1. Update osp guide.

git-svn-id: http://svn.digium.com/svn/asterisk/trunk@49507 f38db490-d61c-443f-a65b-d21fe96a405b

1 files changed, 662 insertions, 321 deletions

diff --git a/doc/osp.txt b/doc/osp.txt
index 08e0edc91..f53789848 100644
--- a/doc/osp.txt
+++ b/doc/osp.txt
@@ -1,176 +1,170 @@
-Asterisk OSP Module User Guide
+OSP User Guide for Asterisk V1.6
 
-June 16, 2006
+4 January 2007
 
 Table of Contents
-1        Introduction
-2        OSP Toolkit
-2.1      Build OSP Toolkit
-2.1.1    Unpacking the Toolkit
-2.1.2    Preparing to build the OSP Toolkit
-2.1.3    Building the OSP Toolkit
-2.1.4    Installing the OSP Toolkit
-2.1.5    Building the Enrollment Utility
-2.2      Obtain Crypto Files
-3        Asterisk
-3.1      OSP Support Implementation
-3.1.1    OSPAuth
-3.1.2    OSPLookup
-3.1.3    OSPNext
-3.1.4    OSPFinish
-3.2      Build with OSP Support
-3.3      Configure with OSP Support
-3.3.1    osp.conf
-3.3.2    zapata/sip/iax.conf
-3.3.3    extensions.conf
+1	Introduction
+2	OSP Toolkit
+2.1	Build OSP Toolkit
+2.1.1	Unpacking the Toolkit
+2.1.2	Preparing to build the OSP Toolkit
+2.1.3	Building the OSP Toolkit
+2.1.4	Installing the OSP Toolkit
+2.1.5	Building the Enrollment Utility
+2.2	Obtain Crypto Files
+3	Asterisk
+3.1	Configure for OSP Support
+3.1.1	Build Asterisk with OSP Toolkit
+3.1.2	osp.conf
+3.1.3	extensions.conf
+3.1.4	zapata/sip/iax/h323/ooh323.conf
+3.2	OSP Dial Plan Functions
+3.2.1	OSPAuth
+3.2.2	OSPLookup
+3.2.3	OSPNext
+3.2.4	OSPFinish
+3.3	extensions.conf Examples
+3.3.1	Source Gateway
+3.3.2	Destination Gateway
+3.3.3	Proxy
 
 Asterisk is a trademark of Digium, Inc.
-TransNexus and OSP Secured are trademarks of TransNexus, Inc.
+TransNexus and OSP Secures are trademarks of TransNexus, Inc.
 
 1 Introduction
   This document provides instructions on how to build and configure Asterisk 
-  V1.4 with the OSP Toolkit to enable secure, multi-lateral peering. The OSP 
-  Toolkit is an open source implementation of the OSP peering protocol and is 
-  freely available from www.sipfoundry.org.  The OSP standard defined by the 
-  European Telecommunications Standards Institute (ETSI TS 101 321) 
-  www.esti.org.  If you have questions or need help, building Asterisk with the
+  V1.6 with the OSP Toolkit to enable secure, multi-lateral peering.  This 
+  document is also available in the Asterisk source package as doc/osp.txt.  
+  The OSP Toolkit is an open source implementation of the OSP peering protocol 
+  and is freely available from www.sipfoundry.org.  The OSP standard defined by 
+  the European Telecommunications Standards Institute (ETSI TS 101 321) 
+  www.esti.org.  If you have questions or need help, building Asterisk with the 
   OSP Toolkit, please post your question on the OSP mailing list at 
-  https://list.sipfoundry.org/mailman/listinfo/osp. 
+  https://list.sipfoundry.org/mailman/listinfo/osp.
 
 2 OSP Toolkit
   Please reference the OSP Toolkit document "How to Build and Test the OSP 
-  Toolkit" available from www.sipfoundry.org/OSP/OSPclient .  
+  Toolkit" available from https://www.sipfoundry.org/OSPclient.  
 
