some new threads created. - -To make sure we have the correct informations, just do: -(gdb) thread apply all bt - -Thread 1 (process 26252): -#0 0x29b45d7e in ?? () -#1 0x08180bf8 in ?? () -#2 0xbcdffa58 in ?? () -#3 0x08180bf8 in ?? () -#4 0xbcdffa60 in ?? () -#5 0x08180bf8 in ?? () -#6 0x180bf894 in ?? () -#7 0x0bf80008 in ?? () -#8 0x180b0818 in ?? () -#9 0x08068008 in ast_stopstream (tmp=0x40758d38) at file.c:180 -#10 0x000000a0 in ?? () -#11 0x000000a0 in ?? () -#12 0x00000000 in ?? () -#13 0x407513c3 in confcall_careful_stream (conf=0x8180bf8, filename=0x8181de8 "Zap/pseudo-1324221520") at app_meetme.c:262 -#14 0x40751332 in streamconfthread (args=0x8180bf8) at app_meetme.c:1965 -#15 0xbcdffbe0 in ?? () -#16 0x40028e51 in pthread_start_thread () from /lib/libpthread.so.0 -#17 0x401ec92a in clone () from /lib/libc.so.6 -(gdb) - - -That output tell us crucial informations for threads. - -Now, just create a output.txt and dump your "bt full" (and/or "bt") -AND with "thread apply all bt". - -Note: Please ATTACH your output, DO NOT paste it as a note. - -And you're ready for upload on bug tracker. - - -Questions or comments regarding this documentation, feel free to pass -by #asterisk-bugs on FreeNode. - diff --git a/1.2-netsec/doc/README.cdr b/1.2-netsec/doc/README.cdr deleted file mode 100644 index cb9813845..000000000 --- a/1.2-netsec/doc/README.cdr +++ /dev/null @@ -1,122 +0,0 @@ -Asterisk billing support - Call Detail Records ----------------------------------------------- -Asterisk generates Call Detail Records in a database or in a comma -separated text file. - - * cdr_csv supports comma separated text file storage, this is the - default driver - * cdr_manager supports CDR information via the AMI, The Asterisk Manager - interface - * cdr_odbc supports UnixODBC databases, see http://www.unixodbc.org - for an updated list of supported databases, from MySQL to MsSQL - and text files. - * cdr_tds supports FreeTDS databases (Among them MS SQL) - NOTE: Please read README.tds for information on possible - problems with the FreeTDS driver - * cdr_sqlite supports SQlite - * cdr_pgsql supports PostgreSQL - -In the asterisk-addons cvs archive, there's a cdr_mysql driver for -MySQL. - - - -Applications ------------- - - * SetAccount Set account code for billing - * SetAMAFlags Sets AMA flags - * NoCDR Make sure no CDR is saved for a specific call - * ResetCDR Reset CDR - * ForkCDR Save current CDR and start a new CDR for this call - * Authenticate Authenticates and sets the account code - * SetCDRUserField Set CDR user field - * AppendCDRUserField Append data to CDR User field - -For more information, use the "show application" command. -You can set default account codes and AMA flags for devices in -channel configuration files, like sip.conf, iax.conf etc. - - -Fields of the CDR in Asterisk ------------------------------ - - 1. accountcode: What account number to use, (string, 20 characters) - 2. src: Caller*ID number (string, 80 characters) - 3. dst: Destination extension (string, 80 characters) - 4. dcontext: Destination context (string, 80 characters) - 5. clid: Caller*ID with text (80 characters) - 6. channel: Channel used (80 characters) - 7. dstchannel: Destination channel if appropriate (80 characters) - 8. lastapp: Last application if appropriate (80 characters) - 9. lastdata: Last application data (arguments) (80 characters) - 10. start: Start of call (date/time) - 11. answer: Answer of call (date/time) - 12. end: End of call (date/time) - 13. duration: Total time in system, in seconds (integer), from dial to hangup - 14. billsec: Total time call is up, in seconds (integer), from answer to hangup - 15. disposition: What happened to the call: ANSWERED, NO ANSWER, BUSY - 16. amaflags: What flags to use: DOCUMENTATION, BILL, IGNORE etc, - specified on a per channel basis like accountcode. - 17. user field: A user-defined field, maximum 255 characters - -In some cases, uniqueid is appended: - - * uniqueid: Unique Channel Identifier (32 characters) - This needs to be enabled in the source code at compile time - - -ONE IMPORTANT NOTE: If you are trying to collect records on IAX to IAX calls -you need to be aware that by default, IAX will attempt to transfer calls -in this situation (if DTMF is not required). When the transfer is completed -the call is dumped from the middle machine and thus the call detail records -will report a short call time. If you want detailed records you must -turn off IAX transfer, but unless your servers are very close together, you -will definitely get a latency hit from doing so. - -____________________________________ -CDR Variables ------------------------------------- - -If the channel has a cdr, that cdr record has its own set of variables which -can be accessed just like channel variables. The following builtin variables -are available. - -${CDR(clid)} Caller ID -${CDR(src)} Source -${CDR(dst)} Destination -${CDR(dcontext)} Destination context -${CDR(channel)} Channel name -${CDR(dstchannel)} Destination channel -${CDR(lastapp)} Last app executed -${CDR(lastdata)} Last app's arguments -${CDR(start)} Time the call started. -${CDR(answer)} Time the call was answered. -${CDR(end)} Time the call ended. -${CDR(duration)} Duration of the call. -${CDR(billsec)} Duration of the call once it was answered. -${CDR(disposition)} ANSWERED, NO ANSWER, BUSY -${CDR(amaflags)} DOCUMENTATION, BILL, IGNORE etc -${CDR(accountcode)} The channel's account code. -${CDR(uniqueid)} The channel's unique id. -${CDR(userfield)} The channels uses specified field. - -In addition, you can set your own extra variables by using Set(CDR(name)=value). - -______________________________ -cdr_csv2 ------------------------------- - -This module is an experimental new cdr module to demonstrate the cdr vars. -usage( - -*) Create a file called cdr.conf and place it in your /etc/asterisk (or wherever your config files are) in the [cdr_csv2] section. -*) Add an entry called format to indicate any format you want for the output. - -The following format string will emulate the regular cdr file format: -[cdr_csv2] - -format => "${CDR(clid)}","${CDR(src)}","${CDR(dst)}","${CDR(dcontext)}","${CDR(channel)}","${CDR(dstchannel)}","${CDR(lastapp)}","${CDR(lastdata)}","${CDR(start)}","${CDR(answer)}","${CDR(end)}","${CDR(duration)}","${CDR(billsec)}","${CDR(disposition)}","${CDR(amaflags)}","${CDR(accountcode)}","${CDR(uniqueid)}","${CDR(userfield)}" - -You can put anything you want as the value of format incuding new cdr vars you make up or any global variables. - diff --git a/1.2-netsec/doc/README.cliprompt b/1.2-netsec/doc/README.cliprompt deleted file mode 100644 index 281d3df5f..000000000 --- a/1.2-netsec/doc/README.cliprompt +++ /dev/null @@ -1,28 +0,0 @@ -* Changing the CLI Prompt -------------------------- - -The CLI prompt is set with the ASTERISK_PROMPT UNIX environment variable that -you set from the Unix shell before starting Asterisk - -You may include the following variables, that will be replaced by -the current value by Asterisk: - -%d Date (year-month-date) -%h Full hostname -%H Short hostname -%t Time -%% Percent sign -%# '#' if Asterisk is run in console mode, '>' if running as remote console -%Cn[;n] Change terminal foreground (and optional background) color to specified - A full list of colors may be found in include/asterisk/term.h - -On Linux systems, you may also use -%l1 Load average over past minute -%l2 Load average over past 5 minutes -%l3 Load average over past 15 minutes -%l4 Process fraction (processes running / total processes) -%l5 The most recently allocated pid - - ------ -04-03-26 diff --git a/1.2-netsec/doc/README.configuration b/1.2-netsec/doc/README.configuration deleted file mode 100644 index 60a077a32..000000000 --- a/1.2-netsec/doc/README.configuration +++ /dev/null @@ -1,180 +0,0 @@ -Asterisk Configuration Parser (version 1.1 and later) ------------------------------------------------------ - -The Asterisk configuration parser in the 1.1 development version (1.2 -stable) and beyond series has been improved in a number of ways. In -addition to the realtime architecture, we now have the ability to create -templates in configuration files, and use these as templates when we -configure phones, voicemail accounts and queues. - -These changes are general to the configuration parser, and works in -all configuration files. - -General syntax --------------- -Asterisk configuration files are defined as follows: - - [section] - label = value - label2 = value - -In some files, (e.g. mgcp.conf, zapata.conf and agents.conf), the syntax -is a bit different. In these files the syntax is as follows: - - [section] - label1 = value1 - label2 = value2 - object => name - - label3 = value3 - label2 = value4 - object2 => name2 - -In this syntax, we create objects with the settings defined above the object -creation. Note that settings are inherited from the top, so in the example -above object2 has inherited the setting for "label1" from the first object. - -For template configurations, the syntax for defining a section is changed -to - [section](options) - label = value - -The options field is used to define templates, refer to templates and hide -templates. Any object can be used as a template. - -No whitespace is allowed between the closing "]" and the parenthesis "(". - -Comments --------- -All lines that starts with semi-colon ";" is treated as comments -and is not parsed. - -The ";--" is a marker for a multi-line comment. Everything after -that marker will be treated as a comment until the end-marker "--;" -is found. Parsing begins directly after the end-marker. - - ;This is a comment - label = value - ;-- This is - a comment --; - - ;-- Comment --; exten=> 1000,1,dial(SIP/lisa) - -Including other files ---------------------- -In all of the configuration files, you may include the content of another -file with the #include statement. The content of the other file will be -included at the row that the #include statement occured. - - #include myusers.conf - -You may also include the output of a program with the #exec directive, -if you enable it in asterisk.conf - -In asterisk.conf, add the execincludes = yes statement in the options -section: - [options] - execincludes=yes - -The exec directive is used like this: - - #exec /usr/local/bin/myasteriskconfigurator.sh - -Adding to an existing section ------------------------------ - - [section] - label = value - - [section](+) - label2 = value2 - -In this case, the plus sign indicates that the second section (with the -same name) is an addition to the first section. The second section can -be in another file (by using the #include statement). If the section -name referred to before the plus is missing, the configuration will fail -to load. - -Defining a template-only section --------------------------------- - [section](!) - label = value - -The exclamation mark indicates to the config parser that this is a only -a template and should not itself be used by the Asterisk module for -configuration. The section can be inherited by other sections (see -section "Using templates" below) but is not used by itself. - -Using templates (or other configuration sections) -------------------------------------------------- - [section](name[,name]) - label = value - -The name within the parenthesis refers to other sections, either -templates or standard sections. The referred sections are included -before the configuration engine parses the local settings within the -section as though their entire contents (and anything they were -previously based upon) were included in the new section. For example -consider the following: - -[foo] -permit=192.168.0.2 -host=asdf -deny=192.168.0.1 - -[bar] -permit=192.168.1.2 -host=jkl -deny=192.168.1.1 - -[baz](foo,bar) -permit=192.168.3.1 -host=bnm - -The [baz] section will be processed as though it had been written in the -following way: - -[baz] -permit=192.168.0.2 -host=asdf -deny=192.168.0.1 -permit=192.168.1.2 -host=jkl -deny=192.168.1.1 -permit=192.168.3.1 -host=bnm - -Additional Examples: --------------------- - -(in top-level sip.conf) - -[defaults](!) -type=friend -nat=yes -qualify=on -dtmfmode=rfc2833 -disallow=all -allow=alaw - -#include accounts/*/sip.conf - -(in accounts/customer1/sip.conf) - -[def-customer1](!,defaults) -secret=this_is_not_secret -context=from-customer1 -callerid=Customer 1 <300> -accountcode=0001 - -[phone1](def-customer1) -mailbox=phone1@customer1 - -[phone2](def-customer1) -mailbox=phone2@customer1 - -This example defines two phones - phone1 and phone2 with settings -inherited from "def-customer1". The "def-customer1" is a template that -inherits from "defaults", which also is a template. - - diff --git a/1.2-netsec/doc/README.cygwin b/1.2-netsec/doc/README.cygwin deleted file mode 100644 index fa1f7845d..000000000 --- a/1.2-netsec/doc/README.cygwin +++ /dev/null @@ -1,7 +0,0 @@ -Cygwin support is completely experimental and usupported at this time. The current state of cygwin support is that it will compile, and start the cli, but will not yet take calls properly. - -To compile with cygwin, you will need at least a standard base cygwin install plus the following packages: - -minires -minires-devel - diff --git a/1.2-netsec/doc/README.dundi b/1.2-netsec/doc/README.dundi deleted file mode 100644 index 89c8d24a1..000000000 --- a/1.2-netsec/doc/README.dundi +++ /dev/null @@ -1,26 +0,0 @@ -Distributed Universal Number Directory (DUNDi) (tm) -=================================================== -http://www.dundi.com -Mark Spencer, Digium, Inc. - -DUNDi is essentially a trusted, peer-to-peer system for being able to -call any phone number from the Internet. DUNDi works by creating a -network of nodes called the "DUNDi E.164 Trust Group" which are bound by -a common peering agreement known as the General Peering Agreement or -GPA. The GPA legally binds the members of the Trust Group to provide -good-faith accurate information to the other nodes on the network, and -provides standards by which the community can insure the integrity of -the information on the nodes themselves. Unlike ENUM or similar -systems, DUNDi is explicitly designed to preclude any necessity for a -single centralized system which could be a source of fees, regulation, -etc. - -You can find the PEERING agreement in the doc directory. - -Much less dramatically, DUNDi can also be used within a private -enterprise to share a dialplan efficiently between multiple nodes, -without incuring a risk of a single point of failure. In this way, -administrators can locally add extensions which become immediately -available to the other nodes in the system. - -For more information visit http://www.dundi.com diff --git a/1.2-netsec/doc/README.enum b/1.2-netsec/doc/README.enum deleted file mode 100644 index ba8484749..000000000 --- a/1.2-netsec/doc/README.enum +++ /dev/null @@ -1,306 +0,0 @@ -README.enum - -2005-09-06 -jtodd@loligo.com - -The ENUMLOOKUP function is more complex than it first may appear, and -this guide is to give a general overview and set of examples that may -be well-suited for the advanced user to evaluate in their -consideration of ENUM or ENUM-like lookup strategies. This document -assumes a familiarity with ENUM (RFC3761) or ENUM-like methods, as -well as familiarity with NAPTR DNS records (RFC2915, RFC3401-3404). -For an overview of NAPTR records, and the use of NAPTRs in the ENUM -global phone-number-to-DNS mapping scheme, please see -http://www.voip-info.org/tiki-index.php?page=ENUM for more detail. - -Using ENUM within Asterisk can be simple or complex, depending on how -many failover methods and redundancy procedures you wish to utilize. -Implementation of ENUM paths is supposedly defined by the person -creating the NAPTR records, but the local administrator may choose to -ignore certain NAPTR response methods (URI types) or prefer some over -others, which is in contradiction to the RFC. The ENUMLOOKUP method -simply provides administrators a method for determining NAPTR results -in either the globally unique ENUM (e164.arpa) DNS tree, or in other -ENUM-like DNS trees which are not globally unique. The methods to -actually create channels ("dial") results given by the ENUMLOOKUP -function is then up to the administrator to implement in a way that -best suits their environment. - -Function: ENUMLOOKUP(<number>[,pointer_type[,options[,zone_suffix]]]) - - Performs an ENUM tree lookup on the specified number, method type, - and (optionally) ordinal offset, and returns one of four different values: - - 1) post-parsed NAPTR of one method (URI) type - 2) count of elements of one method (URI) type - 3) count of all method types - 4) full URI of method at a particular point in the list of all possible methods - -Arguments: - -number = telephone number or search string. Only numeric values -within this string are parsed; all other digits are ignored for -search, but are re-written during NAPTR regexp expansion. - -service_type = tel, sip, h323, iax2, mailto, ...[any other string], - ALL. Default type is "sip". - Special name of "ALL" will create a list of method types across - all NAPTR records for the search number, and then put the results - in an ordinal list starting with 1. The position <number> - specified will then be returned, starting with 1 as the first - record (lowest value) in the list. The service types are not - hardcoded in Asterisk except for the default (sip) if no other - service type specified; any method type string (IANA-approved or - not) may be used except for the string "ALL". - -options = optional specifiers. - c = count. Returns the number of records of this type are returned - (regardless of order or priority.) If "ALL" is the specified - service_type, then a count of all methods will be returned for the - DNS record. - <integer> = The record in priority/order sequence based on the - total count of records passed back by the query. If a service_type - is specified, all entries of that type will be sorted into an - ordinal list starting with 1 (by order first, then priority). - The default of <options> is "1" - -zone_suffix = allows customization of the ENUM zone. Default is e164.arpa. - - -EXAMPLE USES: - -Let's use this ENUM list as an example (note that these examples exist -in the DNS, and will hopefully remain in place as example -destinations, but they may change or become invalid over time. The -end result URIs are not guaranteed to actually work, since some of -these hostnames or SIP proxies are imaginary. Of course, the tel: -replies go to directory assistance for New York City and San -Francisco...) Also note that the complex SIP NAPTR at weight 30 will -strip off the leading "+" from the dialed string if it exists. This -is probably a better NAPTR than hard-coding the number into the NAPTR, -and it is included as a more complex regexp example, though other -simpler NAPTRs will work just as well. - - -0.2.0.1.1.6.5.1.0.3.1.loligo.com. 3600 IN NAPTR 10 100 "u" "E2U+tel" "!^\\+13015611020$!tel:+12125551212!" . -0.2.0.1.1.6.5.1.0.3.1.loligo.com. 3600 IN NAPTR 21 100 "u" "E2U+tel" "!^\\+13015611020$!tel:+14155551212!" . -0.2.0.1.1.6.5.1.0.3.1.loligo.com. 3600 IN NAPTR 25 100 "u" "E2U+sip" "!^\\+13015611020$!sip:2203@sip.fox-den.com!" . -0.2.0.1.1.6.5.1.0.3.1.loligo.com. 3600 IN NAPTR 26 100 "u" "E2U+sip" "!^\\+13015611020$!sip:1234@sip-2.fox-den.com!" . -0.2.0.1.1.6.5.1.0.3.1.loligo.com. 3600 IN NAPTR 30 100 "u" "E2U+sip" "!^\\+*([^\\*]*)!sip:\\1@sip-3.fox-den.com!" . -0.2.0.1.1.6.5.1.0.3.1.loligo.com. 3600 IN NAPTR 55 100 "u" "E2U+mailto" "!^\\+13015611020$!mailto:jtodd@fox-den.com!" . - -Example 1: Simplest case, using first SIP return (use all defaults -except for domain name) -exten => 100,1,Set(foo=${ENUMLOOKUP(+13015611020,,,loligo.com)}) - returns: ${foo}="2203@sip.fox-den.com" - -Example 2: What is the first "tel" pointer type for this number? -(after sorting by order/preference; default of "1" is assumed in -options field) -exten => 100,1,Set(foo=${ENUMLOOKUP(+13015611020,tel,,loligo.com)}) - returns: ${foo}="+12125551212" - -Example 3: How many "sip" pointer type entries are there for this number? -exten => 100,1,Set(foo=${ENUMLOOKUP(+13015611020,sip,c,loligo.com)}) - returns: ${foo}=3 - -Example 4: For all the "tel" pointer type entries, what is the second -one in the list? (after sorting by preference) -exten => 100,1,Set(foo=${ENUMLOOKUP(+13015611020,tel,2,loligo.com)}) - returns: ${foo}="+14155551212" - -Example 5: How many NAPTRs (tel, sip, mailto, etc.) are in the list for this number? -exten => 100,1,Set(foo=${ENUMLOOKUP(+13015611020,ALL,c,loligo.com)}) - returns: ${foo}=6 - -Example 6: Give back the second full URI in the sorted list of all NAPTR URIs: -exten => 100,1,Set(foo=${ENUMLOOKUP(+13015611020,ALL,2,loligo.com)}) - returns: ${foo}="tel:+14155551212" [note the "tel:" prefix in the string] - -Example 7: Look up first SIP entry for the number in the e164.arpa zone (all defaults) -exten => 100,1,Set(foo=${ENUMLOOKUP(+437203001721)}) - returns: ${foo}="enum-test@sip.nemox.net" [note: this result is - subject to change as it is "live" DNS and not under my control] - - -Example 8: Look up the ISN mapping in freenum.org alpha test zone -exten => 100,1,Set(foo=${ENUMLOOKUP(1234*256,,,freenum.org)}) - returns: ${foo}="1234@204.91.156.10" [note: this result is subject - to change as it is "live" DNS] - -Example 9: Give back the first SIP pointer for a number in the -enum.yoydynelabs.com zone (invalid lookup) -exten => 100,1,Set(foo=${ENUMLOOKUP(1234567890,sip,1,enum.yoyodynelabs.com)}) - returns: ${foo}="" - - -Usage notes and subtle features: - - a) The use of "+" in lookups is confusing, and warrants further - explanation. All E.164 numbers ("global phone numbers") by - definition need a leading "+" during ENUM lookup. If you neglect to - add a leading "+", you may discover that numbers that seem to exist - in the DNS aren't getting matched by the system or are returned with - a null string result. This is due to the NAPTR reply requiring a - "+" in the regular expression matching sequence. Older versions of - Asterisk add a "+" from within the code, which may confuse - administrators converting to the new function. Please ensure that - all ENUM (e164.arpa) lookups contain a leading "+" before lookup, so - ensure your lookup includes the leading plus sign. Other DNS trees - may or may not require a leading "+" - check before using those - trees, as it is possible the parsed NAPTRs will not provide correct - results unless you have the correct dialed string. If you get - console messages like "WARNING[24907]: enum.c:222 parse_naptr: NAPTR - Regex match failed." then it is very possible that the returned - NAPTR expects a leading "+" in the search string (or the returned - NAPTR is mis-formed.) - - b) If a query is performed of type "c" ("count") and let's say you - get back 5 records and then some seconds later a query is made - against record 5 in the list, it may not be the case that the DNS - resolver has the same answers as it did a second or two ago - maybe - there are only 4 records in the list in the newest query. The - resolver should be the canonical storage location for DNS records, - since that is the intent of ENUM. However, some obscure future - cases may have wildly changing NAPTR records within several seconds. - This is a corner case, and probably only worth noting as a very rare - circumstance. (note: I do not object to Asterisk's dnsmgr method of - locally caching DNS replies, but this method needs to honor the TTL - given by the remote zone master. Currently, the ENUMLOOKUP function - does not use the dnsmgr method of caching local DNS replies.) - - c) If you want strict NAPTR value ordering, then it will be - necessary to use the "ALL" method to incrementally step through the - different returned NAPTR pointers. You will need to use string - manipulation to strip off the returned method types, since the - results will look like "sip:12125551212" in the returned value. - This is a non-trivial task, though it is required in order to have - strict RFC compliance and to comply with the desires of the remote - party who is presenting NAPTRs in a particular order for a reason. - - d) Default behavior for the function (even in event of an error) is - to move to the next priority, and the result is a null value. Most - ENUM lookups are going to be failures, and it is the responsibility - of the dialplan administrator to manage error conditions within - their dialplan. This is a change from the old app_enumlookup method - and it's arbitrary priority jumping based on result type or failure. - - e) Anything other than digits will be ignored in lookup strings. - Example: a search string of "+4372030blah01721" will turn into - 1.2.7.1.0.0.3.0.2.7.3.4.e164.arpa. for the lookup. The NAPTR - parsing may cause unexpected results if there are strings inside - your NAPTR lookups. - - f) If there exist multiple records with the same weight and order as - a result of your query, the function will RANDOMLY select a single - NAPTR from those equal results. - - g) Currently, the function ignores the settings in enum.conf as the - search zone name is now specified within the function, and the H323 - driver can be chosen by the user via the dialplan. There were no - other values in this file, and so it becomes deprecated. - - h) The function will