 2.1 Build OSP Toolkit
-  The software listed below is required ti build and use the OSP Toolkit:
-  * OpenSSL (required for building) - Open Source SSL protocol and 
-  Cryptographic Algorithms (version 0.9.7g recommended) from www.openssl.org. 
-  Pre-compiled OpenSSL binary packages are not recommended because of the 
-  binary compatibility issue. 
+  The software listed below is required to build and use the OSP Toolkit:
+  * OpenSSL (required for building) - Open Source SSL protocol and Cryptographic 
+    Algorithms (version 0.9.7g recommended) from www.openssl.org. Pre-compiled 
+    OpenSSL binary packages are not recommended because of the binary 
+    compatibility issue. 
   * Perl (required for building) - A programming language used by OpenSSL for 
-  compilation. Any version of Perl should work. One version of Perl is 
-  available from www.activestate.com/ActivePerl. If pre-compiled OpenSSL 
-  packages are used, Perl package is not required.
+    compilation. Any version of Perl should work. One version of Perl is 
+    available from www.activestate.com/Products/ActivePer. If pre-compiled 
+    OpenSSL packages are used, Perl package is not required.
   * C compiler (required for building) - Any C compiler should work.  The GNU 
-  Compiler Collection from www.gnu.org is routinely used for building the OSP 
-  Toolkit for testing.
+    Compiler Collection from www.gnu.org is routinely used for building the OSP 
+    Toolkit for testing.
   * OSP Server (required for testing) - Access to any OSP server should work.  
-  Open source OSP servers are available from www.sipfoundry.org/osp, a free 
-  commercial OSP server may be downloaded from www.transnexus.com and an OSP 
-  server osptestserver.transnexus.com is freely available on the internet for 
-  testing for testing. Please contact support@transnexus.com for testing access
-  to osptestserver.transnexus.com.
+    Open source OSP servers are available from https://www.sipfoundry.org/OSP, 
+    or go to http://www.transnexus.com/OSP%20Toolkit/Peering_Server/VoIP_Peering_Server.htm 
+    to download a free commercial OSP server. 
 
 2.1.1 Unpacking the Toolkit
-  After downloading the OSP Toolkit (version 3.3.4 or later release) from 
+  After downloading the OSP Toolkit (version 3.3.6 or later release) from 
   www.sipfoundry.org, perform the following steps in order:
-  1) Copy the OSP Toolkit distribution into the directory where it will reside,
-  say /usr/src. 
+  1) Copy the OSP Toolkit distribution into the directory where it will reside. 
+     The default directory for the OSP Toolkit is /usr/src. 
   2) Un-package the distribution file by executing the following command: 
-    gunzip -c OSPToolkit-###.tar.gz | tar xvf -
-    Where ### is the version number separated by underlines. For example, if 
-    the version is 3.3.4, then the above command would be: 
-    gunzip -c OSPToolkit-3_3_4.tar.gz | tar xvf -
-  A new directory (TK-3_3_4-20051103) will be created within the same directory
-  as the tar file.
-  3) Go to the TK-3_3_4-20051103 directory by running this command:
-    cd TK-3_3_4-20051103
-  Within this directory, you will find directories and files similar to what is
-  listed below if the command "ls -F" is executed):
-    ls -F
-    enroll/
-    RelNotes.txt        lib/
-    README.txt          license.txt
-    bin/                src/
-    crypto/             test/
-    include/
+       gunzip -c OSPToolkit-###.tar.gz | tar xvf -
+     Where ### is the version number separated by underlines. For example, if 
+     the version is 3.3.6, then the above command would be: 
+       gunzip -c OSPToolkit-3_3_6.tar.gz | tar xvf -
+     A new directory (TK-3_3_6-20060303) will be created within the same 
+     directory as the tar file.
+  3) Go to the TK-3_3_6-20060303 directory by running this command:
+       cd TK-3_3_6-20060303
+     Within this directory, you will find directories and files similar to what 
+     is listed below if the command "ls -F" is executed):
+       ls -F
+       enroll/
+       RelNotes.txt     lib/
+       README.txt       license.txt
+       bin/             src/
+       crypto/          test/
+       include/
 
 2.1.2 Preparing to build the OSP Toolkit
   4) Compile OpenSSL according to the instructions provided with the OpenSSL 
-  distribution (You would need to do this only if you don't have openssl 
-  already).
+     distribution (You would need to do this only if you don't have openssl 
+     already).
   5) Copy the OpenSSL header files (the *.h files) into the crypto/openssl 
-  directory within the osptoolkit directory. The OpenSSL header files are 
-  located under the openssl/include/openssl directory.
+     directory within the osptoolkit directory. The OpenSSL header files are 
+     located under the openssl/include/openssl directory.
   6) Copy the OpenSSL library files (libcrypto.a and libssl.a) into the lib 
-  directory within the osptoolkit directory. The OpenSSL library files are 
-  located under the openssl directory.
-  Note: Since the Asterisk requires the OpenSSL package. If the OpenSSL package
-  has been installed, 4~6 are not necessary.
+     directory within the osptoolkit directory. The OpenSSL library files are 
+     located under the openssl directory.
+     Note: Since the Asterisk requires the OpenSSL package. If the OpenSSL 
+           package has been installed, steps 4 through 6 are not necessary.
+  7) Optionally, change the install directory of the OSP Toolkit. Open the 
+     Makefile in the /usr/src/TK-3_3_6-20060303/src directory, look for the 
+     install path variable - INSTALL_PATH, and edit it to be anywhere you want 
+     (defaults /usr/local). 
+     Note: Please change the install path variable only if you are familiar 
+           with both the OSP Toolkit and the Asterisk. 
 