digest and return NAPTRs which use older - (depricated) style, reversed method strings such as "sip+E2U" - instead of the more modern "E2U+sip" - - i) There is no provision for multi-part methods at this time. If - there are multiple NAPTRs with (as an example) a method of - "E2U+voice:sip" and then another NAPTR in the same DNS record with a - method of ""E2U+sip", the system will treat these both as method - "sip" and they will be separate records from the perspective of the - function. Of course, if both records point to the same URI and have - equal priority/weight (as is often the case) then this will cause no - serious difficulty, but it bears mentioning. - - j) ISN (ITAD Subscriber Number) usage: If the search number is of - the form ABC*DEF (where ABC and DEF are at least one numeric digit) - then perform an ISN-style lookup where the lookup is manipulated to - C.B.A.DEF.domain.tld (all other settings and options apply.) See - http://www.freenum.org/ for more details on ISN lookups. In the - unlikely event you wish to avoid ISN re-writes, put an "n" as the - first digit of the search string - the "n" will be ignored for the search. - - -==EXAMPLES== - -All examples below except where noted use "e164.arpa" as the -referenced domain, which is the default domain name for ENUMLOOKUP. -All numbers are assumed to not have a leading "+" as dialed by the -inbound channel, so that character is added where necessary during -ENUMLOOKUP function calls. - -; example 1 -; -; Assumes North American international dialing (011) prefix. -; Look up the first SIP result and send the call there, otherwise -; send the call out a PRI. This is the most simple possible -; ENUM example, but only uses the first SIP reply in the list of -; NAPTR(s). -; -exten => _011.,1,Set(enumresult=${ENUMLOOKUP(+${EXTEN:3})}) -exten => _011.,n,Dial(SIP/${enumlookup}) -exten => _011.,n,Dial(Zap/g1/${EXTEN}) -; -; end example 1 - -; example 2 -; -; Assumes North American international dialing (011) prefix. -; Check to see if there are multiple SIP NAPTRs returned by -; the lookup, and dial each in order. If none work (or none -; exist) then send the call out a PRI, group 1. -; -exten => _011.,1,Set(sipcount=${ENUMLOOKUP(${EXTEN:3},sip,c)}|counter=0) -exten => _011.,n,While($["${counter}"<"${sipcount}"]) -exten => _011.,n,Set(counter=$[${counter}+1]) -exten => _011.,n,Dial(SIP/${ENUMLOOKUP(+${EXTEN:3},sip,${counter})}) -exten => _011.,n,EndWhile -exten => _011.,n,Dial(Zap/g1/${EXTEN}) -; -; end example 2 - -; example 3 -; -; This example expects an ${EXTEN} that is an e.164 number (like -; 14102241145 or 437203001721) -; Search through e164.arpa and then also search through e164.org -; to see if there are any valid SIP or IAX termination capabilities. -; If none, send call out via Zap channel 1. -; -; Start first with e164.arpa zone... -; -exten => _X.,1,Set(sipcount=${ENUMLOOKUP(+${EXTEN},sip,c)}|counter=0) -exten => _X.,2,GotoIf($["${counter}"<"${sipcount}"]?3:6) -exten => _X.,3,Set(counter=$[${counter}+1]) -exten => _X.,4,Dial(SIP/${ENUMLOOKUP(+${EXTEN},sip,${counter})}) -exten => _X.,5,GotoIf($["${counter}"<"${sipcount}"]?3:6) -; -exten => _X.,6,Set(iaxcount=${ENUMLOOKUP(+${EXTEN},iax2,c)}|counter=0) -exten => _X.,7,GotoIf($["${counter}"<"${iaxcount}"]?8:11) -exten => _X.,8,Set(counter=$[${counter}+1]) -exten => _X.,9,Dial(IAX2/${ENUMLOOKUP(+${EXTEN},iax2,${counter})}) -exten => _X.,10,GotoIf($["${counter}"<"${iaxcount}"]?8:11) -; -exten => _X.,11,NoOp("No valid entries in e164.arpa for ${EXTEN} - checking in e164.org") -; -; ...then also try e164.org, and look for SIP and IAX NAPTRs... -; -exten => _X.,12,Set(sipcount=${ENUMLOOKUP(+${EXTEN},sip,c,e164.org)}|counter=0) -exten => _X.,13,GotoIf($["${counter}"<"${sipcount}"]?14:17) -exten => _X.,14,Set(counter=$[${counter}+1]) -exten => _X.,15,Dial(SIP/${ENUMLOOKUP(+${EXTEN},sip,${counter},e164.org)}) -exten => _X.,16,GotoIf($["${counter}"<"${sipcount}"]?14:17) -; -exten => _X.,17,Set(iaxcount=${ENUMLOOKUP(+${EXTEN},iax2,c,e164.org)}|counter=0) -exten => _X.,18,GotoIf($["${counter}"<"${iaxcount}"]?19:22) -exten => _X.,19,Set(counter=$[${counter}+1]) -exten => _X.,20,Dial(IAX2/${ENUMLOOKUP(+${EXTEN},iax2,${counter},e164.org)}) -exten => _X.,21,GotoIf($["${counter}"<"${iaxcount}"]?19:22) -; -; ...then send out PRI. -; -exten => _X.,22,NoOp("No valid entries in e164.org for ${EXTEN} - sending out via Zap") -exten => _X.,23,Dial(Zap/g1/${EXTEN}) -; -; end example 3 diff --git a/1.2-netsec/doc/README.extconfig b/1.2-netsec/doc/README.extconfig deleted file mode 100644 index 6b147e573..000000000 --- a/1.2-netsec/doc/README.extconfig +++ /dev/null @@ -1,85 +0,0 @@ -Asterisk external configuration -=============================== - -The Asterisk external configuration engine is the result of work by -Anthony Minessale II, Mark Spencer and Constantine Filin. - -It is designed to provide a flexible, seamless integration between -Asterisk's internal configuration structure and external SQL other other -databases (maybe even LDAP one day). - -The external configuration engine is the basis for the ARA, the -Asterisk Realtime Architecture (see README.realtime for more -information). - -* Configuration - -External configuration is configured in /etc/asterisk/extconfig.conf -allowing you to map any configuration file (static mappings) to -be pulled from the database, or to map special runtime entries which -permit the dynamic creation of objects, entities, peers, etc. without -the necessity of a reload. - -Generally speaking, the columns in your tables should line up with the -fields you would specify in the given entity declaration. If an entry -would appear more than once, in the column it should be separated by a -semicolon. For example, an entity that looks like: - -[foo] -host=dynamic -secret=bar -context=default -context=local - -could be stored in a table like this: - -+------+--------+-------+--------------+----------+-----+-----------+ -| name | host | secret| context | ipaddr | port| regseconds| -+------+--------+-------+--------------+----------+-----+-----------+ -| foo | dynamic| bar | default;local| 127.0.0.1| 4569| 1096954152| -+------+--------+-------+--------------+----------+-----+-----------+ - -Note that for use with IAX or SIP, the table will also need the "name", -"ipaddr", "port", "regseconds" columns. If you wanted to be able to -configure the callerid, you could just add a callerid column to the -table, for example. - -A SIP table would look more like this: - -+------+--------+-------+----------+-----+------------+----------+ -| name | host | secret| ipaddr | port| regseconds | username | -+------+--------+-------+----------+-----+------------+----------+ -| foo | dynamic| bar | 127.0.0.1| 4569| 1096954152 | 1234 | -+------+--------+-------+----------+-----+------------+----------+ - -in order to store appropriate parameters required for SIP. - -A Voicemail table would look more like this: - -+----------+---------+----------+----------+-----------+---------------+ -| uniqueid | mailbox | context | password |email | fullname | -+----------+---------+----------+----------+-----------+---------------+ -| 1 | 1234 | default | 4242 | a@b.com | Joe Schmoe | -+----------+---------+----------+----------+-----------+---------------+ - -The uniqueid should be unique to each voicemail user and can be -autoincrement. It need not have any relation to the mailbox or context. - -An extension table would look more like this: - -+----------+---------+----------+-------+-----------+ -| context | exten | priority | app | appdata | -+----------+---------+----------+-------+-----------+ -| default | 1234 | 1 | Dial | Zap/1 | -+----------+---------+----------+-------+-----------+ - -In the dialplan you just use the Realtime switch: - -[foo] -switch => Realtime - -or: - -[bar] -switch => Realtime/bar@extensions - diff --git a/1.2-netsec/doc/README.externalivr b/1.2-netsec/doc/README.externalivr deleted file mode 100644 index a1d4757e7..000000000 --- a/1.2-netsec/doc/README.externalivr +++ /dev/null @@ -1,109 +0,0 @@ -Asterisk External IVR Interface -------------------------------- - -If you load app_externalivr.so in your Asterisk instance, you will -have an ExternalIVR() application available in your dialplan. This -application implements a simple protocol for bidirectional -communication with an external process, while simultaneous playing -audio files to the connected channel (without interruption or -blocking). - -The arguments to ExternalIVR() consist of the command to execute and -any arguments to pass to it, the same as the System() application -accepts. The external command will be executed in a child process, -with its standard file handles connected to the Asterisk process as -follows: - -stdin (0) - DTMF and hangup events will be received on this handle -stdout (1) - Playback and hangup commands can be sent on this handle -stderr (2) - Error messages can be sent on this handle - -The application will also create an audio generator to play audio to -the channel, and will start playing silence. When your application -wants to send audio to the channel, it can send a command (see below) -to add file(s) to the generator's playlist. The generator will then -work its way through the list, playing each file in turn until it -either runs out of files to play, the channel is hung up, or a command -is received to clear the list and start with a new file. At any time, -more files can be added to the list and the generator will play them -in sequence. - -While the generator is playing audio (or silence), any DTMF events -received on the channel will be sent to the child process (see -below). Note that this can happen at any time, since the generator, -the child process and the channel thread are all executing -independently. It is very important that your external application be -ready to receive events from Asterisk at all times (without blocking), -or you could cause the channel to become non-responsive. - -If the child process dies, ExternalIVR() will notice this and hang up -the channel immediately (and also send a message to the log). - -DTMF (and other) events ------------------------ - -All events will be newline-terminated strings. - -Events send to the child's stdin will be in the following format: - -tag,timestamp[,data] - -The tag can be one of the following characters: - -0-9: DTMF event for keys 0 through 9 -A-D: DTMF event for keys A through D -*: DTMF event for key * -#: DTMF event for key # -H: the channel was hung up by the connected party -Z: the previous command was unable to be executed (file does not -exist, etc.) -T: the play list was interrupted (see below) -D: a file was dropped from the play list due to interruption (the -data element will be the dropped file name) -F: a file has finished playing (the data element will be the file -name) - -The timestamp will be 10 digits long, and will be a decimal -representation of a standard Unix epoch-based timestamp. - -Commands --------- - -All commands must be newline-terminated strings. - -The child process can send commands on stdout in the following formats: - -S,filename -A,filename -H,message -O,option - -The 'S' command checks to see if there is a playable audio file with -the specified name, and if so, clear's the generator's playlist and -places the file onto the list. Note that the playability check does -not take into account transcoding requirements, so it is possible for -the file to not be played even though it was found. If the file cannot -be found, a 'Z' event (see above) will be sent to the child. If the -generator is not currently playing silence, then T and D events will -be sent to the child to signal the playlist interruption and notify -it of the files that will not be played. - -The 'A' command checks to see if there is a playable audio file with -the specified name, and if so, adds it to the generator's -playlist. The same playability and exception rules apply as for the -'S' command. - -The 'H' command stops the generator and hangs up the channel, and logs -the supplied message to the Asterisk log. - -The 'O' command allows the child to set/clear options in the -ExternalIVR() application. The supported options are: - autoclear/noautoclear: - Automatically interrupt and clear the playlist upon reception - of DTMF input. - -Errors ------- - -Any newline-terminated output generated by the child process on its -stderr handle will be copied into the Asterisk log. diff --git a/1.2-netsec/doc/README.h323 b/1.2-netsec/doc/README.h323 deleted file mode 100644 index 9951f6353..000000000 --- a/1.2-netsec/doc/README.h323 +++ /dev/null @@ -1,23 +0,0 @@ -The Asterisk PBX supports H.323 via two totally separate -channel drivers. - -You can find more information Asterisk's native H.323 -support in /path/to/asterisk/channels/h323/README or -you can download a third party driver at -http://www.inaccessnetworks.com/projects/asterisk-oh323 - -Asterisk's native H.323 is supported and maintained by -Jeremy McNamara (JerJer in irc). Support for the third -party H.323 driver is supplied by inAccessNetworks. - -If you have trouble with either driver you should direct -your debug and comments to the appropriate party, making -sure to be specific in exactly which H.323 driver you are -running. - -Please, read all supplied documentation before contacting -either party for support. Many issues can be quickly -resolved by simply following the instructions that are -provided. - - diff --git a/1.2-netsec/doc/README.iax b/1.2-netsec/doc/README.iax deleted file mode 100644 index 1a35d6b15..000000000 --- a/1.2-netsec/doc/README.iax +++ /dev/null @@ -1,369 +0,0 @@ -Inter-Asterisk eXchange Protocol -================================ - -INTRODUCTION ------------- - -This document is intended as an introduction to the Inter-Asterisk -eXchange (or simply IAX) protocol. It provides both a theoretical -background and practical information on its use. - -WHY IAX -------- -The first question most people are thinking at this point is "Why do you -need another VoIP protocol? Why didn't you just use SIP or H.323?" - -Well, the answer is a fairly complicated one, but in a nutshell it's like -this... Asterisk is intended as a very flexible and powerful -communications tool. As such, the primary feature we need from a VoIP -protocol is the ability to meet our own goals with Asterisk, and one with -enough flexibility that we could use it as a kind of laboratory for -inventing and implementing new concepts in the field. Neither H.323 or -SIP fit the roles we needed, so we developed our own protocol, which, -while not standards based, provides a number of advantages over both SIP -and H.323, some of which are: - - * Interoperability with NAT/PAT/Masquerade firewalls - IAX seamlessly interoperates through all sorts of NAT and PAT - and other firewalls, including the ability to place and - receive calls, and transfer calls to other stations. - - * High performance, low overhead protocol - When running on low-bandwidth connections, or when running - large numbers of calls, optimized bandwidth utilization is - imperative. IAX uses only 4 bytes of overhead - - * Internationalization support - IAX transmits language information, so that remote PBX - content can be delivered in the native language of the - calling party. - - * Remote dialplan polling - IAX allows a PBX or IP phone to poll the availability of a - number from a remote server. This allows PBX dialplans to - be centralized. - - * Flexible authentication - IAX supports cleartext, md5, and RSA authentication, - providing flexible security models for outgoing calls and - registration services. - - * Multimedia protocol - IAX supports the transmission of voice, video, images, text, - HTML, DTMF, and URL's. Voice menus can be presented in both - audibly and visually. - - * Call statistic gathering - IAX gathers statistics about network performance (including - latency and jitter, as well as providing end-to-end latency - measurement. - - * Call parameter communication - Caller*ID, requested extension, requested context, etc are - all communicated through the call. - - * Single socket design - IAX's single socket design allows up to 32768 calls to be - multiplexed. - -While we value the importance of standards based (i.e. SIP) call handling, -hopefully this will provide a reasonable explanation of why we developed -IAX rather than starting with SIP. - -CONFIG FILE CONVENTIONS ------------------------ -Lines beginning with '>' represent lines which might appear in an actual -configuration file. The '>' is used to help separate them from the -descriptive text and should not actually be included in the file itself. - -Lines within []'s by themselves represent section labels within the -configuration file. like this: - -> [mysection] - -Options are set using the "=" sign, for example - -> myoption = value - -Sometimes an option will have a number of discrete values which it can -take. In that case, in the documentation, the options will be listed -within square brackets (the "[" and "]" ones) separated by the pipe symbol -("|"). For example: - -> myoption = [value1|value2|value3] - -means the option "myoption" can be assigned a value of "value1", "value2", -or "value3". - -Objects, or pseudo-objects are instantiated using the "=>" construct. For -example: - -> myobject => parameter - -creates an object called "myobject" with some parameter whose definition -would be specific to that object. Note that the config file parser -considers "=>" and "=" to be equivalent and their use is purely to make -configuration files more readable and easier to "humanly parse". - -The comment character in Asterisk configuration files is the semicolon -";". The reason it is not "#" is because the "#" symbol can be used as -parts of extensions and it didn't seem like a good idea to have to escape -it. - -IAX CONFIGURATION IN ASTERISK ------------------------------ - -Like everything else in Asterisk, IAX's configuration lies in -/etc/asterisk -- specifically /etc/asterisk/iax.conf - -The IAX configuration file is a collection of sections, each of which -(with the exception of the "general" section) represents an entity within -the IAX scope. - ------------- - -The first section is typically the "general" section. In this area, -a number of parameters which affect the entire system are configured. -Specifically, the default codecs, port and address, jitter behavior, TOS -bits, and registrations. - -The first line of the "general" section is always: - -> [general] - -Following the first line are a number of other possibilities: - -> bindport = <portnum> - -This sets the port that IAX will bind to. The default IAX version 1 -port number is 5036. For IAX version 2, that is now the default in -Asterisk, the default port is 4569. -It is recommended that this value not be altered in general. - -> bindaddr = <ipaddr> - -This allows you to bind IAX to a specific local IP address instead of -binding to all addresses. This could be used to enhance security if, for -example, you only wanted IAX to be available to users on your LAN. - -> bandwidth = [low|medium|high] - -The bandwidth selection initializes the codec selection to appropriate -values for given bandwidths. The "high" selection enables all codecs and -is recommended only for 10Mbps or higher connections. The "medium" -bandwidth eliminates signed linear, Mu-law and A-law codecs, leaving only -the codecs which are 32kbps and smaller (with MP3 as a special case). It -can be used with broadband connections if desired. "low" eliminates ADPCM -and MP3 formats, leaving only the G.723.1, GSM, and LPC10. - -> allow = [gsm|lpc10|g723.1|adpcm|ulaw|alaw|mp3|slinear|all] -> disallow = [gsm|lpc10|g723.1|adpcm|ulaw|alaw|mp3|slinear|all] - -The "allow" and "disallow" allow you to fine tune the codec selection -beyond the initial bandwidth selection on a codec-by-codec basis. - -The recommended configuration is to select "low" bandwidth and then -disallow the LPC10 codec just because it doesn't sound very good. - -> jitterbuffer = [yes|no] -> dropcount = <dropamount> -> maxjitterbuffer = <max> -> maxexcessbuffer = <max> - -These parameters control the operation of the jitter buffer. The -jitterbuffer should always be enabled unless you expect all your -connections to be over a LAN. -* drop count is the maximum number of voice packets to allow to drop - (out of 100). Useful values are 3-10. -* maxjitterbuffer is the maximum amount of jitter buffer to permit to be - used. -* maxexcessbuffer is the maximum amount of excess jitter buffer - that is permitted before the jitter buffer is slowly shrunk to eliminate - latency. -* minexcessbuffer is the minimum amout of excess jitter buffer - -> accountcode = <code> -> amaflags = [default|omit|billing|documentation] - -These parameters affect call detail record generation. The first sets the -account code for records received with IAX. The account code can be -overridden on a per-user basis for incoming calls (see below). The -amaflags controls how the record is labeled ("omit" causes no record to be -written. "billing" and "documentation" label the records as billing or -documentation records respectively, and "default" selects the system -default. - -> tos = [lowdelay|throughput|reliability|mincost|none] - -IAX can optionally set the TOS (Type of Service) bits to specified values -to help improve performance in routing. The recommended value is -"lowdelay", which many routers (including any Linux routers with 2.4 -kernels that have not been altered with ip tables) will give priority to -these packets, improving voice quality. - -> register => <name>[:<secret>]@<host>[:port] - -Any number of registry entries may be instantiated in the general -section. Registration allows Asterisk to notify a remote Asterisk server -(with a fixed address) what our current address is. In order for -registration to work, the remote Asterisk server will need to have a -dynamic peer entry with the same name (and secret if provided). - -The name is a required field, and is the remote peer name that we wish to -identify ourselves as. A secret may be provided as well. The secret is -generally a shared password between the local server and the remote -server. However, if the secret is in square brackets ([]'s) then it is -interpreted as the name of a RSA key to use. In that case, the local Asterisk -server must have the *private* key (/var/lib/asterisk/keys/<name>.key) and -the remote server will have to have the corresponding public key. - -The "host" is a required field and is the hostname or IP address of the -remote Asterisk server. The port specification is optional and is by -default 4569 for iax2 if not specified. - -> notransfer = yes | no - -If an IAX phone calls another IAX phone by using a Asterisk server, -Asterisk will transfer the call to go peer to peer. If you do not -want this, turn on notransfer with a "yes". This is also settable -for peers and users. - -------------- - -The following sections, after "general" define either users, peers or -friends. A "user" is someone who connects to us. A "peer" is someone -that we connect to. A "friend" is simply shorthand for creating a "user" -and "peer" with identical parameters (i.e. someone who can contact us and -who we contact). - -> [identifier] - -The section begins with the identifier in square brackets. The identifier -should be an alphanumeric string. - -> type = [user|peer|friend] - -This line tells Asterisk how to interpret this entity. Users are things -that connect to us, while peers are phones we connect to, and a friend is -shorthand for creating a user and a peer with identical information - ----------------- -User fields: - -> context = <context> - -One or more context lines may be specified in a user, thus giving the user -access to place calls in the given contexts. Contexts are used by -Asterisk to divide dialing plans into logical units each with the ability -to have numbers interpreted differently, have their own security model, -auxiliary switch handling, and include other contexts. Most users are -given access to the default context. Trusted users could be given access -to the local context for example. - -> permit = <ipaddr>/<netmask> -> deny = <ipaddr>/<netmask> - -Permit and deny rules may be applied to users, allowing them to connect -from certain IP addresses and not others. The permit and deny rules are -interpreted in sequence and all are evaluated on a given IP address, with -the final result being the decision. For example: - -> permit = 0.0.0.0/0.0.0.0 -> deny = 192.168.0.0/255.255.255.0 - -would deny anyone in 192.168.0.0 with a netmask of 24 bits (class C), -whereas: - -> deny = 192.168.0.0/24 -> permit = 0.0.0.0/0 - -would not deny anyone since the final rule would permit anyone, thus -overriding the denial. - -If no permit/deny rules are listed, it is assumed that someone may connect -from anywhere. - -> callerid = <callerid> - -You may override the Caller*ID information passed by a user to you (if -they choose to send it) in order that it always be accurate from the -perspective of your server. - -> auth = [md5|plaintext|rsa] - -You may select which authentication methods are permitted to be used by -the user to authenticate to us. Multiple methods may be specified, -separated by commas. If md5 or plaintext authentication is selected, a -secret must be provided. If RSA authentication is specified, then one or -more key names must be specified with "inkeys" - -If no secret is specified and no authentication method is specified, then -no authentication will be required. - -> secret = <secret> - -The "secret" line specifies the shared secret for md5 and plaintext -authentication methods. It is never suggested to use plaintext except in -some cases for debugging. - -> inkeys = key1[:key2...] - -The "inkeys" line specifies which keys we can use to authenticate the -remote peer. If the peer's challenge passes with any of the given keys, -then we accept its authentication. The key files live in -/var/lib/asterisk/keys/<name>.pub and are *public keys*. Public keys are -not typically DES3 encrypted and thus do not usually need initialization. - ---------------- -Peer configuration - -> allow = [gsm|lpc10|g723.1|adpcm|ulaw|alaw|mp3|slinear|all] -> disallow = [gsm|lpc10|g723.1|adpcm|ulaw|alaw|mp3|slinear|all] - -The "allow" and "disallow" may be used to enable or disable specific codec -support on a per-peer basis. - -> host = [<ipaddr>|dynamic] - -The host line specifies the hostname or IP address of the remote host, or -may be the word "dynamic" signifying that the host will register with us -(see register => in the general section above). - -> defaultip = <ipaddr> - -If the host uses dynamic registration, Asterisk may still be given a -default IP address to use when dynamic registration has not been performed -or has timed out. - -> peercontext = <context> - -Specifies the context name to be passed to the peer for it to use when routing -the call through its dial plan. This entry will be used only if a context -is not included in the IAX2 channel name passed to the Dial command. - -> qualify = [yes | no | <value>] - -Qualify turns on checking of availability of the remote peer. If the -peer becomes unavailable, no calls are placed to the peer until -it is reachable again. This is also helpful in certain NAT situations. - -> jitterbuffer = [yes | no] - -Turns on or off the jitterbuffer for this peer - -> mailbox = <mailbox>[@mailboxcontext] - -Specifies a mailbox to check for voicemail notification. - -> permit = <ipaddr>/<netmask> -> deny = <ipaddr>/<netmask> - -Permit and deny rules may be applied to users, allowing them to connect -from certain IP addresses and not others. The permit and deny rules are -interpreted in sequence and all are evaluated on a given IP address, with -the final result being the decision. See the user section above -for examples. - ----------------------------------------------------------------------- -For more examples of a configuration, please see the iax.conf.sample in -your the /configs directory of you source code distribution diff --git a/1.2-netsec/doc/README.ices b/1.2-netsec/doc/README.ices deleted file mode 100644 index d75236357..000000000 --- a/1.2-netsec/doc/README.ices +++ /dev/null @@ -1,12 +0,0 @@ -Icecast + Asterisk -================== -The advent of icecast into Asterisk allows you to do neat things like have -a caller stream right into an ice-cast stream as well as using chan_local -to place things like conferences, music on hold, etc. into the stream. - -You'll need to specify a config file for the ices encoder. An example is -included in contrib/asterisk-ices.xml - -Anyway hope you like it. - -Mark diff --git a/1.2-netsec/doc/README.jitterbuffer b/1.2-netsec/doc/README.jitterbuffer deleted file mode 100644 index e5cd81ce0..000000000 --- a/1.2-netsec/doc/README.jitterbuffer +++ /dev/null @@ -1,137 +0,0 @@ -The new Jitterbuffer in Asterisk --------------------------------- -Steve Kann - - - -The new jitterbuffer, PLC, and the IAX2-integration of the new jitterbuffer -have been integrated into Asterisk. The jitterbuffer is generic and work is -going on to implement it in SIP/RTP as well. - -Also, we've added a feature called "trunktimestamps", which adds individual -timestamps to trunked frames within a trunk frame. - -Here's how to use this stuff: - -1) The new jitterbuffer: ------------------------- -You must add "jitterbuffer=yes" to either the [general] part of -iax.conf, or to a peer or a user. (just like the old jitterbuffer). -Also, you can set "maxjitterbuffer=n", which puts a hard-limit on the size of the -jitterbuffer of "n milliseconds". It is not necessary to have the new jitterbuffer -on both sides of a call; it works on the receive side only. - -2) PLC: -------- -The new jitterbuffer detects packet loss. PLC is done to try to recreate these -lost packets in the codec decoding stage, as the encoded audio is translated to slinear. -PLC is also used to mask jitterbuffer growth. - -This facility is enabled by default in iLBC and speex, as it has no additional cost. -This facility can be enabled in adpcm, alaw, g726, gsm, lpc10, and ulaw by setting -genericplc => true in the [plc] section of codecs.conf. - -3) Trunktimestamps: -------------------- -To use this, both sides must be using Asterisk v1.2. -Setting "trunktimestamps=yes" in iax.conf will cause your box to send 16-bit timestamps -for each trunked frame inside of a trunk frame. This will enable you to use jitterbuffer -for an IAX2 trunk, something that was not possible in the old architecture. - -The other side must also support this functionality, or else, well, bad things will happen. -If you don't use trunktimestamps, there's lots of ways the jitterbuffer can get confused because -timestamps aren't necessarily sent through the trunk correctly. - -4) Communication with Asterisk v1.0.x systems ---------------------------------------------- -You can set up communication with v1.0.x systems with the new jitterbuffer, but -you can't use trunks with trunktimestamps in this communication. - -If you are connecting to an Asterisk server with earlier versions of the software (1.0.x), -do not enable both jitterbuffer and trunking for the involved peers/users -in order to be able to communicate. Earlier systems will not support trunktimestamps. - -You may also compile chan_iax2.c without the new jitterbuffer, enabling the old -backwards compatible architecture. Look in the source code for instructions. - - -5) Testing and monitoring: --------------------------- -You can test the effectiveness of PLC and the new jitterbuffer's detection of loss by using -the new CLI command "iax2 test losspct <n>". This will simulate n percent packet loss -coming _in_ to chan_iax2. You should find that with PLC and the new JB, 10 percent packet -loss should lead to just a tiny amount of distortion, while without PLC, it would lead to -silent gaps in your audio. - -"iax2 show netstats" shows you statistics for each iax2 call you have up. -The columns are "RTT" which is the round-trip time for the last PING, and then a bunch of s -tats for both the local side (what you're receiving), and the remote side (what the other -end is telling us they are seeing). The remote stats may not be complete if the remote -end isn't using the new jitterbuffer. - -The stats shown are: -* Jit: The jitter we have measured (milliseconds) -* Del: The maximum delay imposed by the jitterbuffer (milliseconds) -* Lost: The number of packets we've detected as lost. -* %: The percentage of packets we've detected as lost recently. -* Drop: The number of packets we've purposely dropped (to lower latency). -* OOO: The number of packets we've received out-of-order -* Kpkts: The number of packets we've received / 1000. - -Reporting problems -================== - -There's a couple of things that can make calls sound bad using the jitterbuffer: - -1) The JB and PLC can make your calls sound better, but they can't fix everything. -If you lost 10 frames in a row, it can't possibly fix that. It really can't help much -more than one or two consecutive frames. - -2) Bad timestamps: If whatever is generating timestamps to be sent to you generates -nonsensical timestamps, it can confuse the jitterbuffer. In particular, discontinuities -in timestamps will really upset it: Things like timestamps sequences which go 0, 20, 40, -60, 80, 34000, 34020, 34040, 34060... It's going to think you've got about 34 seconds -of jitter in this case, etc.. -The right solution to this is to find out what's causing the sender to send us such nonsense, -and fix that. But we should also figure out how to make the receiver more robust in -cases like this. - -chan_iax2 will actually help fix this a bit if it's more than 3 seconds or so, but at -some point we should try to think of a better way to detect this kind of thing and -resynchronize. - -Different clock rates are handled very gracefully though; it will actually deal with a -sender sending 20% faster or slower than you expect just fine. - -3) Really strange network delays: If your network "pauses" for like 5 seconds, and then -when it restarts, you are sent some packets that are 5 seconds old, we are going to see -that as a lot of jitter. We already throw away up to the worst 20 frames like this, -though, and the "maxjitterbuffer" parameter should put a limit on what we do in this case. - -Reporting possible bugs ------------------------ -If you do find bad behaviors, here's the information that will help to diagnose this: - -1) Describe - -a) the source of the timestamps and frames: i.e. if they're coming from another chan_iax2 box, -a bridged RTP-based channel, an IAX2 softphone, etc.. - -b) The network between, in brief (i.e. the internet, a local lan, etc). - -c) What is the problem you're seeing. - - -2) Take a look and see what iax2 show netstats is saying about the call, and if it makes sense. - -3) a tcpdump of the frames, (or, tethereal output from), so we can see the timestamps and delivery -times of the frames you're receiving. You can make such a tcpdump with: - -tcpdump -s 2048 -w /tmp/example.dump udp and port 4569 [and host <other-end>] - -Report bugs in the Asterisk bugtracker, http://bugs.digium.com. -Please read the bug guidelines before you post a bug. - -Have fun! - --SteveK diff --git a/1.2-netsec/doc/README.math b/1.2-netsec/doc/README.math deleted file mode 100644 index 7718f9e44..000000000 --- a/1.2-netsec/doc/README.math +++ /dev/null @@ -1,69 +0,0 @@ - -Mathematical dialplan function - -Yeah, I thought it was a little insane too.. - -adds: - -Sum, Multiply, Divide, Subtract, Modulus, GT, LT, GTE, LTE, EQ functions to Asterisk - -All functions follow the same basic pattern for parameters: - -parameter 1 = the math expression -parameter 2 = the type of result - -Perform calculation on number 1 to number 2. Valid ops are: - +,-,/,*,%,<,>,>=,<=,== -and behave as their C equivalents. - -<type_of_result> - wanted type of result: - f, float - float(default) - i, int - integer, - h, hex - hex, - c, char - char - -Each math expression is performed as - - Action param1 on param2 - -eg: - - Action = Divide - Param1 = 10 - Param2 = 2 - -Results in - - Divide 10 by 2 - - -Example dialplan: - -exten => 11099,1,Set(RV=${MATH(1+20)}) -exten => 11099,n,NOOP(${RV}) -exten => 11099,n,Set(RV=${MATH(10*2)}) -exten => 11099,n,NOOP(${RV}) -exten => 11099,n,Set(RV=${MATH(10*2)}) -exten => 11099,n,NOOP(${RV}) -exten => 11099,n,Set(RV=${MATH(10-2)}) -exten => 11099,n,NOOP(${RV}) -exten => 11099,n,Set(RV=${MATH(2%10)}) -exten => 11099,n,NOOP(${RV}) -exten => 11099,n,Set(RV=${MATH(10/0)}) -exten => 11099,n,NOOP(${RV}) -exten => 11099,n,Set(RV=${MATH(10-200)}) -exten => 11099,n,NOOP(${RV}) -exten => 11099,n,Set(RV=${MATH(1-20)}) -exten => 11099,n,NOOP(${RV}) -exten => 11099,n,Set(RV=${MATH(1<20)}) -exten => 11099,n,NOOP(${RV}) -exten => 11099,n,Set(RV=${MATH(1>=20)}) -exten => 11099,n,NOOP(${RV}) -exten => 11099,n,Set(RV=${MATH(101>20)}) -exten => 11099,n,NOOP(${RV}) -exten => 11099,n,Set(RV=${MATH(1==20)}) -exten => 11099,n,NOOP(${RV}) -exten => 11099,n,Set(RV=${MATH(20<=20)}) -exten => 11099,n,NOOP(${RV}) -exten => 11099,n,Set(RV=${MATH(123%16,int)}) -exten => 11099,n,NOOP(${RV}) diff --git a/1.2-netsec/doc/README.misdn b/1.2-netsec/doc/README.misdn deleted file mode 100644 index 652ba70cd..000000000 --- a/1.2-netsec/doc/README.misdn +++ /dev/null @@ -1,342 +0,0 @@ -mISDN Channel Driver for Asterisk PBX -====================================== - - -This package contains the mISDN Channel Driver for the Asterisk PBX. It -supports every mISDN Hardware and provides an interface for asterisk. - -Features: ---------- - -* NT and TE mode -* PP and PMP mode -* BRI and PRI (with BNE1 and BN2E1 Cards) -* DTMF Detection in HW+mISDNdsp (much better than asterisks internal!) -* Display Messages to Phones (which support display msg) -* HOLD/RETRIEVE/TRANSFER on ISDN Phones : ) -* Screen/ Not Screen User Number -* Basic EchoCancellation -* Volume Control -* Crypting with mISDNdsp (Blowfish) -* Data (HDLC) callthrough -* Data Calling (with app_ptyfork +pppd) -* Echo cancellation -* Some other - -Supported Hardware: -------------------- - -chan_misdn supports any mISDN compatible Hardware. - -Overview --------- - -- Fast Installation Guide -- Pre-Requisites -- Compilation -- Installation -- Configuration -- Dial and Options String -- misdn cli commands -- Debugging and sending Bugreports -- Examples -- Known working Configurations -- Known Problems -- Changes - - -Fast Installation Guide ------------------------ - -It is easy to install mISDN and mISDNuser. Using the Makefile from -channels/misdn. You just need to type: - -cd channels/misdn -make misdn - -Then all the necessary files are fetched from jollys homepage and are patched -with the Echocanellor. - - -Pre-Requisites --------------- - -To compile and install this driver, you'll need at least one mISDN Driver and -the mISDNuser package. Chan_misdn works with both, the current release version -and the development (svn trunk) version of Asterisk. mISDNuser and mISDN must -be fetched from jollys homepage and must be patched with the Echocancellor. - -Please Note that mISDN works good for the linux-2.6.x kernels. Some of the -mISDN drivers do not compile against the 2.4.x or older kernels, you can patch -them, but than you'll get mysterious errors. - -Using Kernels > 2.6.9 works perfect. - -Ok so far so good, now follow the compilation instructions. - -!! Dont forget to create the /dev/mISDN device node. - -Compilation ------------ - -The compilation of chan_misdn requires a library which will be generated under -channels/misdn/. - -To compile this library you just need to go into this directory and type -make. Now you can go back to the asterisk source root and type make install -again, which now should compile and install chan_misdn. - - - -Installation ------------- - -Chan_misdn is automatically installed by the asterisk installation process. - -There is a sample init.d script for loading the mISDN modules (mISDN.sample), -5Asimply copy it to /etc/init.d/ and modify it, there you can enter your cards. - -!! Forget to use capi together with chan_misdn. - - -Configuration -------------- - -First of all you must configure the mISDN drivers. Each driver module has got -an options and layermask option, which tells the driver wether to start in -TE, NT, PP or PMP mode (there are lots more, please read docs in misdn for -that). - -After thinking about the above you'll probably want to configure the -misdn.conf file which resides in the asterisk config directory (normally -/etc/asterisk). - -- misdn.conf: [general] -The misdn.conf file contains a "general" Section, and user sections which -contain misdn port settings and different Asterisk contexts. - -The general section contains especially a variable named context with which -the default context is set. There is also the very important debug variable -which you can set from the Asterisk cli (command line interface) or in this -configuration file, bigger numbers will lead to more debug output. There's also a -tracefile option, which takes a path+filename where debug output is written -to. - -- misdn.conf: [default] section -The default section is another special section which can contain all the -options available int the usr/port sections. the user/port section inherit -their parameters from the default section. - -- misdn.conf: user/port sections -The user sections have names which are unequal to "general". Those sections -contain the ports variable which mean the mISDN Ports. Here you can add -multiple ports, comma separated. - -Espacially for TE-Mode Ports there is a msns variable. This variable tells the -chan_misdn driver to listen for incomming calls with the given msns, you can -insert a '*' as single msn, which leads in getting every incoming call (if -you want to share on PMP TE S0 with a asterisk and a phone or isdn card you -should insert here the msns which you'll like to give the Asterisk). Finally -a context variable resides in the user sections, which tells chan_misdn where -to send incoming calls to in the Asterisk dial plan (extension.conf). - -In NT-Mode Ports there is a new option, directly after the port number you can -write ptp, this enables PP Mode for this port, please look at misdn.conf.sample for -an example. - - -Dial and Options String ------------------------ - -The dial string of chan_misdn got more complex, because we added more features, -so the generic dial string looks like: - -mISDN/<port>|g:<group>/<extension>[/<OPTIONSSTRING>] - -The Optionsstring looks Like: -:<optchar1><OptParam1>:<optchar2><OptParam2> - -the ":" character is the delimiter. - -The available Optchars are: - d - Send display text on called phone, text is the optparam - n - don't detect dtmf tones on called channel - h - make digital outgoing call - c - make crypted outgoing call, param is keyindex - e - perform echo cancelation on this channel, - takes taps as arguments (32,64,128,256) - s - send Non Inband DTMF as inband - vr - rxgain control - vt - txgain control - - -chan_misdn registers a new dial plan application "misdn_set_opt" when -loaded. This application takes the Optionsstring as argument. The Syntax is: - -misdn_set_opt(<OPTIONSSTRING>) - - -When you set options in the dialstring, the options are set in the external -channel. When you set options with misdn_set_opt, they are set in the current -incoming channel. So if you like to use static encryption, the scenario looks -as follows: - -Phone1 --> * Box 1 --> PSTN_TE -PSTN_TE --> * Box 2 --> Phone2 - -The Encryption must be done on the PSTN sides, so the dialplan on the boxes -are: - -* Box 1: -exten => _${CRYPT_PREFIX}X.,1,Dial(mISDN/g:outbound/:c1) - -* Box 2: -exten => ${CRYPT_MSN},1,misdn_set_opt(:c1) -exten => ${CRYPT_MSN},2,dial(${PHONE2}) - - - - -misdn cli commands ------------------- - -At the Asterisk cli you can try to type in: - -misdn <tab> <tab> - -Now you should see the misdn cli commands: - -- clean - -> pid (cleans a broken call, use with care, leads often - to a segmentation fault) -- send - -> display (sends a Text Message to a Asterisk channel, - this channel must be an misdn channel) -- set - -> debug (sets debug level) -- show - -> config (shows the configuration options) - -> channels (shows the current active misdn channels) - -> channel (shows details about the given misdn channels) - -> stacks (shows the currend ports, there protocols and states) - -> fullstacks (shows the current active and inactive misdn channels) - -- restart - -> port (restarts given port (L2 Restart) ) - -- reload (reloads misdn.conf) - -You can only use "misdn send display" when an Asterisk channel is created and -isdn is in the correct state. "correct state" means that you have established a -call to another phone (mustn't be isdn though). - -Then you use it like this: - -misdn send display mISDN/1/101 "Hello World!" - -where 1 is the Port of the Card where the phone is plugged in, and 101 is the -msn (callerid) of the Phone to send the text to. - - - -Debugging and sending bug reports ---------------------------------- - -If you encounter problems, you should set up the debugging flag, usually debug=1 -should be enough. the messages are divided in asterisk and misdn parts. -Misdn Debug messages begin with an 'I', asterisk messages begin with an '*', -the rest is clear I think. - -Please take a trace of the problem and open a report in the Asterisk issue -tracker at http://bugs.digium.com in the "channel drivers" project, -"chan_misdn" category. Read the bug guidelines to make sure you -provide all the information needed. - - -Examples --------- - -Here are some examples of how to use chan_misdn in the dialplan (extensions.conf): - - -[globals] -OUT_PORT=1 ; The physical Port of the Card -OUT_GROUP=ExternE1 ; The Group of Ports defined in misdn.conf - -[misdnIn] -exten => _X.,1,Dial(mISDN/${OUT_PORT}/${EXTEN}) -exten => _0X.,1,Dial(mISDN/g:${OUT_GROUP}/${EXTEN:1}) -exten => _1X.,1,Dial(mISDN/g:${OUT_GROUP}/${EXTEN:1}/:dHello) -exten => _1X.,1,Dial(mISDN/g:${OUT_GROUP}/${EXTEN:1}/:dHello Test:n) - -In the last line you will notice the last argument (Hello), this is sended -as Display Message to the Phone. - - -Known working configurations ----------------------------- - -In this section I'll put working configurations for chan_misdn. Beware It -seems that between Kernel 2.6.3 and Kernel 2.6.8 there were lots of mISDN -Bugs. I use Kernel 2.6.9 now, it works quite ok, Kernel 2.6.10+ has changed -the pci_find_subgsys funktion, so hfc_multi from mISDN doesn't compile against -it, you can just change pci_find_subsys to pci_get_subsys, this works. - - -- chan_misdn-0.0.3-rc1: - * linux-kernel >= 2.6.3 (but at least 2.6) - * asterisk >= v1-0 - * mISDN/mISDNuser since September/04 - -- chan_misdn-0.0.3-rc3: - * linux-kernel >= 2.6.3 (but at least 2.6) - * asterisk >= v1-0.2 - * mISDN/mISDNuser since December/04 - -- chan_misdn-0.0.3-rc4: - * linux-kernel >= 2.6.8 (but at least 2.6) - * asterisk >= v1-0.2 - * mISDN/mISDNuser head on cvs.isdn4linux.de - -- chan_misdn-0.0.3-rc6: - * linux-kernel >= 2.6.8 (but at least 2.6) - * asterisk >= v1-0.2 - * mISDN/mISDNuser head on cvs.isdn4linux.de - -- chan_misdn-0.1.0 - * linux-kernel >= 2.6.8 (but at least 2.6) - * asterisk >= v1-0.2 , also CVS Head - * mISDN/mISDNuser (3.0-beta) from isdn.jolly.de - -- chan_misdn-0.2.1 - * linux-kernel >= 2.6.8 (but at least 2.6) - * asterisk >= v1.2 , also CVS Head - * mISDN/mISDNuser (3.0) from isdn.jolly.de - - -Known Problems --------------- - -* When I use mISDN->IAX I cannot make Trunk calls - --> You need to use ztdummy as dummy zaptel interface for the iax timing in -trunking mode, simply grab libpri, zaptel and compile them (i think you need -to modify the makefile in zaptel to add ztdummy to the defaultly compiled -modules) then modprobe ztdummy, this resolves the problem. - - -* I cannot hear any tone after succesfull CONNECT to other end - --> you forgot to load mISDNdsp, which is now needed by chan_misdn for switching -and dtmf tone detection - -* I have strange ISDN behavior: sometimes I hear the other end, sometimes -not. Also I get STATUS Events with cause 100, with misdn debugging - --> Please update to newest version of chan_misdn and set the te_choose_channel -option in misdn.conf to yes - -Changes -------- -in the Changes File - diff --git a/1.2-netsec/doc/README.mp3 b/1.2-netsec/doc/README.mp3 deleted file mode 100644 index 5c7ca16c7..000000000 --- a/1.2-netsec/doc/README.mp3 +++ /dev/null @@ -1,19 +0,0 @@ -* Asterisk MP3 Support -====================== - -* MP3 Music On Hold -Asterisk supports mp3 playback for music on hold via the mpg123 -program, available from www.mpg123.de. - -The latest release of mpg123 is mpg123 0.59r. -The latest development release of mpg123 is mpg123 pre0.59s. - -Please use mpg123 0.59r. Using mpg123 pre0.59s can/may/will result -in crashes and/or unreliable playback. - -Running "make mpg123" in the Asterisk source directory will -download and make the working version. - -* MP3 Format driver -A format driver for reading MP3 audio files is available -in the asterisk-addons SVN repository on svn.digium.com diff --git a/1.2-netsec/doc/README.mysql b/1.2-netsec/doc/README.mysql deleted file mode 100644 index 27adaa956..000000000 --- a/1.2-netsec/doc/README.mysql +++ /dev/null @@ -1,15 +0,0 @@ -MYSQL LICENSING UPDATE -====================== -We were recently contacted by MySQL and informed that the MySQL client -libraries are now under GPL license and not LGPL license as before. - -Since Asterisk does allow exceptions to GPL, we are removing MySQL support -from standard Asterisk. We will, where appropriate, make it available via -a separate package which will only be usable when Asterisk is used completely -within GPL (i.e. not in conjunction with G.729, OpenH.323, etc). We -apologize for the confusion. - -You may find this in the new "asterisk-addons" package. - -Mark Spencer -Digium diff --git a/1.2-netsec/doc/README.odbcstorage b/1.2-netsec/doc/README.odbcstorage deleted file mode 100644 index 2cba0b4b9..000000000 --- a/1.2-netsec/doc/README.odbcstorage +++ /dev/null @@ -1,33 +0,0 @@ -ODBC Voicemail Storage -====================== - -ODBC Storage allows you to store voicemail messages within a database -instead of using a file. This is *not* a full realtime engine and -*only* supports ODBC. The table description for the "voicemessages" -table is as follows: - -+----------------+-------------+------+-----+---------+-------+ -| Field | Type | Null | Key | Default | Extra | -+----------------+-------------+------+-----+---------+-------+ -| msgnum | int(11) | YES | | NULL | | -| dir | varchar(80) | YES | MUL | NULL | | -| context | varchar(80) | YES | | NULL | | -| macrocontext | varchar(80) | YES | | NULL | | -| callerid | varchar(40) | YES | | NULL | | -| origtime | varchar(40) | YES | | NULL | | -| duration | varchar(20) | YES | | NULL | | -| mailboxuser | varchar(80) | YES | | NULL | |* -| mailboxcontext | varchar(80) | YES | | NULL | |* -| recording | longblob | YES | | NULL | | -+----------------+-------------+------+-----+---------+-------+ - -*=Denotes new fieldnames, define EXTENDED_ODBC_STORAGE in -apps/Makefile to enable the functionality. - -The database name (from /etc/asterisk/res_odbc.conf) is in the -"odbcstorage" variable in the general section of voicemail.conf. - -You may modify the voicemessages table name by using -odbctable=??? in voicemail.conf - - diff --git a/1.2-netsec/doc/README.privacy b/1.2-netsec/doc/README.privacy deleted file mode 100644 index 3a990fa4b..000000000 --- a/1.2-netsec/doc/README.privacy +++ /dev/null @@ -1,361 +0,0 @@ -Title: Everything About The Privacy Options In The Dial Command That -You Never Wanted To Know, And Even A Little More On Zapateller and -PrivacyManager: - -by Steve Murphy - - -So, you want to avoid talking to pesky telemarketers/charity -seekers/poll takers/magazine renewers/etc? - -============= -First of all: -============= - -Try the FTC "Don't call" database, this alone will reduce your -telemarketing call volume considerably. (see: -https://www.donotcall.gov/default.aspx ) But, this list won't protect -from the Charities, previous business relationships, etc. - - -================================= -Next, Fight against autodialers!! -================================= - -Zapateller detects if callerid is present, and if not, plays the -da-da-da tones that immediately precede messages like, "I'm sorry, -the number you have called is no longer in service." - -Most humans, even those with unlisted/callerid-blocked numbers, will -not immediately slam the handset down on the hook the moment they hear -the three tones. But autodialers seem pretty quick to do this. - -I just counted 40 hangups in Zapateller over the last year in my -CDR's. So, that is possibly 40 different telemarketers/charities that have -hopefully slashed my back-waters, out-of-the-way, humble home phone -number from their lists. - -I highly advise Zapateller for those seeking the nirvana of "privacy". - - -======================================= -Next, Fight against the empty CALLERID! -======================================= - -A considerable percentage of the calls you don't want, come from -sites that do not provide CallerID. - -Null callerid's are a fact of life, and could be a friend with an -unlisted number, or some charity looking for a handout. The -PrivacyManager application can help here. It will ask the caller to -enter a 10-digit phone number. They get 3 tries(configurable), and this is -configurable, with control being passed to priority+101 if they won't -supply one. - -PrivacyManager can't guarantee that the number they supply is any -good, tho, as there is no way to find out, short of hanging up and -calling them back. But some answers are obviously wrong. For instance, -it seems a common practice for telemarketers to use your own number -instead of giving you theirs. A simple test can detect this. More -advanced tests would be to look for -555- numbers, numbers that count -up or down, numbers of all the same digit, etc. - -My logs show that 39 have hung up in the PrivacyManager script over -the last year. - -(Note: Demanding all unlisted incoming callers to enter their CID may -not always be appropriate for all users. Another option might be to -use call screening. See below.) - -========================== -Next, use a WELCOME MENU ! -========================== - -Experience has shown that simply presenting incoming callers with -a set of options, no matter how simple, will deter them from calling -you. In the vast majority of situations, a telemarketer will simply -hang up rather than make a choice and press a key. - -This will also immediately foil all autodialers that simply belch a -message in your ear and hang up. - - ----------------------------------------------- -Example usage of Zapateller and PrivacyManager: ----------------------------------------------- - -[homeline] -exten => s,1,Answer -exten => s,2,SetVar,repeatcount=0 -exten => s,3,Zapateller,nocallerid -exten => s,4,PrivacyManager -exten => s,105,Background(tt-allbusy) ;; do this if they don't enter a number to Privacy Manager -exten => s,106,Background(tt-somethingwrong) -exten => s,107,Background(tt-monkeysintro) -exten => s,108,Background(tt-monkeys) -exten => s,109,Background(tt-weasels) -exten => s,110,Hangup -exten => s,5,GotoIf($[ "${CALLERIDNUM}" = "7773334444" & "${CALLERIDNAME}" : "Privacy Manager" ]?callerid-liar|s|1:s|7) - -I suggest using Zapateller at the beginning of the context, before -anything else, on incoming calls.This can be followed by the -PrivacyManager App. - -Make sure, if you do the PrivacyManager app, that you take care of the -error condition! or their non-compliance will be rewarded with access -to the system. In the above, if they can't enter a 10-digit number in -3 tries, they get the humorous "I'm sorry, but all household members -are currently helping other telemarketers...", "something is terribly -wrong", "monkeys have carried them away...", various loud monkey -screechings, "weasels have...", and a hangup. There are plenty of -other paths to my torture scripts, I wanted to have some fun. - -In nearly all cases now, the telemarketers/charity-seekers that -usually get thru to my main intro, hang up. I guess they can see it's -pointless, or the average telemarketer/charity-seeker is instructed -not to enter options when encountering such systems. Don't know. - -=================== -Next: Torture Them! -=================== - -I have developed an elaborate script to torture Telemarketers, and -entertain friends. (See -http://www.voip-info.org/wiki-Asterisk+Telemarketer+Torture ) - -While mostly those that call in and traverse my teletorture scripts -are those we know, and are doing so out of curiosity, there have been -these others from Jan 1st,2004 thru June 1st, 2004: -(the numbers may or may not be correct.) - -603890zzzz hung up telemarket options. -"Integrated Sale" called a couple times. hung up in telemarket options -"UNITED STATES GOV" (-- maybe a military recruiter, trying to lure one of my sons). -800349zzzz -- hung up in charity intro -800349zzzz -- hung up in charity choices, intro, about the only one who actually travelled to the bitter bottom of the scripts! -216377zzzz -- hung up the magazine section -626757zzzz = "LIR " (pronounced "Liar"?) hung up in telemarket intro, then choices -757821zzzz -- hung up in new magazine subscription options. - -That averages out to maybe 1 a month. That puts into question whether -the ratio of the amount of labor it took to make the scripts versus -the benefits of lower call volumes was worth it, but, well, I had fun, -so what the heck. - -but, that's about it. Not a whole lot. But I haven't had to say "NO" -or "GO AWAY" to any of these folks for about a year now ...! - -======================================== - Using Call Screening -======================================= - -Another option is to use call screening in the Dial command. It has -two main privacy modes, one that remembers the CID of the caller, and -how the callee wants the call handled, and the other, which does not -have a "memory". - -Turning on these modes in the dial command results in this sequence of -events, when someone calls you at an extension: - -1. The caller calls the Asterisk system, and at some point, selects an -option or enters an extension number that would dial your extension. - -2. Before ringing your extension, the caller is asked to supply an -introduction. The application asks them: "After the tone, say your -name". They are allowed 4 seconds of introduction. - -3. After that, they are told "Hang on, we will attempt to connect you -to your party. Depending on your dial options, they will hear ringing -indications, or get music on hold. I suggest music on hold. - -4. Your extension is then dialed. When (and if) you pick up, you are -told that a caller presenting themselves as <their recorded intro is -played> is calling, and you have options, like being connected, -sending them to voicemail, torture, etc. - -5. You make your selection, and the call is handled as you chose. - - -There are some variations, and these will be explained in due course. - - -To use these options, set your Dial to something like: - -exten => 3,3,Dial(Zap/5r3&Zap/6r3|35|tmPA(beep)) - -or - -exten => 3,3,Dial(Zap/5r3&Zap/6r3|35|tmP(something)A(beep)) - -or - -exten => 3,3,Dial(Zap/5r3&Zap/6r3|35|tmpA(beep)) - - -The 't' allows the dialed party to transfer the call using '#'. It's -optional. - -The 'm' is for music on hold. I suggest it. Otherwise, the calling -party gets to hear all the ringing, and lack thereof. It is generally -better to use Music On Hold. Lots of folks hang up after the 3rd or -4th ring, and you might lose the call before you can enter an option! - -The 'P' option alone will database everything using the extension as a -default 'tree'. To get multiple extensions sharing the same database, use -P(some-shared-key). Also, if the same person has multiple extensions, -use P(unique-id) on all their dial commands. - -Use little 'p' for screening. Every incoming call will include a -prompt for the callee's choice. - -the A(beep), will generate a 'beep' that the callee will hear if they -choose to talk to the caller. It's kind of a prompt to let the callee -know that he has to say 'hi'. It's not required, but I find it -helpful. - -When there is no CallerID, P and p options will always record an intro -for the incoming caller. This intro will be stored temporarily in the -/var/lib/asterisk/sounds/priv-callerintros dir, under the name -NOCALLERID_<extension><channelname> and will be erased after the -callee decides what to do with the call. - -Of course, NOCALLERID is not stored in the database. All those with no -CALLERID will be considered "Unknown". - -======================== - The 'N' and 'n' options -======================== - -Two other options exist, that act as modifiers to the privacy options -'P' and 'p'. They are 'N' and 'n'. You can enter them as dialing -options, but they only affect things if P or p are also in the -options. - -'N' says, "Only screen the call if no CallerID is present". So, if a -callerID were supplied, it will come straight thru to your extension. - -'n' says, "Don't save any introductions". Folks will be asked to -supply an introduction ("At the tone, say your name") every time they -call. Their introductions will be removed after the callee makes a -choice on how to handle the call. Whether the P option or the p option -is used, the incoming caller will have to supply their intro every -time they call. - -======================= -Recorded Introductions -======================= - -[Philosophical Side Note: -The 'P' option stores the CALLERID in the database, along with the -callee's choice of actions, as a convenience to the CALLEE, whereas -introductions are stored and re-used for the convenience of the CALLER.] - -Unless instructed to not save introductions (see the 'n' option above), -the screening modes will save the recordings of the caller's names in -the directory /var/lib/asterisk/sounds/priv-callerintros, if they have -a CallerID. Just the 10-digit callerid numbers are used as filenames, -with a ".gsm" at the end. - -Having these recordings around can be very useful, however... - -First of all, if a callerid is supplied, and a recorded intro for that -number is already present, the caller is spared the inconvenience of -having to supply their name, which shortens their call a bit. - -Next of all, these intros can be used in voicemail, played over -loudspeakers, and perhaps other nifty things. For instance: - -exten => s,7,System(/usr/bin/play /var/lib/asterisk/sounds/priv-callerintros/${CALLERIDNUM}.gsm&|0) - -When a call comes in at the house, the above priority gets executed, -and the callers intro is played over the phone systems speakers. This -gives us a hint who is calling. - -(Note: the |0 option at the end of the System command above, is a -local mod I made to the System command. It forces a 0 result code to -be returned, whether the play command successfully completed or -not. Therefore, I don't have to ensure that the file exists or -not. While I've turned this mod into the developers, it hasn't been -incorporated yet. You might want to write an AGI or shell script to -handle it a little more intelligently) - -And one other thing. You can easily supply your callers with an option -to listen to, and re-record their introductions. Here's what I did in -the home system's extensions.conf. (assume that a -Goto(home-introduction|s|1) exists somewhere in your main menu as an -option): - -[home-introduction] -exten => s,1,Background,intro-options ;; Script: To hear your Introduction, dial 1. - ;; to record a new introduction, dial 2. - ;; to return to the main menu, dial 3. - ;; to hear what this is all about, dial 4. -exten => 1,1,Playback,priv-callerintros/${CALLERIDNUM} -exten => 1,2,Goto(s,1) -exten => 2,1,Goto(home-introduction-record,s,1) -exten => 3,1,Goto(homeline,s,7) -exten => 4,1,Playback,intro-intro ;; Script: - ;; This may seem a little strange, but it really is a neat - ;; thing, both for you and for us. I've taped a short introduction - ;; for many of the folks who normally call us. Using the Caller ID - ;; from each incoming call, the system plays the introduction - ;; for that phone number over a speaker, just as the call comes in. - ;; This helps the folks - ;; here in the house more quickly determine who is calling. - ;; and gets the right ones to gravitate to the phone. - ;; You can listen to, and record a new intro for your phone number - ;; using this menu. -exten => 4,2,Goto(s,1) -exten => t,1,Goto(s,1) -exten => i,1,Background,invalid -exten => i,2,Goto(s,1) -exten => o,1,Goto(s,1) - -[home-introduction-record] -exten => s,1,Background,intro-record-choices ;; Script: - ;; If you want some advice about recording your - ;; introduction, dial 1. - ;; otherwise, dial 2, and introduce yourself after - ;; the beep. -exten => 1,1,Playback,intro-record - ;; Your introduction should be short and sweet and crisp. - ;; Your introduction will be limited to 4 seconds. - ;; This is NOT meant to be a voice mail message, so - ;; please, don't say anything about why you are calling. - ;; After we are done making the recording, your introduction - ;; will be saved for playback. - ;; If you are the only person that would call from this number, - ;; please state your name. Otherwise, state your business - ;; or residence name instead. For instance, if you are - ;; friend of the family, say, Olie McPherson, and both - ;; you and your kids might call here a lot, you might - ;; say: "This is the distinguished Olie McPherson Residence!" - ;; If you are the only person calling, you might say this: - ;; "This is the illustrious Kermit McFrog! Pick up the Phone, someone!!" - ;; If you are calling from a business, you might pronounce a more sedate introduction,like, - ;; "Fritz from McDonalds calling.", or perhaps the more original introduction: - ;; "John, from the Park County Morgue. You stab 'em, we slab 'em!". - ;; Just one caution: the kids will hear what you record every time - ;; you call. So watch your language! - ;; I will begin recording after the tone. - ;; When you are done, hit the # key. Gather your thoughts and get - ;; ready. Remember, the # key will end the recording, and play back - ;; your intro. Good Luck, and Thank you!" -exten => 1,2,Goto(2,1) -exten => 2,1,Background,intro-start - ;; OK, here we go! After the beep, please give your introduction. -exten => 2,2,Background,beep -exten => 2,3,Record,priv-callerintros/${CALLERIDNUM}:gsm|4 -exten => 2,4,Background,priv-callerintros/${CALLERIDNUM} -exten => 2,5,Goto(home-introduction,s,1) -exten => t,1,Goto(s,1) -exten => i,1,Background,invalid -exten => i,2,Goto(s,1) -exten => o,1,Goto(s,1) - - -In the above, you'd most likely reword the messages to your liking, -and maybe do more advanced things with the 'error' conditions (i,o,t priorities), -but I hope it conveys the idea... - - diff --git a/1.2-netsec/doc/README.realtime b/1.2-netsec/doc/README.realtime deleted file mode 100644 index 1de651cd0..000000000 --- a/1.2-netsec/doc/README.realtime +++ /dev/null @@ -1,111 +0,0 @@ -The Asterisk Realtime Architecture ----------------------------------- - -The Asterisk Realtime Architecture is a new set of drivers and -functions implemented in Asterisk 1.2. The benefits of this -architecture are many, both from a code management -standpoint and from an installation perspective. - -Additional information on the configuration of Realtime with Asterisk -can be found in README.extconfig - -The ARA is designed to be independent of storage. Currently, most -drivers are based on SQL, but the architecture should be able to handle -other storage methods in the future, like LDAP. - -The main benefit comes in the database support. In Asterisk v1.0 some -functions supported MySQL database, some PostgreSQL and other ODBC. -With the ARA, we have a unified database interface internally in Asterisk, -so if one function supports database integration, all databases that has a -realtime driver will be supported in that function. - -Currently there are three realtime database drivers: - -* ODBC: Support for UnixODBC, integrated into Asterisk - The UnixODBC subsystem supports many different databases, - please check www.unixodbc.org for more information. -* MySQL: Found in the asterisk-addons cvs archive on cvs.digium.com -* Res_perl: Found in the asterisk-addons cvs archive on cvs.digium.com - - -* Two modes: Static and Realtime --------------------------------- -The ARA realtime mode is used to dynamically load and update objects. -This mode is used in the SIP and IAX2 channels, as well as in the voicemail -system. For SIP and IAX2 this is similar to the v1.0 MYSQL_FRIENDS -functionality. With the ARA, we now support many more databases for -dynamic configuration of phones. - -The ARA static mode is used to load configuration files. For the Asterisk -modules that read configurations, there's no difference between a static -file in the file system, like extensions.conf, and a configuration loaded -from a database. - -* Realtime SIP friends ----------------------- -The SIP realtime objects are users and peers that are loaded in memory -when needed, then deleted. This means that Asterisk currently can't handle -voicemail notification and NAT keepalives for these peers. Other than that, -most of the functionality works the same way for realtime friends as for -the ones in static configuration. - -There is some work to create a solution for Realtime SIP devices that -loads from database and stays in memory for the duration of a call or -a registration, but that work is not integrated into Asterisk yet. - -* New function in the dial plan: The Realtime Switch ----------------------------------------------------- -The realtime switch is more than a port of functionality in v1.0 to the -new architecture, this is a new feature of Asterisk based on the -ARA. The realtime switch lets your Asterisk server do database lookups -of extensions in realtime from your dial plan. You can have many Asterisk -servers sharing a dynamically updated dial plan in real time with this -solution. - -Note that this switch does _NOT_ support Caller ID matching, only -extension name/pattern matching. - -* So what can you do? ---------------------- -The realtime Architecture lets you store all of your configuration in -databases and reload it whenever you want. You can force a reload over -the AMI, Asterisk Manager Interface or by calling Asterisk from a -shell script with - asterisk -rx "reload" - -You may also dynamically add SIP and IAX devices and extensions -and making them available without a reload, by using the realtime -objects and the realtime switch. - - -* Configuration in extconfig.conf ---------------------------------- -You configure the ARA in extconfig.conf (yes, it's a strange name, but -is was defined in the early days of the realtime architecture and kind -of stuck). Please see README.extconfig for database schemas. - -The part of Asterisk that connects to the ARA use a well defined family -name to find the proper database driver. The syntax is easy: - <family> => <realtime driver>,<db name>[,<table>] - -The options following the realtime driver identified depends on the -driver. - -Defined well-known family names are: - -* sippeers, sipusers SIP peers and users -* iaxfriends IAX2 peers -* voicemail Voicemail accounts - -There is documentation of the SQL database in the file -README.extconfig in your Asterisk source code tree, the /doc -directory. - -For voicemail storage with the support of ODBC, there is a -README.odbcstorage documentation file. - - -* FreeTDS not supported ------------------------ -Due to architectural constraints in the FreeTDS libraries, FreeTDS is not -supported for use with realtime at this time. diff --git a/1.2-netsec/doc/README.sms b/1.2-netsec/doc/README.sms deleted file mode 100644 index 02e38764b..000000000 --- a/1.2-netsec/doc/README.sms +++ /dev/null @@ -1,147 +0,0 @@ -* The SMS application ---------------------- -SMS() is an application to handles calls to/from text message capable phones and -message centres using ETSI ES 201 912 protocol 1 FSK messaging over analog calls. - -Basically it allows sending and receiving of text messages over the PSTN. It is -compatible with BT Text service in the UK and works on ISDN and PSTN lines. It is -designed to connect to an ISDN or zap interface directly and uses FSK so would -probably not work over any sort of compressed link (like a VoIP call using GSM codec). - -Typical applications include:- - -1. Connection to a message centre to send text messages - probably initiated via the - manager interface or "outgoing" directory -2. Connection to an POTS line with an SMS capable phone to send messages - probably - initiated via the manager interface or "outgoing" directory -3. Acceptance of calls from the message centre (based on CLI) and storage of - received messages -4. Acceptance of calls from a POTS line with an SMS capable phone and storage of - received messages - -* Arguments to sms(): - -- First argument is queue name -- Second is options: - a: SMS() is to act as the answering side, and so send the initial FSK frame - s: SMS() is to act as a service centre side rather than as terminal equipment - -- If a third argument is specified, then SMS does not handle the call at all, - but takes the third argument as a destination number to send an SMS to -- The forth argument onward is a message to be queued to the number in the - third argument. All this does is create the file in the me-sc directory. - If 's' is set then the number is the source - address and the message placed in the sc-me directory. - -All text messages are stored in /var/spool/asterisk/sms -A log is recorded in /var/log/asterisk/sms - -There are two subdirectories called sc-me.<queuename> holding all -messages from service centre to phone, and me-sc.