 2.1.3 Building the OSP Toolkit
-  7) Optionally, change the install directory of the OSP Toolkit. Open the 
-  Makefile in the /usr/src/TK-3_3_4-20051103/src directory, look for the 
-  install path variable - INSTALL_PATH, and edit it to be anywhere you want 
-  (defaults /usr/local). 
-  Note: Please change the install path variable only if you are familiar with 
-  both the OSP Toolkit and the Asterisk. Otherwise, it may case that the 
-  Asterisk does not support the OSP protocol.
-  8) From within the OSP Toolkit directory (/usr/src/TK-3_3_4-20051103), start 
-  the compilation script by executing the following commands:
-    cd src
-    make clean; make build
+  8) From within the OSP Toolkit directory (/usr/src/TK-3_3_6-20060303), start 
+     the compilation script by executing the following commands:
+       cd src
+       make clean; make build
 
 2.1.4 Installing the OSP Toolkit
   The header files and the library of the OSP Toolkit should be installed. 
   Otherwise, you must specify the OSP Toolkit path for the Asterisk.
-  9) Use the same script to install the Toolkit.
-    make install
-  The make script is also used to install the OSP Toolkit header files and the 
-  library into the INSTALL_PATH specified in the Makefile. 
-  Note: Please make sure you have the rights to access the INSTALL_PATH 
-  directory. For example, in order to access /usr/local directory, normally, 
-  you should be root.
-  By default, the OSP Toolkit is compiled in the production mode. The following
-  table identifies which default features are activated with each compile 
-  option:
-    Default Feature                Production    Development
-    Debug Information Displayed    No            Yes
-  The "Development" option is recommended for a first time build. The CFLAGS 
-  definition in the Makefile must be modified to build in development mode.  
+  9) Use the make script to install the Toolkit.
+       make install
+     The make script is also used to install the OSP Toolkit header files and 
+     the library into the INSTALL_PATH specified in the Makefile. 
+     Note: Please make sure you have the rights to access the INSTALL_PATH 
+           directory. For example, in order to access /usr/local directory, 
+           root privileges are required.
 
 2.1.5 Building the Enrollment Utility
   Device enrollment is the process of establishing a trusted cryptographic 
-  relationship between the VoIP device and the OSP Server. The Enroll program 
-  is a utility application for establishing a trusted relationship between and 
-  OSP client and an OSP server. Please see the document "Device Enrollment" at 
-  www.sipfoundry.org/OSP/OSPclient for more information about the enroll 
+  relationship between the VoIP device and the OSP Server. The Enroll program is
+  a utility application for establishing a trusted relationship between an OSP 
+  client and an OSP server. Please see the document "Device Enrollment" at 
+  https://www.sipfoundry.org/OSPclient for more information about the enroll 
   application.
-  10) From within the OSP Toolkit directory (/usr/src/TK-3_3_4-20051103), 
-  execute the following commands at the command prompt:
-    cd enroll
-    make clean; make linux
-  Compilation is successful if there are no errors anywhere in the compiler 
-  output. The enroll program is now located in the 
-  /usr/src/TK-3_3_4-20051103/bin directory. By this point, a fully functioning 
-  OSP Toolkit should have been successfully built.
+  10) From within the OSP Toolkit directory (example: 
+      /usr/src/TK-3_3_6-20060303), execute the following commands at the command 
+      prompt:
+        cd enroll
+        make clean; make linux
+      Compilation is successful if there are no errors in the compiler output. 
+      The enroll program is now located in the OSP Toolkit/bin directory 
+      (example: /usr/src/ TK-3_3_6-20060303/bin). 
 