<queuename> holding all -messages from phone to service centre. - -In each directory are messages in files, one per file, using any filename not -starting with a dot. - -When connected as a service centre, SMS(s) will send all messages waiting in -the sc-me-<queuename> directory, deleting the files as it goes. Any -received in this mode are placed in the me-sc-<queuename> directory. - -When connected as a client, SMS() will send all messages waiting in the -me-sc-<queuename> directory, deleting the files as it goes. Any received in -this mode are placed in the sc-me-<queuename> directory. - -Message files created by SMS() use a time stamp/reference based filename. - -The format of the sms file is lines that have the form of key=value -Keys are : - -oa Originating Address - Telephone number, national number if just digits - Telephone number starting with + then digits for international - Ignored on sending messages to service centre (CLI used) -da Destination Address - Telephone number, national number if just digits - Telephone number starting with + then digits for international -scts Service Centre Time Stamp - In the format YYYY-MM-DD HH:MM:SS -pid Protocol Identifier (decimal octet value) -dcs Data coding scheme (decimal octet value) -mr Message reference (decimal octet value) -ud The message (see escaping below) -srr 0/1 Status Report Request -rp 0/1 Return Path -vp mins validity period - -Omitted fields have default values. - -Note that there is special format for ud, ud# instead of ud= which is followed -by raw hex (2 characters per octet). This is used in output where characters -other than 10,13,32-126,128-255 are included in the data. In this case a comment (line -starting ;) is added showing the printable characters - -When generating files to send to a service centre, only da and ud need be -specified. oa is ignored. - -When generating files to send to a phone, only oa and ud need be specified. da is ignored. - -When receing a message as a service centre, only the destination address is -sent, so the originating address is set to the callerid. - -EXAMPLES - -The following are examples of use within the UK using BT Text SMS/landline -service. - -This is a context to use with a manager script. - -[smsdial] -; create and send a text message, expects number+message and -; connect to 17094009 -exten => _X.,1,SMS(${CALLERIDNUM},,${EXTEN},${CALLERIDNAME}) -exten => _X.,n,SMS(${CALLERIDNUM}) -exten => _X.,n,Hangup - -The script sends - - action: originate - callerid: message <from> - exten: to - channel: Local/17094009 - context: smsdial - priority: 1 - -You put the message as the name of the caller ID (messy, I know), the -originating number and hence queue name as the number of the caller ID and the -exten as the number to which the sms is to be sent. The context uses SMS to -create the message in the queue and then SMS to communicate iwth 17094009 to -actually send the message. - -Note that the 9 on the end of 17094009 is the sub address 9 meaning no sub -address (BT specific). If a different digit is used then that is the sub -address for the sending message source address (appended to the outgoing CLI -by BT). - -For incoming calls you can use a context like this :- - -[incoming] -exten => _XXXXXX/_8005875290,1,SMS(${EXTEN:3},a) -exten => _XXXXXX/_8005875290,n,System(/usr/lib/asterisk/smsin ${EXTEN:3}) -exten => _XXXXXX/_80058752[0-8]0,1,SMS(${EXTEN:3}${CALLERIDNUM:8:1},a) -exten => _XXXXXX/_80058752[0-8]0,n,System(/usr/lib/asterisk/smsin ${EXTEN>:3}${CALLERIDNUM:8:1}) -exten => _XXXXXX/_80058752[0-8]0,n,Hangup - - -In this case the called number we get from BT is 6 digits (XXXXXX) and we are -using the last 3 digits as the queue name. - -Priority 1 causes the SMS to be received and processed for the incoming call. -It is from 080058752X0. The two versions handle the queue name as 3 digits (no -sub address) or 4 digits (with sub address). In both cases, after the call a -script (smsin) is run - this is optional, but is useful to actually processed -the received queued SMS. In our case we email them based on the target number. -Priority 3 hangs up. - -If using the CAPI drivers they send the right CLI and so the _800... would be -_0800... - diff --git a/1.2-netsec/doc/README.tds b/1.2-netsec/doc/README.tds deleted file mode 100644 index e1c27fba3..000000000 --- a/1.2-netsec/doc/README.tds +++ /dev/null @@ -1,18 +0,0 @@ -PLEASE NOTE - -The cdr_tds module is NOT compatible with version 0.63 of FreeTDS. - -The cdr_tds module is known to work with FreeTDS version 0.62.1; -it should also work with 0.62.2, 0.62.3 and 0.62.4, which are bug -fix releases. - -The cdr_tds module uses the raw "libtds" API of FreeTDS. It appears -that from 0.63 onwards, this is not considered a published API -of FreeTDS and is subject to change without notice. - -Between 0.62.x and 0.63 of FreeTDS, many incompatible changes -have been made to the libtds API. - -For newer versions of FreeTDS, it is recommended that you use the -ODBC driver. - diff --git a/1.2-netsec/doc/README.variables b/1.2-netsec/doc/README.variables deleted file mode 100644 index 8d7b8ec61..000000000 --- a/1.2-netsec/doc/README.variables +++ /dev/null @@ -1,792 +0,0 @@ ----------------------------- -Asterisk dial plan variables ----------------------------- - -There are two levels of parameter evaluation done in the Asterisk -dial plan in extensions.conf. -* The first, and most frequently used, is the substitution of variable - references with their values. -* Then there are the evaluations of expressions done in $[ .. ]. - This will be discussed below. - -Asterisk has user-defined variables and standard variables set -by various modules in Asterisk. These standard variables are -listed at the end of this document. - -___________________________ -PARAMETER QUOTING: ---------------------------- - -exten => s,5,BackGround,blabla - -The parameter (blabla) can be quoted ("blabla"). In this case, a -comma does not terminate the field. However, the double quotes -will be passed down to the Background command, in this example. - -Also, characters special to variable substitution, expression evaluation, etc -(see below), can be quoted. For example, to literally use a $ on the -string "$1231", quote it with a preceding \. Special characters that must -be quoted to be used, are [ ] $ " \. (to write \ itself, use \\). - -These Double quotes and escapes are evaluated at the level of the -asterisk config file parser. - -Double quotes can also be used inside expressions, as discussed below. - -___________________________ -VARIABLES: ---------------------------- - -Parameter strings can include variables. Variable names are arbitrary strings. -They are stored in the respective channel structure. - -To set a variable to a particular value, do : - - exten => 1,2,Set(varname=value) - -You can substitute the value of a variable everywhere using ${variablename}. -For example, to stringwise append $lala to $blabla and store result in $koko, -do: - - exten => 1,2,Set(koko=${blabla}${lala}) - - -There are two reference modes - reference by value and reference by name. -To refer to a variable with its name (as an argument to a function that -requires a variable), just write the name. To refer to the variable's value, -enclose it inside ${}. For example, Set takes as the first argument -(before the =) a variable name, so: - - exten => 1,2,Set(koko=lala) - exten => 1,3,Set(${koko}=blabla) - -stores to the variable "koko" the value "lala" and to variable "lala" the -value "blabla". - -In fact, everything contained ${here} is just replaced with the value of -the variable "here". - -____________________ -VARIABLE INHERITANCE --------------------- - -Variable names which are prefixed by "_" will be inherited to channels -that are created in the process of servicing the original channel in -which the variable was set. When the inheritance takes place, the -prefix will be removed in the channel inheriting the variable. If the -name is prefixed by "__" in the channel, then the variable is -inherited and the "__" will remain intact in the new channel. - -In the dialplan, all references to these variables refer to the same -variable, regardless of having a prefix or not. Note that setting any -version of the variable removes any other version of the variable, -regardless of prefix. - -Example: - -Set(__FOO=bar) ; Sets an inherited version of "FOO" variable -Set(FOO=bar) ; Removes the inherited version and sets a local - ; variable. - -However, - -NoOp(${__FOO}) is identical to NoOp(${FOO}) - - - -___________________________________ -SELECTING CHARACTERS FROM VARIABLES ------------------------------------ - -The format for selecting characters from a variable can be expressed as: - - ${variable_name[:offset[:length]]} - -If you want to select the first N characters from the string assigned -to a variable, simply append a colon and the number of characters to -skip from the beginning of the string to the variable name. - - ;Remove the first character of extension, save in "number" variable - exten => _9X.,1,Set(number=${EXTEN:1}) - -Assuming we've dialed 918005551234, the value saved to the 'number' variable -would be 18005551234. This is useful in situations when we require users to -dial a number to access an outside line, but do not wish to pass the first -digit. - -If you use a negative offset number, Asterisk starts counting from the end -of the string and then selects everything after the new position. The following -example will save the numbers 1234 to the 'number' variable, still assuming -we've dialed 918005551234. - - ;Remove everything before the last four digits of the dialed string - exten => _9X.,1,Set(number=${EXTEN:-4}) - -We can also limit the number of characters from our offset position that we -wish to use. This is done by appending a second colon and length value to the -variable name. The following example will save the numbers 555 to the 'number' -variable. - - ;Only save the middle numbers 555 from the string 918005551234 - exten => _9X.,1,Set(number=${EXTEN:5:3}) - -The length value can also be used in conjunction with a negative offset. This -may be useful if the length of the string is unknown, but the trailing digits -are. The following example will save the numbers 555 to the 'number' variable, -even if the string starts with more characters than expected (unlike the -previous example). - - ;Save the numbers 555 to the 'number' variable - exten => _9X.,1,Set(number=${EXTEN:-7:3}) - -If a negative length value is entered, it is ignored and Asterisk will match -to the end of the string. -___________________________ -EXPRESSIONS: ---------------------------- - -Everything contained inside a bracket pair prefixed by a $ (like $[this]) is -considered as an expression and it is evaluated. Evaluation works similar to -(but is done on a later stage than) variable substitution: the expression -(including the square brackets) is replaced by the result of the expression -evaluation. - -For example, after the sequence: - -exten => 1,1,Set(lala=$[1 + 2]) -exten => 1,2,Set(koko=$[2 * ${lala}]) - -the value of variable koko is "6". - -and, further: - -exten => 1,1,Set,(lala=$[ 1 + 2 ]); - -will parse as intended. Extra spaces are ignored. - - -______________________________ -SPACES INSIDE VARIABLE VALUES ------------------------------- -If the variable being evaluated contains spaces, there can be problems. - -For these cases, double quotes around text that may contain spaces -will force the surrounded text to be evaluated as a single token. -The double quotes will be counted as part of that lexical token. - -As an example: - -exten => s,6,GotoIf($[ "${CALLERIDNAME}" : "Privacy Manager" ]?callerid-liar|s|1:s|7) - -The variable CALLERIDNAME could evaluate to "DELOREAN MOTORS" (with a space) -but the above will evaluate to: - -"DELOREAN MOTORS" : "Privacy Manager" - -and will evaluate to 0. - -The above without double quotes would have evaluated to: - -DELOREAN MOTORS : Privacy Manager - -and will result in syntax errors, because token DELOREAN is immediately -followed by token MOTORS and the expression parser will not know how to -evaluate this expression, because it does not match its grammar. - -_____________________ -OPERATORS ---------------------- -Operators are listed below in order of increasing precedence. Operators -with equal precedence are grouped within { } symbols. - - expr1 | expr2 - Return the evaluation of expr1 if it is neither an empty string - nor zero; otherwise, returns the evaluation of expr2. - - expr1 & expr2 - Return the evaluation of expr1 if neither expression evaluates to - an empty string or zero; otherwise, returns zero. - - expr1 {=, >, >=, <, <=, !=} expr2 - Return the results of integer comparison if both arguments are - integers; otherwise, returns the results of string comparison - using the locale-specific collation sequence. The result of each - comparison is 1 if the specified relation is true, or 0 if the - relation is false. - - expr1 {+, -} expr2 - Return the results of addition or subtraction of integer-valued - arguments. - - expr1 {*, /, %} expr2 - Return the results of multiplication, integer division, or - remainder of integer-valued arguments. - - - expr1 - Return the result of subtracting expr1 from 0. - This, the unary minus operator, is right associative, and - has the same precedence as the ! operator. - - ! expr1 - Return the result of a logical complement of expr1. - In other words, if expr1 is null, 0, an empty string, - or the string "0", return a 1. Otherwise, return a 0. - It has the same precedence as the unary minus operator, and - is also right associative. - - expr1 : expr2 - The `:' operator matches expr1 against expr2, which must be a - regular expression. The regular expression is anchored to the - beginning of the string with an implicit `^'. - - If the match succeeds and the pattern contains at least one regu- - lar expression subexpression `\(...\)', the string correspond- - ing to `\1' is returned; otherwise the matching operator - returns the number of characters matched. If the match fails and - the pattern contains a regular expression subexpression the null - string is returned; otherwise 0. - - Normally, the double quotes wrapping a string are left as part - of the string. This is disastrous to the : operator. Therefore, - before the regex match is made, beginning and ending double quote - characters are stripped from both the pattern and the string. - - expr1 =~ expr2 - Exactly the same as the ':' operator, except that the match is - not anchored to the beginning of the string. Pardon any similarity - to seemingly similar operators in other programming languages! - The ":" and "=~" operators share the same precedence. - - expr1 ? expr2 :: expr3 - Traditional Conditional operator. If expr1 is a number - that evaluates to 0 (false), expr3 is result of the this - expression evaluation. Otherwise, expr2 is the result. - If expr1 is a string, and evaluates to an empty string, - or the two characters (""), then expr3 is the - result. Otherwise, expr2 is the result. In Asterisk, all - 3 exprs will be "evaluated"; if expr1 is "true", expr2 - will be the result of the "evaluation" of this - expression. expr3 will be the result otherwise. This - operator has the lowest precedence. - -Parentheses are used for grouping in the usual manner. - -Operator precedence is applied as one would expect in any of the C -or C derived languages. - -Examples - - "One Thousand Five Hundred" =~ "(T[^ ]+)" - returns: Thousand - - "One Thousand Five Hundred" =~ "T[^ ]+" - returns: 8 - - "One Thousand Five Hundred" : "T[^ ]+" - returns: 0 - - "8015551212" : "(...)" - returns: 801 - - "3075551212":"...(...)" - returns: 555 - - ! "One Thousand Five Hundred" =~ "T[^ ]+" - returns: 0 (because it applies to the string, which is non-null, - which it turns to "0", and then looks for the pattern - in the "0", and doesn't find it) - - !( "One Thousand Five Hundred" : "T[^ ]+" ) - returns: 1 (because the string doesn't start with a word starting - with T, so the match evals to 0, and the ! operator - inverts it to 1 ). - - 2 + 8 / 2 - returns 6. (because of operator precedence; the division is done first, then the addition). - - 2+8/2 - returns 6. Spaces aren't necessary. - -(2+8)/2 - returns 5, of course. - -Of course, all of the above examples use constants, but would work the -same if any of the numeric or string constants were replaced with a -variable reference ${CALLERIDNUM}, for instance. - -__________________________ -NUMBERS VS STRINGS --------------------------- - -Tokens consisting only of numbers are converted to 64-bit numbers for -most of the operators. This means that overflows can occur when the -numbers get above 18 digits. Warnings will appear in the logs in this -case. -___________________________ -CONDITIONALS ---------------------------- - -There is one conditional application - the conditional goto : - - exten => 1,2,gotoif(condition?label1:label2) - -If condition is true go to label1, else go to label2. Labels are interpreted -exactly as in the normal goto command. - -"condition" is just a string. If the string is empty or "0", the condition -is considered to be false, if it's anything else, the condition is true. -This is designed to be used together with the expression syntax described -above, eg : - - exten => 1,2,gotoif($[${CALLERID} = 123456]?2|1:3|1) - -Example of use : - -exten => s,2,Set(vara=1) -exten => s,3,Set(varb=$[${vara} + 2]) -exten => s,4,Set(varc=$[${varb} * 2]) -exten => s,5,GotoIf($[${varc} = 6]?99|1:s|6) - -___________________________ -PARSE ERRORS ---------------------------- - -Syntax errors are now output with 3 lines. - -If the extensions.conf file contains a line like: - -exten => s,6,GotoIf($[ "${CALLERIDNUM}" = "3071234567" & & "${CALLERIDNAME}" : "Privacy Manager" ]?callerid-liar|s|1:s|7) - -You may see an error in /var/log/asterisk/messages like this: - -Jul 15 21:27:49 WARNING[1251240752]: ast_yyerror(): syntax error: parse error, unexpected TOK_AND, expecting TOK_MINUS or TOK_LP or TOKEN; Input: -"3072312154" = "3071234567" & & "Steves Extension" : "Privacy Manager" - ^ - -The log line tells you that a syntax error was encountered. It now -also tells you (in grand standard bison format) that it hit an "AND" -(&) token unexpectedly, and that was hoping for for a MINUS (-), LP -(left parenthesis), or a plain token (a string or number). - -The next line shows the evaluated expression, and the line after -that, the position of the parser in the expression when it became confused, -marked with the "^" character. - -___________________________ -NULL STRINGS ---------------------------- - -Testing to see if a string is null can be done in one of two different ways: - - exten => _XX.,1,GotoIf($["${calledid}" != ""]?3) - - exten => _XX.,1,GotoIf($[foo${calledid} != foo]?3) - - -The second example above is the way suggested by the WIKI. It will -work as long as there are no spaces in the evaluated value. - -The first way should work in all cases, and indeed, might now -be the safest way to handle this situation. - -___________________________ -WARNING ---------------------------- - -If you need to do complicated things with strings, asterisk expressions -is most likely NOT the best way to go about it. AGI scripts are an -excellent option to this need, and make available the full power of -whatever language you desire, be it Perl, C, C++, Cobol, RPG, Java, -Snobol, PL/I, Scheme, Common Lisp, Shell scripts, Tcl, Forth, Modula, -Pascal, APL, assembler, etc. - ----------------------------- -INCOMPATIBILITIES ----------------------------- - -The asterisk expression parser has undergone some evolution. It is hoped -that the changes will be viewed as positive. - -The "original" expression parser had a simple, hand-written scanner, -and a simple bison grammar. This was upgraded to a more involved bison -grammar, and a hand-written scanner upgraded to allow extra spaces, -and to generate better error diagnostics. This upgrade required bison -1.85, and part of the user community felt the pain of having to -upgrade their bison version. - -The next upgrade included new bison and flex input files, and the makefile -was upgraded to detect current version of both flex and bison, conditionally -compiling and linking the new files if the versions of flex and bison would -allow it. - -If you have not touched your extensions.conf files in a year or so, the -above upgrades may cause you some heartburn in certain circumstances, as -several changes have been made, and these will affect asterisk's behavior on -legacy extension.conf constructs. The changes have been engineered -to minimize these conflicts, but there are bound to be problems. - -The following list gives some (and most likely, not all) of areas -of possible concern with "legacy" extension.conf files: - -1. Tokens separated by space(s). - Previously, tokens were separated by spaces. Thus, ' 1 + 1 ' would evaluate - to the value '2', but '1+1' would evaluate to the string '1+1'. If this - behavior was depended on, then the expression evaluation will break. '1+1' - will now evaluate to '2', and something is not going to work right. - To keep such strings from being evaluated, simply wrap them in double - quotes: ' "1+1" ' - -2. The colon operator. In versions previous to double quoting, the - colon operator takes the right hand string, and using it as a - regex pattern, looks for it in the left hand string. It is given - an implicit ^ operator at the beginning, meaning the pattern - will match only at the beginning of the left hand string. - If the pattern or the matching string had double quotes around - them, these could get in the way of the pattern match. Now, - the wrapping double quotes are stripped from both the pattern - and the left hand string before applying the pattern. This - was done because it recognized that the new way of - scanning the expression doesn't use spaces to separate tokens, - and the average regex expression is full of operators that - the scanner will recognize as expression operators. Thus, unless - the pattern is wrapped in double quotes, there will be trouble. - For instance, ${VAR1} : (Who|What*)+ - may have have worked before, but unless you wrap the pattern - in double quotes now, look out for trouble! This is better: - "${VAR1}" : "(Who|What*)+" - and should work as previous. - -3. Variables and Double Quotes - Before these changes, if a variable's value contained one or more double - quotes, it was no reason for concern. It is now! - -4. LE, GE, NE operators removed. The code supported these operators, - but they were not documented. The symbolic operators, <=, >=, and != - should be used instead. - -5. Added the unary '-' operator. So you can 3+ -4 and get -1. - -6. Added the unary '!' operator, which is a logical complement. - Basically, if the string or number is null, empty, or '0', - a '1' is returned. Otherwise a '0' is returned. - -7. Added the '=~' operator, just in case someone is just looking for - match anywhere in the string. The only diff with the ':' is that - match doesn't have to be anchored to the beginning of the string. - -8. Added the conditional operator 'expr1 ? true_expr :: false_expr' - First, all 3 exprs are evaluated, and if expr1 is false, the 'false_expr' - is returned as the result. See above for details. - -9. Unary operators '-' and '!' were made right associative. - --------------------------------------------------------- -DEBUGGING HINTS FOR $[ ] EXPRESSIONS --------------------------------------------------------- - -There are two utilities you can build to help debug the $[ ] in -your extensions.conf file. - -The first, and most simplistic, is to issue the command: - -make testexpr2 - -in the top level asterisk source directory. This will build a small -executable, that is able to take the first command line argument, and -run it thru the expression parser. No variable substitutions will be -performed. It might be safest to wrap the expression in single -quotes... - -testexpr2 '2*2+2/2' - -is an example. - -And, in the utils directory, you can say: - -make check_expr - -and a small program will be built, that will check the file mentioned -in the first command line argument, for any expressions that might be -have problems when you move to flex-2.5.31. It was originally -designed to help spot possible incompatibilities when moving from the -pre-2.5.31 world to the upgraded version of the lexer. - -But one more capability has been added to check_expr, that might make -it more generally useful. It now does a simple minded evaluation of -all variables, and then passes the $[] exprs to the parser. If there -are any parse errors, they will be reported in the log file. You can -use check_expr to do a quick sanity check of the expressions in your -extensions.conf file, to see if they pass a crude syntax check. - -The "simple-minded" variable substitution replaces ${varname} variable -references with '555'. You can override the 555 for variable values, -by entering in var=val arguments after the filename on the command -line. So... - - check_expr /etc/asterisk/extensions.conf CALLERIDNUM=3075551212 DIALSTATUS=TORTURE EXTEN=121 - -will substitute any ${CALLERIDNUM} variable references with -3075551212, any ${DIALSTATUS} variable references with 'TORTURE', and -any ${EXTEN} references with '121'. If there is any fancy stuff -going on in the reference, like ${EXTEN:2}, then the override will -not work. Everything in the ${...