 2.2 Obtain Crypto Files
-  The OSP module in Asterisk requires three crypto files containing local 
+  The OSP module in Asterisk requires three crypto files containing a local 
   certificate (localcert.pem), private key (pkey.pem), and CA certificate 
   (cacert_0.pem).  Asterisk will try to load the files from the Asterisk 
-  public/private key directory - /var/lib/asterisk/key.  If the files are not 
+  public/private key directory - /var/lib/asterisk/keys.  If the files are not 
   present, the OSP module will not start and the Asterisk will not support the 
   OSP protocol.  Use the enroll.sh script from the toolkit distribution to 
-  enroll the Asterisk OSP module with an OSP server to obtain the crypto files.
-  Documentation explaining how to use the enroll.sh script (Device Enrollment) 
-  to enroll with an OSP server is available at 
-  www.sipfoundry.org/OSP/ospclient.  Copy the files file generated by the 
-  enrollment process to the Asterisk configuration directory.  
+  enroll Asterisk with an OSP server and obtain the crypto files.  Documentation 
+  explaining how to use the enroll.sh script (Device Enrollment) to enroll with 
+  an OSP server is available at https://www.sipfoundry.org/OSPclient.  Copy the 
+  files generated by the enrollment process to the Asterisk 
+  /var/lib/asterisk/keys directory.  
   Note: The osptestserver.transnexus.com is configured only for sending and 
-  receiving non-SSL messages, and issuing signed tokens. If you need help, post
-  a message on the OSP mailing list of www.sipfoundry.org or send an e-mail to 
-  support@transnexus.com.
-  The enroll.sh script takes the domain name or IP addresses of the OSP servers
-  that the OSP Toolkit needs to enroll with as arguments, and then generates 
-  pem files - cacert_#.pem, certreq.pem, localcert.pem, and pkey.pem. The '#'
-  in the cacert file name is used to differentiate the ca certificate file 
-  names for the various SP's (OSP servers). If only one address is provided at 
-  the command line, cacert_0.pem will be generated. If 2 addresses are provided
-  at the command line, 2 files will be generated - cacert_0.pem and 
-  cacert_1.pem, one for each SP. The example below shows the usage when the 
-  client is registering with osptestserver.transnexus.com. If all goes well, 
-  the following text will be displayed. The gray boxes indicate required input.
+        receiving non-SSL messages, and issuing signed tokens. If you need help, 
+        post a message on the OSP mailing list of www.sipfoundry.org or send an 
+        e-mail to support@transnexus.com.
+  The enroll.sh script takes the domain name or IP addresses of the OSP servers 
+  that the OSP Toolkit needs to enroll with as arguments, and then generates pem 
+  files - cacert_#.pem, certreq.pem, localcert.pem, and pkey.pem. The "#" in the 
+  cacert file name is used to differentiate the ca certificate file names for 
+  the various SP's (OSP servers). If only one address is provided at the command 
+  line, cacert_0.pem will be generated. If 2 addresses are provided at the 
+  command line, 2 files will be generated - cacert_0.pem and cacert_1.pem, one 
+  for each SP (OSP server). The example below shows the usage when the client 
+  is registering with osptestserver.transnexus.com. 
     ./enroll.sh osptestserver.transnexus.com
     Generating a 512 bit RSA private key
     ........................++++++++++++
@@ -191,17 +185,17 @@ TransNexus and OSP Secured are trademarks of TransNexus, Inc.
     Organizational Unit Name (eg, section) []:_______
     Common Name (eg, YOUR name) []:_______
     Email Address []:_______
-
+    
     Please enter the following 'extra' attributes
     to be sent with your certificate request
     A challenge password []:_______
     An optional company name []:_______
-
+   
     Error Code returned from openssl command : 0
-
+  
     CA certificate received
     [SP: osptestserver.transnexus.com]Error Code returned from getcacert command : 0
-
+      
     output buffer after operation: operation=request
     output buffer after nonce: operation=request&nonce=1655976791184458
     X509 CertInfo context is null pointer
@@ -213,69 +207,359 @@ TransNexus and OSP Secured are trademarks of TransNexus, Inc.
     verify return:1
     The certificate request was successful.
     Error Code returned from localcert command : 0
-  The files generated should be copied to the /var/lib/asterisk/key 
-  directory. 
+  The files generated should be copied to the /var/lib/asterisk/keys directory. 
   Note: The script enroll.sh requires AT&T korn shell (ksh) or any of its 
-  compatible variants. The /usr/src/TK-3_3_4-20051103/bin directory should be 
-  in the PATH variable. Otherwise, enroll.sh cannot find the enroll file.
+        compatible variants. The /usr/src/TK-3_3_6-20060303/bin directory should 
+        be in the PATH variable. Otherwise, enroll.sh cannot find the enroll 
+        file.
 