} has to match. So, to substitute -#{EXTEN:2} references, you'd best say: - - check_expr /etc/asterisk/extensions.conf CALLERIDNUM=3075551212 DIALSTATUS=TORTURE EXTEN:2=121 - -on stdout, you will see something like: - - OK -- $[ "${DIALSTATUS}" = "TORTURE" | "${DIALSTATUS}" = "DONTCALL" ] at line 416 - -In the expr2_log file that is generated, you will see: - - line 416, evaluation of $[ "TORTURE" = "TORTURE" | "TORTURE" = "DONTCALL" ] result: 1 - -check_expr is a very simplistic algorithm, and it is far from being -guaranteed to work in all cases, but it is hoped that it will be -useful. - ---------------------------------------------------------- -Asterisk standard channel variables ---------------------------------------------------------- -There are a number of variables that are defined or read -by Asterisk. Here is a list of them. More information is -available in each application's help text. All these variables -are in UPPER CASE only. - -Variables marked with a * are builtin functions and can't be set, -only read in the dialplan. Writes to such variables are silently -ignored. - -${ACCOUNTCODE} * Account code (if specified) (Deprecated; use ${CDR(accountcode)}) -${BLINDTRANSFER} The name of the channel on the other side of a blind transfer -${BRIDGEPEER} Bridged peer -${CALLERANI} * Caller ANI (PRI channels) (Deprecated; use ${CALLERID(ani)}) -${CALLERID} * Caller ID (Deprecated; use ${CALLERID(all)}) -${CALLERIDNAME} * Caller ID Name only (Deprecated; use ${CALLERID(name)}) -${CALLERIDNUM} * Caller ID Number only (Deprecated; use ${CALLERID(num)}) -${CALLINGANI2} * Caller ANI2 (PRI channels) -${CALLINGPRES} * Caller ID presentation for incoming calls (PRI channels) -${CALLINGTNS} * Transit Network Selector (PRI channels) -${CALLINGTON} * Caller Type of Number (PRI channels) -${CHANNEL} * Current channel name -${CONTEXT} * Current context -${DATETIME} * Current date time in the format: DDMMYYYY-HH:MM:SS (Deprecated; use ${STRFTIME(${EPOCH},,%d%m%Y-%H:%M:%S)}) -${DB_RESULT} Result value of DB_EXISTS() dial plan function -${DNID} * Dialed Number Identifier (Deprecated; use ${CALLERID(dnid)}) -${EPOCH} * Current unix style epoch -${EXTEN} * Current extension -${ENV(VAR)} Environmental variable VAR -${GOTO_ON_BLINDXFR} Transfer to the specified context/extension/priority - after a blind transfer (use ^ characters in place of - | to separate context/extension/priority when setting - this variable from the dialplan) -${HANGUPCAUSE} * Asterisk cause of hangup (inbound/outbound) -${HINT} * Channel hints for this extension -${HINTNAME} * Suggested Caller*ID name for this extension -${INVALID_EXTEN} The invalid called extension (used in the "i" extension) -${LANGUAGE} * Current language (Deprecated; use ${LANGUAGE()}) -${LEN(VAR)} * String length of VAR (integer) -${PRIORITY} * Current priority in the dialplan -${PRIREDIRECTREASON} Reason for redirect on PRI, if a call was directed -${RDNIS} * Redirected Dial Number ID Service (Deprecated; use ${CALLERID(rdnis)}) -${TIMESTAMP} * Current date time in the format: YYYYMMDD-HHMMSS (Deprecated; use ${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}) -${TRANSFER_CONTEXT} Context for transferred calls -${UNIQUEID} * Current call unique identifier - -Application return values -------------------------- -In Asterisk 1.2, many applications return the result in a variable -instead of, as in Asterisk 1.0, changing the dial plan priority (+101). -For the various status values, see each application's help text. - -${AQMSTATUS} * addqueuemember() -${AVAILSTATUS} * chanisavail() -${CHECKGROUPSTATUS} * checkgroup() -${CHECKMD5STATUS} * checkmd5() -${CPLAYBACKSTATUS} * controlplayback() -${DIALSTATUS} * dial() -${DBGETSTATUS} * dbget() -${ENUMSTATUS} * enumlookup() -${HASVMSTATUS} * hasnewvoicemail() -${LOOKUPBLSTATUS} * lookupblacklist() -${OSPLOOKUPSTATUS} * osplookup() -${OSPNEXTSTATUS} * ospnext() -${OSPFINISHSTATUS} * ospfinish() -${PLAYBACKSTATUS} * playback() -${PQMSTATUS} * pausequeuemember() -${PRIVACYMGRSTATUS} * privacymanager() -${QUEUESTATUS} * queue() -${RQMSTATUS} * removequeuemember() -${SENDIMAGESTATUS} * sendimage() -${SENDTEXTSTATUS} * sendtext() -${SENDURLSTATUS} * sendurl() -${SYSTEMSTATUS} * system() -${TRANSFERSTATUS} * transfer() -${TXTCIDNAMESTATUS} * txtcidname() -${UPQMSTATUS} * unpausequeuemember() -${VMSTATUS} * voicmail() -${VMBOXEXISTSSTATUS} * vmboxexists() -${WAITSTATUS} * waitforsilence() - - -Various application variables ------------------------------ -${CURL} * Resulting page content for curl() -${ENUM} * Result of application EnumLookup -${EXITCONTEXT} Context to exit to in IVR menu (app background()) - or in the RetryDial() application -${MONITOR} * Set to "TRUE" if the channel is/has been monitored (app monitor()) -${MONITOR_EXEC} Application to execute after monitoring a call -${MONITOR_EXEC_ARGS} Arguments to application -${MONITOR_FILENAME} File for monitoring (recording) calls in queue -${QUEUE_PRIO} Queue priority -${QUEUESTATUS} Status of the call, one of: - (TIMEOUT | FULL | JOINEMPTY | LEAVEEMPTY | JOINUNAVAIL | LEAVEUNAVAIL) -${RECORDED_FILE} * Recorded file in record() -${TALK_DETECTED} * Result from talkdetect() -${TOUCH_MONITOR} The filename base to use with Touch Monitor (auto record) -${TOUCH_MONITOR_FORMAT} The audio format to use with Touch Monitor (auto record) -${TXTCIDNAME} * Result of application TXTCIDName -${VPB_GETDTMF} chan_vpb - -The MeetMe Conference Bridge uses the following variables: ----------------------------------------------------------- -${MEETME_RECORDINGFILE} Name of file for recording a conference with - the "r" option -${MEETME_RECORDINGFORMAT} Format of file to be recorded -${MEETME_EXIT_CONTEXT} Context for exit out of meetme meeting -${MEETME_AGI_BACKGROUND} AGI script for Meetme (zap only) -${MEETMESECS} * Number of seconds a user participated in a MeetMe conference - -The VoiceMail() application uses the following variables: ---------------------------------------------------------- -${VM_CATEGORY} Sets voicemail category -${VM_NAME} * Full name in voicemail -${VM_DUR} * Voicemail duration -${VM_MSGNUM} * Number of voicemail message in mailbox -${VM_CALLERID} * Voicemail Caller ID (Person leaving vm) -${VM_CIDNAME} * Voicemail Caller ID Name -${VM_CIDNUM} * Voicemail Caller ID Number -${VM_DATE} * Voicemail Date -${VM_MESSAGEFILE} * Path to message left by caller - -The VMAuthenticate() application uses the following variables: ---------------------------------------------------------- -${AUTH_MAILBOX} * Authenticated mailbox -${AUTH_CONTEXT} * Authenticated mailbox context - -DUNDiLookup() uses the following variables ---------------------------------------------------------- -${DUNDTECH} * The Technology of the result from a call to DUNDiLookup() -${DUNDDEST} * The Destination of the result from a call to DUNDiLookup() - -The Zaptel channel sets the following variables: ---------------------------------------------------------- -${ANI2} * The ANI2 Code provided by the network on the incoming call. - (ie, Code 29 identifies call as a Prison/Inmate Call) -${CALLTYPE} * Type of call (Speech, Digital, etc) -${CALLEDTON} * Type of number for incoming PRI extension - i.e. 0=unknown, 1=international, 2=domestic, 3=net_specific, - 4=subscriber, 6=abbreviated, 7=reserved -${CALLINGSUBADDR} * Called PRI Subaddress -${FAXEXTEN} * The extension called before being redirected to "fax" -${PRIREDIRECTREASON} * Reason for redirect, if a call was directed - -The SIP channel uses the following variables: ---------------------------------------------------------- -${SIPCALLID} * SIP Call-ID: header verbatim (for logging or CDR matching) -${SIPDOMAIN} * SIP destination domain of an inbound call (if appropriate) -${SIPUSERAGENT} * SIP user agent -${SIPURI} * SIP uri -${SIP_CODEC} Set the SIP codec for a call -${SIP_URI_OPTIONS} * additional options to add to the URI for an outgoing call - -The Agent channel uses the following variables: ---------------------------------------------------------- -${AGENTMAXLOGINTRIES} Set the maximum number of failed logins -${AGENTUPDATECDR} Whether to update the CDR record with Agent channel data -${AGENTGOODBYE} Sound file to use for "Good Bye" when agent logs out -${AGENTACKCALL} Whether the agent should acknowledge the incoming call -${AGENTAUTOLOGOFF} Auto logging off for an agent -${AGENTWRAPUPTIME} Setting the time for wrapup between incoming calls -${AGENTNUMBER} * Agent number (username) set at login -${AGENTSTATUS} * Status of login ( fail | on | off ) -${AGENTEXTEN} * Extension for logged in agent - -The Dial() application uses the following variables: ---------------------------------------------------------- -${DIALEDPEERNAME} * Dialed peer name -${DIALEDPEERNUMBER} * Dialed peer number -${DIALEDTIME} * Time for the call (seconds) -${ANSWEREDTIME} * Time from dial to answer (seconds) -${DIALSTATUS} * Status of the call, one of: - (CHANUNAVAIL | CONGESTION | BUSY | NOANSWER - | ANSWER | CANCEL | DONTCALL | TORTURE) -${DYNAMIC_FEATURES} * The list of features (from the [applicationmap] section of - features.conf) to activate during the call, with feature - names separated by '#' characters -${LIMIT_PLAYAUDIO_CALLER} Soundfile for call limits -${LIMIT_PLAYAUDIO_CALLEE} Soundfile for call limits -${LIMIT_WARNING_FILE} Soundfile for call limits -${LIMIT_TIMEOUT_FILE} Soundfile for call limits -${LIMIT_CONNECT_FILE} Soundfile for call limits -${OUTBOUND_GROUP} Default groups for peer channels (as in SetGroup) -* See "show application dial" for more information - -The chanisavail() application sets the following variables: ------------------------------------------------------------ -${AVAILCHAN} * the name of the available channel if one was found -${AVAILORIGCHAN} * the canonical channel name that was used to create the channel -${AVAILSTATUS} * Status of requested channel - -When using macros in the dialplan, these variables are available ---------------------------------------------------------- -${MACRO_EXTEN} * The calling extensions -${MACRO_CONTEXT} * The calling context -${MACRO_PRIORITY} * The calling priority -${MACRO_OFFSET} Offset to add to priority at return from macro - -If you compile with OSP support in the SIP channel, these -variables are used: ---------------------------------------------------------- -${OSPHANDLE} Handle from the OSP Library -${OSPTECH} OSP Technology from Library -${OSPDEST} OSP Destination from Library -${OSPTOKEN} OSP Token to use for call from Library -${OSPRESULTS} Number of OSP results - -____________________________________ -CDR Variables ------------------------------------- - -If the channel has a cdr, that cdr record has it's own set of variables which -can be accessed just like channel variables. The following builtin variables -are available. - -${CDR(clid)} Caller ID -${CDR(src)} Source -${CDR(dst)} Destination -${CDR(dcontext)} Destination context -${CDR(channel)} Channel name -${CDR(dstchannel)} Destination channel -${CDR(lastapp)} Last app executed -${CDR(lastdata)} Last app's arguments -${CDR(start)} Time the call started. -${CDR(answer)} Time the call was answered. -${CDR(end)} Time the call ended. -${CDR(duration)} Duration of the call. -${CDR(billsec)} Duration of the call once it was answered. -${CDR(disposition)} ANSWERED, NO ANSWER, BUSY -${CDR(amaflags)} DOCUMENTATION, BILL, IGNORE etc -${CDR(accountcode)} The channel's account code. -${CDR(uniqueid)} The channel's unique id. -${CDR(userfield)} The channels uses specified field. - - -In addition, you can set your own extra variables with a traditional -Set(CDR(var)=val) to anything you want. - -Certain functional variables may be accessed with ${foo(<args>)}. A list -of these functional variables may be found by typing "show functions" -at the Asterisk CLI. diff --git a/1.2-netsec/doc/app_sms.html b/1.2-netsec/doc/app_sms.html deleted file mode 100644 index 2a076b38b..000000000 --- a/1.2-netsec/doc/app_sms.html +++ /dev/null @@ -1,834 +0,0 @@ -<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN"> -<html> -<head> -</head> -<body> -<h1>* Application SMS</h1> -The SMS module for asterisk was developed by Adrian Kennard, and is an -implementation of the ETSI specification for landline SMS, ETSI ES 201 -912, which is available from www.etsi.org. Landline SMS is starting to -be available in various parts of Europe, and is available from BT in -the UK. However, asterisk would allow gateways to be created in other -locations such as the US, and use of SMS capable phones such as the -Magic Messenger. SMS works using analogue or ISDN lines.<br> -<h2>Background</h2> -Short Message Service (SMS), or <span style="font-style: italic;">texting</span> -is very popular between mobile phones. A message can be sent between -two phones, and normally contains 160 characters. There are ways in -which various types of data can be encoded in a text message such as -ring tones, and small graphic, etc. Text messaging is being used for -voting and competitions, and also SPAM...<br> -<br> -Sending a message involves the mobile phone contacting a message centre -(SMSC) and passing the message to it. The message centre then contacts -the destination mobile to deliver the message. The SMSC is responsible -for storing the message and trying to send it until the destination -mobile is available, or a timeout.<br> -<br> -Landline SMS works in basically the same way. You would normally have a -suitable text capable landline phone, or a separate texting box such as -a Magic Messenger on your phone line. This sends a message to a message -centre your telco provides by making a normal call and sending the data -using 1200 Baud FSK signaling according to the ETSI spec. To receive a -message the message centre calls the line with a specific calling -number, and the text capable phone answers the call and receives the -data using 1200 Baud FSK signaling. This works particularly well in the -UK as the calling line identity is sent before the first ring, so no -phones in the house would ring when a message arrives.<br> -<h2>Typical use with asterisk</h2> -Sending messages from an asterisk box can be used for a variety of -reasons, including notification from any monitoring systems, email -subject lines, etc.<br> -Receiving messages to an asterisk box is typically used just to email -the messages to someone appropriate - we email and texts that are -received to our direct numbers to the appropriate person. Received -messages could also be used to control applications, manage -competitions, votes, post items to IRC, anything.<br> -Using a terminal such as a magic messenger, an asterisk box could ask -as a message centre sending messages to the terminal, which will beep -and pop up the message (and remember 100 or so messages in its memory).<br> -<h2>Terminology</h2> -<table style="text-align: left;" border="1" cellpadding="2" - cellspacing="2"> - <tbody> - <tr> - <td style="vertical-align: top;">SMS<br> - </td> - <td style="vertical-align: top;">Short Message Service<br> - </td> - <td style="vertical-align: top;">i.e. text messages<br> - </td> - </tr> - <tr> - <td style="vertical-align: top;">SMSC<br> - </td> - <td style="vertical-align: top;">Short Message Service Centre<br> - </td> - <td style="vertical-align: top;">The system responsible for -storing and forwarding messages<br> - </td> - </tr> - <tr> - <td style="vertical-align: top;">MO<br> - </td> - <td style="vertical-align: top;">Mobile Originated<br> - </td> - <td style="vertical-align: top;">A message on its way from a -mobile or landline device to the SMSC<br> - </td> - </tr> - <tr> - <td style="vertical-align: top;">MT<br> - </td> - <td style="vertical-align: top;">Mobile Terminated<br> - </td> - <td style="vertical-align: top;">A message on its way from the -SMSC to the mobile or landline device<br> - </td> - </tr> - <tr> - <td style="vertical-align: top;">RX<br> - </td> - <td style="vertical-align: top;">Receive<br> - </td> - <td style="vertical-align: top;">A message coming in to the -asterisk box<br> - </td> - </tr> - <tr> - <td style="vertical-align: top;">TX<br> - </td> - <td style="vertical-align: top;">Transmit<br> - </td> - <td style="vertical-align: top;">A message going out of the -asterisk box<br> - </td> - </tr> - </tbody> -</table> -<h2>Sub address</h2> -When sending a message to a landline, you simply send to the landline -number. In the UK, all of the mobile operators (bar one) understand -sending messages to landlines and pass the messages to the BTText -system for delivery to the landline.<br> -<br> -The specification for landline SMS allows for the possibility of more -than one device on a single landline. These can be configured with <span - style="font-style: italic;">Sub addresses</span> which are a single -digit. To send a message to a specific device the message is sent to -the landline number with an extra digit appended to the end. The telco -can define a default sub address (9 in the UK) which is used when the -extra digit is not appended to the end. When the call comes in, part of -the calling line ID is the sub address, so that only one device on the -line answers the call and receives the message.<br> -<br> -Sub addresses also work for outgoing messages. Part of the number -called by the device to send a message is its sub address. Sending from -the default sub address (9 in the UK) means the message is delivered -with the <span style="font-style: italic;">sender </span>being the -normal landline number. Sending from any other sub address makes the <span - style="font-style: italic;">sender</span> the landline number with an -extra digit on the end.<br> -<br> -Using asterisk, you can make use of the sub addresses for sending and -receiving messages. Using DDI (DID, i.e. multiple numbers on the line -on ISDN) you can also make use of many different numbers for SMS.<br> -<h2>Build / installation</h2> -<span style="font-weight: bold;">app_sms.c</span> is included in the -latest cvs. It lives in the asterisk source <span - style="font-weight: bold;">apps</span> directory and is included in -the object list (<span style="font-weight: bold;">app_sms.so</span>) in -<span style="font-weight: bold;">apps/Makefile</span>.<br> -<span style="font-weight: bold;">smsq.c</span> is a stand alone helper -application which is used to send SMSs from the command line. It uses -the <span style="font-weight: bold;">popt</span> library. A line for -your make file is:-<br> -<pre>smsq: smsq.c<br> cc -O -o smsq smsq.c -lpopt<br></pre> -<span style="font-family: monospace;"></span> -<h2>extensions.conf</h2> -The following contexts are recommended.<br> -<pre>; Mobile Terminated, RX. This is used when an incoming call from the SMS arrives, with the queue (called number and sub address) in ${EXTEN}<br>; Running an app after receipt of the text allows the app to find all messages in the queue and handle them, e.g. email them.<br>; The app may be something like smsq --process=somecommand --queue=${EXTEN} to run a command for each received message<br>; See below for usage<br>[smsmtrx]<br>exten = _X.,1, SMS(${EXTEN}|a)<br>exten = _X.,2,System("someapptohandleincomingsms ${EXTEN}")<br>exten = _X.,3,Hangup<br><br>; Mobile originated, RX. This is receiving a message from a device, e.g. a Magic Messenger on a sip extension<br>; Running an app after receipt of the text allows the app to find all messages in the queue and handle then, e.g. sending them to the public SMSC<br>; The app may be something like smsq --process=somecommand --queue=${EXTEN} to run a command for each received message<br>; See below for example usage<br>[smsmorx]<br>exten = _X.,1, SMS(${EXTEN}|sa)<br>exten = _X.,2,System("someapptohandlelocalsms ${EXTEN}")<br>exten = _X.,3,Hangup<span - style="font-family: sans-serif;"></span><span - style="font-family: sans-serif;"></span></pre> -<span style="font-family: sans-serif;"></span><span - style="font-weight: bold;">smsmtrx</span> is normally accessed by an -incoming call from the SMSC. In the UK this call is from a CLI of -080058752X0 where X is the sub address. As such a typical usage in the -extensions.conf at the point of handling an incoming call is:-<br> -<pre>exten = _X./8005875290,1,Goto(smsmtrx,${EXTEN},1)<br>exten = _X./_80058752[0-8]0,1,Goto(smsmtrx,${EXTEN}-${CALLERIDNUM:8:1},1)<br></pre> -Alternatively, if you have the correct national prefix on incoming CLI, -e.g. using zaphfc, you might use:-<br> -<pre>exten = _X./08005875290,1,Goto(smsmtrx,${EXTEN},1)<br>exten = _X./_080058752[0-8]0,1,Goto(smsmtrx,${EXTEN}-${CALLERIDNUM:9:1},1)</pre> -<span style="font-weight: bold;">smsmorx</span> is normally accessed by -a call from a local sip device connected to a Magic Messenger. It could -however by that you are operating asterisk as a message centre for -calls from outside. Either way, you look at the called number and goto -smsmorx. In the UK, the SMSC number that would be dialed is 1709400X -where X is the caller sub address. As such typical usage in -extension.config at the point of handling a call from a sip phone is:-<br> -<pre>exten = 17094009,1,Goto(smsmorx,${CALLERIDNUM},1)<br>exten = _1709400[0-8],1,Goto(smsmorx,${CALLERIDNUM}-{EXTEN:7:1},1)<br></pre> -<h2>Using smsq</h2> -<span style="font-weight: bold;">smsq</span> is a simple helper -application designed to make it easy to send messages from a command -line. it is intended to run on the asterisk box and have direct access -to the queue directories for SMS and for asterisk.<br> -<br> -In its simplest form you can send an SMS by a command such as <br> -<br> -smsq 0123456789 This is a test to 0123456789<br> -<br> -This would create a queue file for a mobile originated TX message in -queue 0 to send the text "This is a test to 0123456789" to 0123456789. -It would then place a file in the /var/spool/asterisk/outgoing -directory to initiate a call to 17094009 (the default message centre in -smsq) attached to application SMS with argument of the queue name (0).<br> -<br> -Normally smsq will queue a message ready to send, and will then create -a file in the asterisk outgoing directory causing asterisk to actually -connect to the message centre or device and actually send the pending -message(s).<br> -<br> -Using --process, smsq can however be used on received queues to run a -command for each file (matching the queue if specified) with various -environment variables set based on the message (see below);<br> -<br> -smsq options:-<br> -<br> -<table style="text-align: left;" border="1" cellpadding="2" - cellspacing="2"> - <tbody> - <tr> - <td style="vertical-align: top;">--help</td> - <td style="vertical-align: top;"><br> - </td> - <td style="vertical-align: top;">Show help text<br> - </td> - </tr> - <tr> - <td style="vertical-align: top;">--usage<br> - </td> - <td style="vertical-align: top;"><br> - </td> - <td style="vertical-align: top;">Show usage<br> - </td> - </tr> - <tr> - <td style="vertical-align: top;">--queue<br> - </td> - <td style="vertical-align: top;">-q<br> - </td> - <td style="vertical-align: top;">Specify a specific queue<br> -In no specified, messages are queued under queue "0"<br> - </td> - </tr> - <tr> - <td style="vertical-align: top;">--da<br> - </td> - <td style="vertical-align: top;">-d<br> - </td> - <td style="vertical-align: top;">Specify destination address<br> - </td> - </tr> - <tr> - <td style="vertical-align: top;">--oa<br> - </td> - <td style="vertical-align: top;">-o<br> - </td> - <td style="vertical-align: top;">Specify originating address<br> -This also implies that we are generating a mobile terminated message<br> - </td> - </tr> - <tr> - <td style="vertical-align: top;">--ud<br> - </td> - <td style="vertical-align: top;">-m<br> - </td> - <td style="vertical-align: top;">Specify the actual message<br> - </td> - </tr> - <tr> - <td style="vertical-align: top;">--ud-file<br> - </td> - <td style="vertical-align: top;">-f<br> - </td> - <td style="vertical-align: top;">Specify a file to be read for -the context of the message<br> -A blank filename (e.g. --ud-file= on its own) means read stdin. Very -useful when using via ssh where command line parsing could mess up the -message.<br> - </td> - </tr> - <tr> - <td style="vertical-align: top;">--mt<br> - </td> - <td style="vertical-align: top;">-t<br> - </td> - <td style="vertical-align: top;">Mobile terminated message to be -generated<br> - </td> - </tr> - <tr> - <td style="vertical-align: top;">--mo<br> - </td> - <td style="vertical-align: top;"><br> - </td> - <td style="vertical-align: top;">Mobile originated message to be -generated<br> -Default<br> - </td> - </tr> - <tr> - <td style="vertical-align: top;">--tx<br> - </td> - <td style="vertical-align: top;"><br> - </td> - <td style="vertical-align: top;">Transmit message<br> -Default<br> - </td> - </tr> - <tr> - <td style="vertical-align: top;">--rx<br> - </td> - <td style="vertical-align: top;">-r<br> - </td> - <td style="vertical-align: top;">Generate a message in the -receive queue<br> - </td> - </tr> - <tr> - <td style="vertical-align: top;">--UTF-8<br> - </td> - <td style="vertical-align: top;"><br> - </td> - <td style="vertical-align: top;">Treat the file as UTF-8 encoded -(default) </td> - </tr> - <tr> - <td style="vertical-align: top;">--UCS-1<br> - </td> - <td style="vertical-align: top;"><br> - </td> - <td style="vertical-align: top;">Treat the file as raw 8 bit -UCS-1 data, not UTF-8 encoded<br> - </td> - </tr> - <tr> - <td style="vertical-align: top;">--UCS-2<br> - </td> - <td style="vertical-align: top;"><br> - </td> - <td style="vertical-align: top;">Treat the file as raw 16 bit -bigendian USC-2 data<br> - </td> - </tr> - <tr> - <td style="vertical-align: top;">--process<br> - </td> - <td style="vertical-align: top;"><br> - </td> - <td style="vertical-align: top;">Specific a command to process -for each file in the queue<br> -Implies --rx and --mt if not otherwise specified.<br> -Sets environment variables for every possible variable, -and also ud, ud8 (USC-1 hex), and ud16 (USC-2 hex) for each call. -Removes files.<br> - </td> - </tr> - <tr> - <td style="vertical-align: top;">--motx-channel<br> - </td> - <td style="vertical-align: top;"><br> - </td> - <td style="vertical-align: top;">Specify the channel for motx -calls<br> -May contain X to use sub address based on queue name or may be full -number<br> -Default is Local/1709400X<br> - </td> - </tr> - <tr> - <td style="vertical-align: top;">--motx-callerid<br> - </td> - <td style="vertical-align: top;"><br> - </td> - <td style="vertical-align: top;">Specify the caller ID for motx -calls<br> -The default is the queue name without -X suffix<br> - </td> - </tr> - <tr> - <td style="vertical-align: top;">--motx-wait<br> - </td> - <td style="vertical-align: top;"><br> - </td> - <td style="vertical-align: top;">Wait time for motx call<br> -Default 10<br> - </td> - </tr> - <tr> - <td style="vertical-align: top;">--motx-delay<br> - </td> - <td style="vertical-align: top;"><br> - </td> - <td style="vertical-align: top;">Retry time for motx call<br> -Default 1<br> - </td> - </tr> - <tr> - <td style="vertical-align: top;">--motx-retries<br> - </td> - <td style="vertical-align: top;"><br> - </td> - <td style="vertical-align: top;">Retries for motx call<br> -Default 10<br> - </td> - </tr> - <tr> - <td style="vertical-align: top;">--mttx-channel<br> - </td> - <td style="vertical-align: top;"><br> - </td> - <td style="vertical-align: top;">Specify the channel for mttx -calls<br> -Default is Local/ and the queue name without -X suffix<br> - </td> - </tr> - <tr> - <td style="vertical-align: top;">--mtttx-callerid<br> - </td> - <td style="vertical-align: top;"><br> - </td> - <td style="vertical-align: top;">Specify the callerid for mttx -calls<br> -May include X to use sub address based on queue name or may be full -number<br> -Default is 080058752X0<br> - </td> - </tr> - <tr> - <td style="vertical-align: top;">--mttx-wait<br> - </td> - <td style="vertical-align: top;"><br> - </td> - <td style="vertical-align: top;">Wait time for mttx call<br> -Default 10<br> - </td> - </tr> - <tr> - <td style="vertical-align: top;">--mttx-delay<br> - </td> - <td style="vertical-align: top;"><br> - </td> - <td style="vertical-align: top;">Retry time for mttx call<br> -Default 30<br> - </td> - </tr> - <tr> - <td style="vertical-align: top;">--mttx-retries<br> - </td> - <td style="vertical-align: top;"><br> - </td> - <td style="vertical-align: top;">Retries for mttx call<br> -Default 100<br> - </td> - </tr> - <tr> - <td style="vertical-align: top;">--default-sub-address<br> - </td> - <td style="vertical-align: top;"><br> - </td> - <td style="vertical-align: top;">The default sub address assumed -(e.g. for X in CLI and dialled numbers as above) when none added (-X) -to queue<br> -Default 9<br> - </td> - </tr> - <tr> - <td style="vertical-align: top;">--no-dial<br> - </td> - <td style="vertical-align: top;">-x<br> - </td> - <td style="vertical-align: top;">Create queue, but do not dial to -send message<br> - </td> - </tr> - <tr> - <td style="vertical-align: top;">--no-wait<br> - </td> - <td style="vertical-align: top;"><br> - </td> - <td style="vertical-align: top;">Do not wait if a call appears to -be in progress<br> -This could have a small window where a mesdsage is queued but not sent, -so regular calls to smsq should be done to pick up any missed messages<br> - </td> - </tr> - <tr> - <td style="vertical-align: top;">--concurrent<br> - </td> - <td style="vertical-align: top;"><br> - </td> - <td style="vertical-align: top;">How many concurrent calls to -allow (per queue), default 1<br> - </td> - </tr> - <tr> - <td style="vertical-align: top;">--mr<br> - </td> - <td style="vertical-align: top;">-n<br> - </td> - <td style="vertical-align: top;">Message reference<br> - </td> - </tr> - <tr> - <td style="vertical-align: top;">--pid<br> - </td> - <td style="vertical-align: top;">-p<br> - </td> - <td style="vertical-align: top;">Protocol ID<br> - </td> - </tr> - <tr> - <td style="vertical-align: top;">--dcs<br> - </td> - <td style="vertical-align: top;"><br> - </td> - <td style="vertical-align: top;">Data coding scheme<br> - </td> - </tr> - <tr> - <td style="vertical-align: top;">--udh<br> - </td> - <td style="vertical-align: top;"><br> - </td> - <td style="vertical-align: top;">Specific hex string of user data -header specified (not including the initial length byte)<br> -May be a blank string to indicate header is included in the user data -already but user data header indication to be set.<br> - </td> - </tr> - <tr> - <td style="vertical-align: top;">--srr<br> - </td> - <td style="vertical-align: top;"><br> - </td> - <td style="vertical-align: top;">Status report requested<br> - </td> - </tr> - <tr> - <td style="vertical-align: top;">--rp<br> - </td> - <td style="vertical-align: top;"><br> - </td> - <td style="vertical-align: top;">Return path requested<br> - </td> - </tr> - <tr> - <td style="vertical-align: top;">--vp<br> - </td> - <td style="vertical-align: top;"><br> - </td> - <td style="vertical-align: top;">Specify validity period (seconds)<br> - </td> - </tr> - <tr> - <td style="vertical-align: top;">--scts<br> - </td> - <td style="vertical-align: top;"><br> - </td> - <td style="vertical-align: top;">Specify timestamp -(YYYY-MM-DDTHH:MM:SS)<br> - </td> - </tr> - <tr> - <td style="vertical-align: top;">--spool-dir<br> - </td> - <td style="vertical-align: top;"><br> - </td> - <td style="vertical-align: top;">Spool dir (in which sms and -outgoing are found)<br> -Default /var/spool/asterisk<br> - </td> - </tr> - </tbody> -</table> -<p>Other arguments starting '-' or '--' are invalid and will cause an -error. Any trailing arguments are processed as follows:-<br> -</p> -<ul> - <li>If the message is mobile originating and no destination address -has been specified, then the first argument is assumed to be a -destination address</li> - <li>If the message is mobile terminating and no destination address -has been specified, then the first argument is assumed to be the queue -name</li> - <li>If there is no user data, or user data file specified, then any -following arguments are assumed to be the message, which are -concatenated.</li> - <li>If no user data is specified, then no message is sent. However, -unless --no-dial is specified, smsq checks for pending messages and -generates an outgoing anyway</li> -</ul> -Note that when smsq attempts to make a file in -/var/spool/asterisk/outgoing, it checks if there is already a call -queued for that queue. It will try several filenames, up to the ---concorrent setting. If these files -exists, then this means asterisk is already queued to send all messages -for that queue, and so asterisk should pick up the message just queued. -However, this alone could create a race condition, so if the files -exist then smsq will wait up to 3 seconds to confirm it still exists or -if the queued messages have been sent already. -The --no-wait turns off this behaviour. Basically, this means that if -you have a lot of messages to send all at -once, asterisk will not make unlimited concurrent calls to the same -message centre or device for the same queue. This is because it is -generally more efficient to make one call and send all of the messages -one after the other.<br> -<br> -smsq can be used with no arguments, or with a queue name only, and it -will check for any pending messages and cause an outgoing if there are -any. It only sets up one outgoing call at a time based on the first -queued message it finds. A outgoing call will normally send all queued -messages for that queue. One way to use smsq would be to run with no -queue name (so any queue) every minute or every few seconds to send -pending message. This is not normally necessary unless --no-dial is -selected. Note that smsq does only check motx or mttx depending on the -options selected, so it would need to be called twice as a general -check.<br> -<br> -UTF-8 is used to parse command line arguments for user data, and is the -default when reading a file. If an invalid UTF-8 sequence is found, it -is treated as UCS-1 data (i.e, as is).<br> -<br> -The --process option causes smsq to scan the specified queue (default -is mtrx) for messages (matching the queue specified, or any if queue -not specified) and run a command and delete the file. The command is -run with a number of environment variables set as follows. Note that -these are unset if not needed and not just taken from the calling -environment. This allows simple processing of incoming messages<br> -<br> -<table style="text-align: left;" border="1" cellpadding="2" - cellspacing="2"> - <tbody> - <tr> - <td style="vertical-align: top;">$queue<br> - </td> - <td style="vertical-align: top;">Set if a queue specified<br> - </td> - </tr> - <tr> - <td style="vertical-align: top;">$?srr<br> - </td> - <td style="vertical-align: top;">srr is set (to blank) if srr -defined and has value 1.<br> - </td> - </tr> - <tr> - <td style="vertical-align: top;">$?rp<br> - </td> - <td style="vertical-align: top;">rp is set (to blank) if rp -defined and has value 1.<br> - </td> - </tr> - <tr> - <td style="vertical-align: top;">$ud<br> - </td> - <td style="vertical-align: top;">User data, UTF-8 encoding, -including any control characters, but with nulls stripped out<br> -Useful for the content of emails, for example, as it includes any -newlines, etc.<br> - </td> - </tr> - <tr> - <td style="vertical-align: top;">$ude<br> - </td> - <td style="vertical-align: top;">User data, escaped UTF-8, -including all characters, but control characters \n, \r, \t, \f, \xxx -and \ is escaped as \\<br> -Useful fGuaranteed one line printable text, so useful in Subject lines -of emails, etc<br> - </td> - </tr> - <tr> - <td style="vertical-align: top;">$ud8<br> - </td> - <td style="vertical-align: top;">Hex UCS-1 coding of user data (2 -hex digits per character)<br> -Present only if all user data is in range U+0000 to U+00FF<br> - </td> - </tr> - <tr> - <td style="vertical-align: top;">$ud16<br> - </td> - <td style="vertical-align: top;">Hex UCS-2 coding of user data (4 -hex digits per chartacter)<br> - </td> - </tr> - <tr> - <td style="vertical-align: top;"><span style="font-style: italic;">other</span><br> - </td> - <td style="vertical-align: top;">Other fields set using their -field name, e.g. mr, pid, dcs, etc. udh is a hex byte string<br> - </td> - </tr> - </tbody> -</table> -<h2>File formats</h2> -By default all queues are held in a director /var/spool/asterisk/sms. -Within this directory are sub directories mtrx, mttx, morx, motx which -hold the received messages and the messages ready to send. Also, -/var/log/asterisk/sms is a log file of all messages handled.<br> -<br> -The file name in each queue directory starts with the queue parameter -to SMS which is normally the CLI used for an outgoing message or the -called number on an incoming message, and may have -X (X being sub -address) appended. If no queue ID is known, then 0 is used by smsq by -default. After this is a dot, and then any text. Files are scanned for -matching queue ID and a dot at the start. This means temporary files -being created can be given a different name not starting with a queue -(we recommend a . on the start of the file name for temp files).<br> -<br> -Files in these queues are in the form of a simple text file where each -line starts with a keyword and an = and then data. udh and ud have -options for hex encoding, see below.<br> -<br> -UTF-8. The user data (ud) field is treated as being UTF-8 encoded -unless the DCS is specified indicating 8 bit formart. If 8 bit format -is specified then the user data is sent as is.<br> -<br> -The keywords are as -follows:-<br> -<table style="text-align: left;" border="1" cellpadding="2" - cellspacing="2"> - <tbody> - <tr> - <td style="vertical-align: top;">oa</td> - <td style="vertical-align: top;">Originating address<br> -The phone number from which the message came<br> -Present on mobile terminated messages and is the CLI for morx messages<br> - </td> - </tr> - <tr> - <td style="vertical-align: top;">da<br> - </td> - <td style="vertical-align: top;">Destination Address<br> -The phone number to which the message is sent<br> -Present on mobile originated messages<br> - </td> - </tr> - <tr> - <td style="vertical-align: top;">scts<br> - </td> - <td style="vertical-align: top;">The service centre time stamp<br> -Format YYYY-MM-DDTHH:MM:SS<br> -Present on mobile terminated messages<br> - </td> - </tr> - <tr> - <td style="vertical-align: top;">pid<br> - </td> - <td style="vertical-align: top;">One byte decimal protocol ID<br> -See GSM specs for more details<br> -Normally 0 or absent<br> - </td> - </tr> - <tr> - <td style="vertical-align: top;">dcs<br> - </td> - <td style="vertical-align: top;">One byte decimal data coding -scheme<br> -If omitted, a sensible default is used (see below)<br> -See GSM specs for more details<br> - </td> - </tr> - <tr> - <td style="vertical-align: top;">mr<br> - </td> - <td style="vertical-align: top;">One byte decimal message -reference<br> -Present on mobile originated messages, added by default if absent<br> - </td> - </tr> - <tr> - <td style="vertical-align: top;">srr<br> - </td> - <td style="vertical-align: top;">0 or 1 for status report request<br> -Does not work in UK yet, not implemented in app_sms yet<br> - </td> - </tr> - <tr> - <td style="vertical-align: top;">rp<br> - </td> - <td style="vertical-align: top;">0 or 1 return path<br> -See GSM specs for details<br> - </td> - </tr> - <tr> - <td style="vertical-align: top;">vp<br> - </td> - <td style="vertical-align: top;">Validity period in seconds<br> -Does not work in UK yet<br> - </td> - </tr> - <tr> - <td style="vertical-align: top;">udh<br> - </td> - <td style="vertical-align: top;">Hex string of user data header -prepended to the SMS contents, excluding initial length byte.<br> -Consistent with ud, this is specified as udh# rather than udh=<br> -If blank, this means that the udhi flag will be set but any user data -header must be in the ud field<br> - </td> - </tr> - <tr> - <td style="vertical-align: top;">ud<br> - </td> - <td style="vertical-align: top;">User data, may be text, or hex, -see below<br> - </td> - </tr> - </tbody> -</table> -<br> -udh is specified as as udh# followed by hex (2 hex digits per byte). If -present, then the user data header indicator bit is set, and the length -plus the user data header is added to the start of the user data, with -padding if necessary (to septet boundary in 7 bit format).<br> -<br> -User data can hold an USC character codes U+0000 to U+FFFF. Any other -characters are coded as U+FEFF<br> -ud can be specified as ud= followed by UTF-8 encoded text if it -contains no control characters, i.e. only (U+0020 to U+FFFF). Any -invalid UTF-8 sequences are treated as is (U+0080-U+00FF).<br> -ud can also be specified as ud# followed by hex (2 hex digits per byte) -containing characters U+0000 to U+00FF only.<br> -ud can also be specified as ud## followed by hex (4 hex digits per -byte) containing UCS-2 characters.<br> -When written by app_sms (e.g. incoming messages), the file is written -with ud= if it can be (no control characters). If it cannot, the a -comment line ;ud= is used to show the user data for human readability -and ud# or ud## is used.<br> -<h2>Delivery reports</h2> -The SMS specification allows for delivery reports. These are requested -using the srr bit. However, as these do not work in the UK yet they are -not fully implemented in this application. If anyone has a telco that -does implement these, please let me know. BT in the UK have a non -standard way to do this by starting the message with *0#, and so this -application may have a UK specific bodge in the near future to handle -these.<br> -<br> -The main changes that are proposed for delivery report handling are :-<br> -<ul> - <li>New queues for sent messages, one file for each destination -address and message reference.</li> - <li>New field in message format, user reference, allowing -applications to tie up their original message with a report.</li> - <li>Handling of the delivery confirmation/rejection and connecting to -the outgoing message - the received message file would then have fields -for the original outgoing message and user reference allowing -applications to handle confirmations better.<br> - </li> -</ul> -<br> -</body> -</html> diff --git a/1.2-netsec/doc/apps.txt b/1.2-netsec/doc/apps.txt deleted file mode 100644 index c9696a1a5..000000000 --- a/1.2-netsec/doc/apps.txt +++ /dev/null @@ -1,10 +0,0 @@ -Asterisk applications register themselves with ast_application_register. -They should have a short, unique name, and an exec function which takes -as its arguments a channel and some data that might be useful for callback -stuff. Remember to keep track of how many and which channels are using -your application so that should the module need to be unloaded -(particularly force unloaded), you will be able to ast_softhangup all the -channels. An application should *never* call ast_hangup on the channel -that it is running on (although it could conceivably hang up other -channels that it allocates). See app_playback.c as an example of a simple -application. diff --git a/1.2-netsec/doc/cdr.txt b/1.2-netsec/doc/cdr.txt deleted file mode 100644 index 02dc7d1db..000000000 --- a/1.2-netsec/doc/cdr.txt +++ /dev/null @@ -1,171 +0,0 @@ -Call data records can be stored in many different databases or even CSV text. - -MSSQL: Asterisk can currently store CDRs into an MSSQL database in - two different ways: cdr_odbc.c or cdr_tds.c - - Call Data Records can be stored using unixODBC (which requires - the FreeTDS package) [cdr_odbc.c] or directly by using just the - FreeTDS package [cdr_tds.c] The following provide some - examples known to get asterisk working with mssql. - NOTE: Only choose one db connector. - - ODBC [cdr_odbc.c]: - Compile, configure, and install the latest unixODBC package: - tar -zxvf unixODBC-2.2.9.tar.gz && - cd unixODBC-2.2.9 && - ./configure --sysconfdir=/etc --prefix=/usr --disable-gui && - make && - make install - - Compile, configure, and install the latest FreeTDS package: - tar -zxvf freetds-0.62.4.tar.gz && - cd freetds-0.62.4 && - ./configure --prefix=/usr --with-tdsver=7.0 \ - --with-unixodbc=/usr/lib && - make && - make install - - Compile, or recompile, asterisk so that it will now add support - for cdr_odbc.c - - make clean && - make update && - make && - make install - - Setup odbc configuration files. These are working examples - from my system. You will need to modify for your setup. - You are not required to store usernames or passwords here. - - /etc/odbcinst.ini - [FreeTDS] - Description = FreeTDS ODBC driver for MSSQL - Driver = /usr/lib/libtdsodbc.so - Setup = /usr/lib/libtdsS.so - FileUsage = 1 - - /etc/odbc.ini - [MSSQL-asterisk] - description = Asterisk ODBC for MSSQL - driver = FreeTDS - server = 192.168.1.25 - port = 1433 - database = voipdb - tds_version = 7.0 - language = us_english - - Only install one database connector. Do not confuse asterisk - by using both ODBC (cdr_odbc.c) and FreeTDS (cdr_tds.c). - This command will erase the contents of cdr_tds.conf - - [ -f /etc/asterisk/cdr_tds.conf ] > /etc/asterisk/cdr_tds.conf - - NOTE: unixODBC requires the freeTDS package, but asterisk does - not call freeTDS directly. - - Setup cdr_odbc configuration files. These are working samples - from my system. You will need to modify for your setup. Define - your usernames and passwords here, secure file as well. - - /etc/asterisk/cdr_odbc.conf - [global] - dsn=MSSQL-asterisk - username=voipdbuser - password=voipdbpass - loguniqueid=yes - - And finally, create the 'cdr' table in your mssql database. - - CREATE TABLE cdr ( - [calldate] [datetime] NOT NULL , - [clid] [varchar] (80) NOT NULL , - [src] [varchar] (80) NOT NULL , - [dst] [varchar] (80) NOT NULL , - [dcontext] [varchar] (80) NOT NULL , - [channel] [varchar] (80) NOT NULL , - [dstchannel] [varchar] (80) NOT NULL , - [lastapp] [varchar] (80) NOT NULL , - [lastdata] [varchar] (80) NOT NULL , - [duration] [int] NOT NULL , - [billsec] [int] NOT NULL , - [disposition] [varchar] (45) NOT NULL , - [amaflags] [int] NOT NULL , - [accountcode] [varchar] (20) NOT NULL , - [uniqueid] [varchar] (32) NOT NULL , - [userfield] [varchar] (255) NOT NULL - ) - - Start asterisk in verbose mode, you should see that asterisk - logs a connection to the database and will now record every - call to the database when it's complete. - - TDS [cdr_tds.c]: - Compile, configure, and install the latest FreeTDS package: - tar -zxvf freetds-0.62.4.tar.gz && - cd freetds-0.62.4 && - ./configure --prefix=/usr --with-tdsver=7.0 - make && - make install - - Compile, or recompile, asterisk so that it will now add support - for cdr_tds.c (Currently only asterisk SVN supports cdr_tds.c) - - make clean && - make update && - make && - make install - - Only install one database connector. Do not confuse asterisk - by using both ODBC (cdr_odbc.c) and FreeTDS (cdr_tds.c). - This command will erase the contents of cdr_odbc.conf - - [ -f /etc/asterisk/cdr_odbc.conf ] > /etc/asterisk/cdr_odbc.conf - - Setup cdr_tds configuration files. These are working samples - from my system. You will need to modify for your setup. Define - your usernames and passwords here, secure file as well. - - /etc/asterisk/cdr_tds.conf - [global] - hostname=192.168.1.25 - port=1433 - dbname=voipdb - user=voipdbuser - password=voipdpass - charset=BIG5 - - And finally, create the 'cdr' table in your mssql database. - - CREATE TABLE cdr ( - [accountcode] [varchar] (20) NULL , - [src] [varchar] (80) NULL , - [dst] [varchar] (80) NULL , - [dcontext] [varchar] (80) NULL , - [clid] [varchar] (80) NULL , - [channel] [varchar] (80) NULL , - [dstchannel] [varchar] (80) NULL , - [lastapp] [varchar] (80) NULL , - [lastdata] [varchar] (80) NULL , - [start] [datetime] NULL , - [answer] [datetime] NULL , - [end] [datetime] NULL , - [duration] [int] NULL , - [billsec] [int] NULL , - [disposition] [varchar] (20) NULL , - [amaflags] [varchar] (16) NULL , - [uniqueid] [varchar] (32) NULL - ) - - Start asterisk in verbose mode, you should see that asterisk - logs a connection to the database and will now record every - call to the database when it's complete. - -MYSQL: - -PGSQL: - -SQLLITE: - -------------------------------------------------------------------------------- - -08/02/2004 : Duane Cox <dcox@illicom.net> - added mssql information diff --git a/1.2-netsec/doc/channel.txt b/1.2-netsec/doc/channel.txt deleted file mode 100644 index 4cef6a4c6..000000000 --- a/1.2-netsec/doc/channel.txt +++ /dev/null @@ -1,40 +0,0 @@ -Implementing a Channel -====================== - -* What is a channel? - -A channel is a unit which brings in a call to the Asterisk PBX. A channel -could be connected to a real telephone (like the Internet Phone Jack) or -to a logical call (like an Internet phone call). Asterisk makes no -distinction between "FXO" and "FXS" style channels (that is, it doesn't -distinguish between telephone lines and telephones). - -Every call is placed or received on a distinct channel. Asterisk uses a -channel driver (typically named chan_xxx.so) to support each type of -hardware. - -* What do I need to create a channel? - -In order to support a new piece of hardware you need to write a channel -driver. The easiest way to do so is to look at an existing channel driver -and model your own code after it. - -* What's the general architecture? - -Typically, a channel reads a configuration file on startup which tells it -something about the hardware it's going to be servicing. Then, it -launches a thread which monitors all the idle channels (See the chan_modem -or the chan_ixj for an example of this). When a "RING" or equivalent is -detected, the monitoring thread should allocate a channel structure and -assign all the callbacks to it (see ixj_new, for example), and then call -ast_pbx_start on that channel. ast_pbx_start will launch a new thread to -handle the channel as long as the call is up, so once pbx_start has -successfully been run, the monitor should no longer monitor that channel. -The PBX thread will use the channel, reading, writing, calling, etc., and -multiplexing that channel with others using select() on the channel's -file descriptor (if your channel doesn't have an associated file -descriptor, you'll need to emulate one somehow, perhaps along the lines of -what the translator API does with its channel. - -When the PBX is finished with the line, it will hang up the line, at which -point it the hardware should again be monitored by the monitoring thread. diff --git a/1.2-netsec/doc/extensions.txt b/1.2-netsec/doc/extensions.txt deleted file mode 100644 index bab08d319..000000000 --- a/1.2-netsec/doc/extensions.txt +++ /dev/null @@ -1,58 +0,0 @@ -The Asterisk dialplan -===================== - -The Asterisk dialplan is divided into contexts. A context is simply a group -of extensions. For each "line" that should be able to be called, an extension -must be added to a context. Then, you configure the calling "line" to have -access to this context. - -If you change the dialplan, you can use the Asterisk CLI command -"extensions reload" to load the new dialplan without disrupting -service in your PBX. - -Extensions are routed according to priority and may be based on any set -of characters (a-z), digits, #, and *. Please note that when matching a -pattern, "N", "X", and "Z" are interpreted as classes of digits. - -For each extension, several actions may be listed and must be given a unique -priority. When each action completes, the call continunes at the next priority -(except for some modules which use explicitly GOTO's). - -When each action completes, it generally moves to the next priority (except for -some modules which use explicitly GOTO's. - -Extensions frequently have data they pass to the executing application -(most frequently a string). You can see the available dialplan applications -by entering the "show applications" command in the CLI. - -In this version of Asterisk, dialplan functions are added. These can -be used as arguments to any application. For a list of the installed -functions in your Asterisk, use the "show functions" command. - -* Example dial plan - -The example dial plan, in the configs/extensions.conf.sample file -is installed as extensions.conf if you run "make samples" after -installation of Asterisk. This file includes many more instructions -and examples than this file, so it's worthwile to read it. - -* Special extensions - -There are some extensions with important meanings: - - s: What to do when an extension context is entered (unless - overridden by the low level channel interface) - This is used in macros, and some special cases. - "s" is not a generic catch-all wildcard extension. - i: What to do if an invalid extension is entered - h: The hangup extension, executed at hangup - t: What to do if nothing is entered in the requisite amount - of time. - T: This is the extension that is executed when the 'absolute' - timeout is reached. See "show function TIMEOUT" for more - information on setting timeouts. - -And finally, the extension context "default" is used when either a) an -extension context is deleted while an extension is in use, or b) a specific -starting extension handler has not been defined (unless overridden by the -low level channel interface). diff --git a/1.2-netsec/doc/iax.txt b/1.2-netsec/doc/iax.txt deleted file mode 100644 index 48bb3e10e..000000000 --- a/1.2-netsec/doc/iax.txt +++ /dev/null @@ -1,55 +0,0 @@ -Inter-Asterisk eXchange Protocol -================================ - -Usage: -====== -The format for the dialing string on Asterisk is: -IAX/[user@]peer[:exten[@context]] - -(Note, []'s denote optional fields). The peer is either an IP address -or a peer as specified in the /etc/asterisk/iax.conf file. Exten is -an optional requested extension (otherwise "s" will be used), and -"context" is an optional context to request. The user is an optional -username specified in the peer's iax.conf. If the user is not specified, -the peer will select one. - -Protocol and rationale: -======================= -IAX is a simple, low overhead and low bandwidth VoIP protocol designed to -allow multiple Asterisk PBX's to communicate with one another without -the overhead of more complex protocols like H.323. Payload is sent with -a header overhead of only 4 octets. Control functions (and one payload packet -per minute or so) is sent with a more complex header of 12 octets. - -IAX is slightly stateful. - -IAX contains two kinds of packets: The full header packet type, which -contains much information about the frame, in addition to its contents, -and the mini header type, which is used only for non-reliable voice -packet delivery. - -All packets are immediately transmitted. Packets are received, but not -delivered to the actual channels until a given time quantum has passed, in -order to try to eliminate jitter. - -All full header packets must be ackd (except, obviously for the ACK packets -themselves and not so obviously for hangup packets). The "timestamp" field of -ack packets is not the normal offset, but rather a quote of the timestamp as -included with the original packet that you're acking, and likewise the -seqno field is the seqno of the packet you're acking, not your own seqno, -and you do not increment your own sequence number. ACKing is based on the -sequence number. - -See iax.h for a description of the frame formats. - -IAX internal frames use the AST_FRAME_IAX type. The subclass of these -frames is the IAX control number, as seen in iax.h. The first frame sent -must be an AST_FRAME_IAX with the control AST_IAX_CONTROL_NEW. - -The AST_IAX_CONTROL_NEW establishes a new connection. - -The first frame sent MUST be an AST_CONTROL_NEW to start a connection. - -IAX connnections may require authentication using either simple plaintext -passwords or an md5 challenge/response pair. - diff --git a/1.2-netsec/doc/linkedlists.README b/1.2-netsec/doc/linkedlists.README deleted file mode 100644 index 340933548..000000000 --- a/1.2-netsec/doc/linkedlists.README +++ /dev/null @@ -1,98 +0,0 @@ -As of 2004-12-23, this documentation is no longer maintained. The doxygen documentation -generated from linkedlists.h should be referred to in its place, as it is more complete -and better maintained. - -2nd version, implemented as macros. - - include <asterisk/linkedlists.h> - -AST_LIST_ENTRY declares pointers inside the object structure : - - struct ast_var_t { - char *name; - char *value; - AST_LIST_ENTRY(ast_var_t) listpointers; - }; - -AST_LIST_HEAD declares a head structure, which is initialized -to AST_LIST_HEAD_NULL : - - AST_LIST_HEAD(head, ast_var_t) head - -Next, we declare a pointer to this structure : - - struct headtype *headp = head; - -AST_LIST_INIT initializes the head pointer to a null value - - AST_LIST_INIT(headp); - -AST_LIST_INSERT_HEAD inserts an element to the head of the list : - - struct ast_var_t *node; - - node=malloc(sizeof(struct ast_var_t)); - (...we fill data in struct....) - data->name=malloc(100); - strcpy(data->name,"lalalalaa"); - etc etc - - (then we insert the node in the head of the list :) - - AST_LIST_INSERT_HEAD(headp,node,listpointers); - -AST_LIST_INSERT_HEAD_AFTER inserts an element after another : - - struct ast_var_t *node1; - ... - AST_LIST_INSERT_AFTER(node,node1,listpointers); - -AST_LIST_REMOVE removes an arbitrary element from the head: - - AST_LIST_REMOVE(headp,node1,ast_var_t,listpointers); - -AST_LIST_REMOVE_HEAD removes the entry at the head of the list and -returns a pointer to the removed entry: - - AST_LIST_REMOVE_HEAD(headp,node,listpointers); - -AST_LIST_FIRST returns a pointer to the first element of the list; - - struct ast_var_t *firstnode; - firstnode=AST_LIST_FIRST(headp); - -AST_LIST_NEXT returns a pointer to the next element : - - struct ast_var_t *nextnode; - nextnode=AST_LIST_NEXT(firstnode,listpointers); - -AST_LIST_TRAVERSE traverses all elements of the list : - - struct ast_var_t *node; - - AST_LIST_TRAVERSE(headp,node,listpointers) { - printf("%s\n",node->name); - } - -AST_LIST_EMPTY evaluates to a true condition if there are no elements on -the list. - -To completely delete a list : - - struct ast_var_t *vardata; - - while (!AST_LIST_EMPTY(headp)) { /* List Deletion. */ - vardata = AST_LIST_REMOVE_HEAD(head, ast_var_t, listpointers); - free(vardata->name); - free(vardata->value); - } - -AST_LIST_LOCK returns true if it can lock the list, AST_LIST_UNLOCK unlocks -the list : - -if (AST_LIST_LOCK(headp)) { - ...do all list operations here... - AST_LIST_UNLOCK(headp); -} else { - ast_log(LOG_WARNING,"List locked bla bla bla\n"); -} diff --git a/1.2-netsec/doc/localchannel.txt b/1.2-netsec/doc/localchannel.txt deleted file mode 100644 index f96ea15ec..000000000 --- a/1.2-netsec/doc/localchannel.txt +++ /dev/null @@ -1,49 +0,0 @@ -The Local channel ------------------ - -chan_local is a pseudo-channel. Use of this channel simply loops calls back into the dialplan in a different context. Useful for recursive routing. - -* Syntax: - - Local/extension@context[/n] - -Adding "/n" at the end of the string will make the Local channel not do a native transfer (the "n" stands for "n"o release) upon the remote end answering the line. This is an esoteric, but important feature if you expect the Local channel to handle calls exactly like a normal channel. If you do not have the "no release" feature set, then as soon as the destination (inside of the Local channel) answers the line, the variables and dial plan will revert back to that of the original call, and the Local channel will become a zombie and be removed from the active channels list. This is desirable in some circumstances, but can result in unexpected dialplan behavior if you are doing fancy things with variables in your call handling. - -* Purpose: - -The Local channel construct can be used to establish dialing into any part of the dialplan. - -Imagine you have a TE410P in your box. You want to do something for which you must use a Dial statement (for instance when dropping files in /var/spool/outgoing) but you do want to be able to use your dialplans least-cost-routes or other intelligent stuff. What you could do before we had chan_local was create a cross-link between two ports of the TE410P and then Dial out one port and in the other. This way you could control where the call was going. - -Of course, this was a nasty hack, and to make it more sensible, chan_local was built. - -The "Local" channel driver allows you to convert an arbitrary extension into a channel. It is used in a variety of places, including agents, etc. - -This also allows us to hop to contexts like a GoSub routine; See examples below. - -Examples: ---------- - -[inbound] ; here falls all incoming calls -exten => s,1,Answer -exten => s,2,Dial(local/200@internal,30,r) -exten => s,3,Playback(sorrynoanswer) -exten => s,4,Hangup - -[internal] ; here where our phones falls for default -exten => 200,1,Dial(sip/blah) -exten => 200,102,VoiceMail(${EXTEN}@default) - -exten => 201,1,Dial(zap/1) -exten => 201,102,VoiceMail(${EXTEN}@default) - -exten => _0.,1,Dial(Zap/g1/${EXTEN:1}) ; outgoing calls with 0+number - - -Caveats: -If you use chan_local from a call-file and you want to pass channel variables into your context, make sure you append the '/n', because otherwise chan_local will 'optimize' itself out of the call-path, and the variables will get lost. i.e. - - Local/00531234567@pbx becomes Local/00531234567@pbx/n - ----------- -2004-01-17 diff --git a/1.2-netsec/doc/manager.txt b/1.2-netsec/doc/manager.txt deleted file mode 100644 index bd5df8077..000000000 --- a/1.2-netsec/doc/manager.txt +++ /dev/null @@ -1,297 +0,0 @@ -The Asterisk Manager TCP/IP API - AMI -===================================== - -The manager is a client/server model over TCP. With the manager interface, -you'll be able to control the PBX, originate calls, check mailbox status, -monitor channels and queues as well as execute Asterisk commands. - -AMI is the standard management interface into your Asterisk server. -You configure AMI in manager.conf. By default, AMI is available on -TCP port 5038 if you enable it in manager.conf. - -AMI receive commands, called "actions". These generate a "response" -from Asterisk. Asterisk will also send "Events" containing various -information messages about changes within Asterisk. Some actions -generate an initial response and data in the form list of events. -This format is created to make sure that extensive reports do not -block the manager interface fully. - -Management users are configured in the configuration file manager.conf and are -given permissions for read and write, where write represents their ability -to perform this class of "action", and read represents their ability to -receive this class of "event". - -The Asterisk manager interface in version 1.0.x of Asterisk is -not very well standardized. Work is under way to change this -to Asterisk 1.2. If you develop AMI applications, treat the headers -in Actions, Events and Responses as local to that particular -message. There is no cross-message standardization of headers. - -If you develop applications, please try to reuse existing manager -headers and their interpretation. If you are unsure, discuss on -the asterisk-dev mailing list. - - -Command Syntax --------------- -Management communication consists of tags of the form "header: value", -terminated with an empty newline (\r\n) in the style of SMTP, HTTP, and -other headers. - - -The first tag MUST be one of the following: - - * Action: An action requested by the CLIENT to the Asterisk SERVER. Only one "Action" may be outstanding at any time. - * Response: A response to an action from the Asterisk SERVER to the CLIENT. - * Event: An event reported by the Asterisk SERVER to the CLIENT - - -Manager commands ----------------- -Output from the CLI command 'show manager' command: - - * Ping: Ping - * Logoff: Logoff Manager - * Hangup: Hangup Channel - * Status: Status - * Redirect: Redirect - * Originate: Originate Call - * MailboxStatus: Check Mailbox - * Command: Execute Command - * ExtensionState: Check Extension Status - * AbsoluteTimeout: Set Absolute Timeout - * MailboxCount: Check Mailbox Message Count - * Monitor: Monitor a channel - * StopMonitor: Stop monitoring a channel - * ChangeMonitor: Change monitoring filename of a channel - * IAXpeers: List IAX Peers (Defaults to IAX2) - * SIPpeers: List SIP peers - * SIPshowpeer: Show data about one SIP peer - * Queues: Queues - * QueueStatus: Queue Status - -This list depends on the version of Asterisk you are using, as -well as which modules that are loaded. - -Command Summary --------------- - -Command: Command -Parameters: Command - -Command: ExtensionState -Parameters: Exten, Context, ActionID - -Command: Hangup -Parameters: Channel - -Command: Logoff -Parameters: None - -Command: MailboxCount -Parameters: Mailbox, ActionID - -Command: MailboxStatus -Parameters: Mailbox, ActionID - -Command: Originate -Parameters: Channel, Exten, Context, Priority, Timeout, - CallerID, Variable, Account, Application, Data, Async - -Command: Ping -Parameters: None - -Command: Redirect -Parameters: Channel, ExtraChannel, Exten, Context, Priority - -Command: Timeout -Parameters: Channel, Timeout - -You can always get more information about a manager command -with the "show manager command <command>" CLI command in Asterisk. - -Examples --------- -Login - Log a user into the manager interface. - - Action: Login - Username: testuser - Secret: testsecret - -Originate - Originate a call from a channel to an extension. - - Action: Originate - Channel: sip/12345 - Exten: 1234 - Context: default - -Originate - Originate a call from a channel to an extension without waiting -for call to complete. - - Action: Originate - Channel: sip/12345 - Exten: 1234 - Context: default - Async: yes - - -Redirect with ExtraChannel: - Attempted goal: - Have a 'robot' program Redirect both ends of an already-connected call - to a meetme room using the ExtraChannel feature through the management interface. - - Action: Redirect - Channel: Zap/1-1 - ExtraChannel: SIP/3064-7e00 (varies) - Exten: 680 - Priority: 1 - -Where 680 is an extension that sends you to a MeetMe room. - -There are a number of GUI tools that use the manager interface, please search -the mailing list archives and the documentation page on the -http://www.asterisk.org web site for more information. - - -Some standard AMI headers: --------------------------- - - Account: -- Account Code (Status) - AccountCode: -- Account Code (cdr_manager) - ACL: <Y | N> -- Does ACL exist for object ? - Action: <action> -- request or notification of a particular action - Address-IP: -- IPaddress - Address-Port: -- IP port number - Agent: <string> -- Agent name - AMAflags: -- AMA flag (cdr_manager, sippeers) - AnswerTime: -- Time of answer (cdr_manager) - Append: <bool> -- CDR userfield Append flag - Application: -- Application to use - Async: -- Whether or not to use fast setup - AuthType: -- Authentication type (for login or challenge) - "md5" - BillableSeconds: -- Billable seconds for call (cdr_manager) - CallerID: -- Caller id (name and number in Originate & cdr_manager) - CallerID: -- CallerID number - Number or "<unknown>" or "unknown" - (should change to "<unknown>" in app_queue) - CallerID1: -- Channel 1 CallerID (Link event) - CallerID2: -- Channel 2 CallerID (Link event) - CallerIDName: -- CallerID name - Name or "<unknown>" or "unknown" - (should change to "<unknown>" in app_queue) - Callgroup: -- Call group for peer/user - CallsTaken: <num> -- Queue status variable - Cause: <value> -- Event change cause - "Expired" - Cause: <value> -- Hangupcause (channel.c) - CID-CallingPres: -- Caller ID calling presentation - Channel: <channel> -- Channel specifier - Channel: <dialstring> -- Dialstring in Originate - Channel: <tech/[peer/username]> -- Channel in Registry events (SIP, IAX2) - Channel: <tech> -- Technology (SIP/IAX2 etc) in Registry events - ChannelType: -- Tech: SIP, IAX2, ZAP, MGCP etc - Channel1: -- Link channel 1 - Channel2: -- Link channel 2 - ChanObjectType: -- "peer", "user" - Codecs: -- Codec list - CodecOrder: -- Codec order, separated with comma "," - Command: -- Cli command to run - Context: -- Context - Count: <num> -- Number of callers in queue - Data: -- Application data - Default-addr-IP: -- IP address to use before registration - Default-Username: -- Username part of URI to use before registration - Destination: -- Destination for call (Dialstring ) (dial, cdr_manager) - DestinationContext: -- Destination context (cdr_manager) - DestinationChannel: -- Destination channel (cdr_manager) - DestUniqueID: -- UniqueID of destination (dial event) - Disposition: -- Call disposition (CDR manager) - Domain: <domain> -- DNS domain - Duration: <secs> -- Duration of call (cdr_manager) - Dynamic: <Y | N> -- Device registration supported? - Endtime: -- End time stamp of call (cdr_manager) - EventList: <flag> -- Flag being "Start", "End", "Cancelled" or "ListObject" - Events: <eventmask> -- Eventmask filter ("on", "off", "system", "call", "log") - Exten: -- Extension (Redirect command) - Extension: -- Extension (Status) - Family: <string> -- ASTdb key family - File: <filename> -- Filename (monitor) - Format: <format> -- Format of sound file (monitor) - From: <time> -- Parking time (ParkedCall event) - Hint: -- Extension hint - Incominglimit: -- SIP Peer incoming limit - Key: - Key: -- ASTdb Database key - LastApplication: -- Last application executed (cdr_manager) - LastCall: <num> -- Last call in queue - LastData: -- Data for last application (cdr_manager) - Link: -- (Status) - ListItems: <number> -- Number of items in Eventlist (Optionally sent in "end" packet) - Location: -- Interface (whatever that is -maybe tech/name in app_queue ) - Loginchan: -- Login channel for agent - Logintime: <number> -- Login time for agent - Mailbox: -- VM Mailbox (id@vmcontext) (mailboxstatus, mailboxcount) - MD5SecretExist: <Y | N> -- Whether secret exists in MD5 format - Membership: <string> -- "Dynamic" or "static" member in queue - Message: <text> -- Text message in ACKs, errors (explanation) - Mix: <bool> -- Boolean parameter (monitor) - NewMessages: <count> -- Count of new Mailbox messages (mailboxcount) - Newname: - ObjectName: -- Name of object in list - OldName: -- Something in Rename (channel.c) - OldMessages: <count> -- Count of old mailbox messages (mailboxcount) - Outgoinglimit: -- SIP Peer outgoing limit - Paused: <num> -- Queue member paused status - Peer: <tech/name> -- "channel" specifier :-) - PeerStatus: <tech/name> -- Peer status code - "Unregistred", "Registred", "Lagged", "Reachable" - Penalty: <num> -- Queue penalty - Priority: -- Extension priority - Privilege: <privilege> -- AMI authorization class (system, call, log, verbose, command, agent, user) - Pickupgroup: -- Pickup group for peer - Position: <num> -- Position in Queue - Queue: -- Queue name - Reason: -- "Autologoff" - Response: <response> -- response code, like "200 OK" - "Success", "Error", "Follows" - Restart: -- "True", "False" - RegExpire: -- SIP registry expire - RegExpiry: -- SIP registry expiry - Reason: -- Originate reason code - Seconds: -- Seconds (Status) - Secret: <password> -- Authentication secret (for login) - SecretExist: <Y | N> -- Whether secret exists - Shutdown: -- "Uncleanly", "Cleanly" - SIP-AuthInsecure: - SIP-FromDomain: -- Peer FromDomain - SIP-FromUser: -- Peer FromUser - SIP-NatSupport: - SIPLastMsg: - Source: -- Source of call (dial event, cdr_manager) - SrcUniqueID: -- UniqueID of source (dial event) - StartTime: -- Start time of call (cdr_manager) - State: -- Channel state - Status: -- Registration status (Registry events SIP) - Status: -- Extension status (Extensionstate) - Status: -- Peer status (if monitored) ** Will change name ** - "unknown", "lagged", "ok" - Status: <num> -- Queue Status - Status: -- DND status (DNDState) - Time: <sec> -- Roundtrip time (latency) - Timeout: -- Parking timeout time - Timeout: -- Timeout for call setup (Originate) - Timeout: <seconds> -- Timout for call - Uniqueid: -- Channel Unique ID - Uniqueid1: -- Channel 1 Unique ID (Link event) - Uniqueid2: -- Channel 2 Unique ID (Link event) - User: -- Username (SIP registry) - UserField: -- CDR userfield (cdr_manager) - Val: -- Value to set/read in ASTdb - Variable: -- Variable AND value to set (multiple separated with | in Originate) - Variable: <name> -- For channel variables - Value: <value> -- Value to set - VoiceMailbox: -- VM Mailbox in SIPpeers - Waiting: -- Count of mailbox messages (mailboxstatus) - - ** Please try to re-use existing headers to simplify manager message parsing in clients. - diff --git a/1.2-netsec/doc/model.txt b/1.2-netsec/doc/model.txt deleted file mode 100644 index 10d2d0e05..000000000 --- a/1.2-netsec/doc/model.txt +++ /dev/null @@ -1,15 +0,0 @@ -Description of call model: - -Incoming Call: - - Channel backend waits for a RING or equivalent on some sort of -interface. Typically this is done in its own thread. When a RING is -detected, the backend should create a channel structure and then call -ast_pbx_start() on that channel, which will create a thread to monitor -that interface. At this point, the PBX and/or applications it launches -will manage the interface, and it need not be monitored by the -aforementioned thread. When the applications are finished, the requisite -hangup function will be called, at which the channel can be considered to -be no longer valid, and the thread that controls it will imminently be -terminated. - diff --git a/1.2-netsec/doc/modules.txt b/1.2-netsec/doc/modules.txt deleted file mode 100644 index 4f6d4c67b..000000000 --- a/1.2-netsec/doc/modules.txt +++ /dev/null @@ -1,26 +0,0 @@ -All modules must have at least the following functions: - -int load_module(): - - Do what you need to do when you get started. This function -returns 0 on success and non-zero on failure (it is not considered loaded -if it fails. - -int unload_module(): - - The module will soon be unloaded. If any channels are using your -features, you should give them a softhangup in an effort to keep the -program from crashing. Generally, unload_module is only called when the -usecount is 0 or less, but the user can force unloading at their -discretion, and thus a module should do its best to comply (although in -some cases there may be no way to avoid a crash). This function should -return 0 on success and non-zero on failure (i.e. it cannot yet be -unloaded). - -char *description(): - - Return a description of the module's functionality. - -int usecnt(): - - Return the number of channels, etc that are using you. diff --git a/1.2-netsec/doc/queuelog.txt b/1.2-netsec/doc/queuelog.txt deleted file mode 100644 index a11812eef..000000000 --- a/1.2-netsec/doc/queuelog.txt +++ /dev/null @@ -1,77 +0,0 @@ -Queue Log Information -===================== - -In order to properly manage ACD queues, it is important to be able to -keep track of details of call setups and teardowns in much greater detail -than traditional call detail records provide. In order to support this, -extensive and detailed tracing of every queued call is stored in the -queue log, located (by default) in /var/log/asterisk/queue_log. - -These are the events (and associated information) in the queue log: - -ABANDON(position|origposition|waittime) -The caller abandoned their position in the queue. The position is the -caller's position in the queue when they hungup, the origposition is -the original position the caller was when they first entered the -queue, and the waittime is how long the call had been waiting in the -queue at the time of disconnect. - -AGENTDUMP -The agent dumped the caller while listening to the queue announcement. - -AGENTLOGIN(channel) -The agent logged in. The channel is recorded. - -AGENTCALLBACKLOGIN(exten@context) -The callback agent logged in. The login extension and context is recorded. - -AGENTLOGOFF(channel|logintime) -The agent logged off. The channel is recorded, along with the total time -the agent was logged in. - -AGENTCALLBACKLOGOFF(exten@context|logintime|reason) -The callback agent logged off. The last login extension and context is -recorded, along with the total time the agent was logged in, and the -reason for the logoff if it was not a normal logoff (e.g., Autologoff) - -COMPLETEAGENT(holdtime|calltime|origposition) -The caller was connected to an agent, and the call was terminated normally -by the *agent*. The caller's hold time and the length of the call are both -recorded. The caller's original position in the queue is recorded in -origposition. - -COMPLETECALLER(holdtime|calltime|origposition) -The caller was connected to an agent, and the call was terminated normally -by the *caller*. The caller's hold time and the length of the call are both -recorded. The caller's original position in the queue is recorded in -origposition. - -CONFIGRELOAD -The configuration has been reloaded (e.g. with asterisk -rx reload) - -CONNECT(holdtime) -The caller was connected to an agent. Hold time represents the amount -of time the caller was on hold. - -ENTERQUEUE(url|callerid) -A call has entered the queue. URL (if specified) and Caller*ID are placed -in the log. - -EXITWITHKEY(key|position) -The caller elected to use a menu key to exit the queue. The key and -the caller's position in the queue are recorded. - -EXITWITHTIMEOUT(position) -The caller was on hold too long and the timeout expired. - -QUEUESTART -The queueing system has been started for the first time this session. - -SYSCOMPAT -A call was answered by an agent, but the call was dropped because the -channels were not compatible. - -TRANSFER(extension,context) -Caller was transferred to a different extension. Context and extension -are recorded. - |