 3 Asterisk
+  In Asterisk, all OSP support is implemented as dial plan functions. In 
+  Asterisk V1.6, all combinations of routing between OSP and non-OSP enabled 
+  networks using any combination of SIP, H.323 and IAX protocols are fully 
+  supported.  Section 3.1 describes the three easy steps to add OSP support to 
+  Asterisk:
+    1. Build Asterisk with OSP Toolkit
+    2. Configure osp.conf file
+    3. Cut and paste to extensions.conf
+  Sections 3.2 and 3.3 provide a detailed explanation of OSP dial plan functions 
+  and configuration examples.  The detailed information provided in Sections 3.2 
+  and 3.3 is not required for operating Asterisk with OSP, but may be helpful to 
+  developers who want to customize their Asterisk OSP implementation.
+
+3.1 Configure for OSP Support
+
+3.1.1 Build Asterisk with OSP Toolkit
+  The first step is to build Asterisk with the OSP Toolkit.  If the OSP Toolkit 
+  is installed in the default install directory, /usr/local, no additional 
+  configuration is required.  Compile Asterisk according to the instructions 
+  provided with the Asterisk distribution. 
+  If the OSP Toolkit is installed in another directory, such as /myosp, Asterisk 
+  must be configured with the location of the OSP Toolkit.  See the example 
+  below.
+    --with-osptk=/myosp
+  Note: Please change the install path only if you familiar with both the OSP 
+        Toolkit and the Asterisk. Otherwise, the change may result in Asterisk 
+        not supporting the OSP protocol.
 
-3.1 OSP Support Implementation
-  In Asterisk, all OSP support is implemented as dial plan functions.
+3.1.2 osp.conf
+  The /etc/asterisk/osp.conf file, shown below, contains configuration 
+  parameters for using OSP.  Two parameters, servicepoint and source must be 
+  configured.  The default values for all other parameters will work well for 
+  standard OSP implementations.
+    ;
+    ; Open Settlement Protocol Sample Configuration File
+    ;
+    ; This file contains configuration of OSP server providers that
+    ; are used by the Asterisk OSP module.  The section "general" is 
+    ; reserved for global options.  All other sections describe specific 
+    ; OSP Providers.  The provider "default" is used when no provider is 
+    ; otherwise specified.
+    :
+    : The "servicepoint" and "source" parameters must be configured.  For
+    ; most implementations the other parameters in this file can be left 
+    ; unchanged.
+    ;
+    [general]
+    ;
+    ; Enable cryptographic acceleration hardware.  
+    ;
+    accelerate=no
+    ;
+    ; Defines the status of tokens that Asterisk will validate. 
+    ; 0 - signed tokens only 
+    ; 1 - unsigned tokens only 
+    ; 2 - both signed and unsigned
+    ; The default value is 0, i.e. the Asterisk will only validate signed
+    ; tokens.
+    ;
+    tokenformat=0
+    ;
+    [default]
+    ;
+    ; List all service points (OSP servers) for this provider.  Use 
+    ; either domain name or IP address.  Most OSP servers use port 1080.
+    ;
+    ;servicepoint=http://osptestserver.transnexus.com:1080/osp
+    servicepoint=http://OSP server IP:1080/osp
+    ;
+    ; Define the "source" device for requesting OSP authorization.
+    : This value is usually the domain name or IP address of the
+    : the Asterisk server.
+    ;
+    ;source=domain name or [IP address in brackets]
+    source=[host IP]
+    ;
+    ; Define path and file name of crypto files.
+    ; The default path for crypto file is /var/lib/asterisk/keys.  If no
+    ; path is defined, crypto files should be in  
+    ; /var/lib/asterisk/keys directory.
+    ;
+    ; Specify the private key file name.  
+    ; If this parameter is unspecified or not present, the default name 
+    ; will be the osp.conf section name followed by "-privatekey.pem" 
+    ; (for example: default-privatekey.pem)
+    ;
+    privatekey=pkey.pem
+    ;
+    ; Specify the local certificate file.  
+    ; If this parameter is unspecified or not present, the default name 
+    ; will be the osp.conf section name followed by "- localcert.pem " 
+    ; (for example: default-localcert.pem)
+    ;
+    localcert=localcert.pem
+    ;
+    ; Specify one or more Certificate Authority key file names.  If none 
+    ; are listed, a single Certificate Authority key file name is added 
+    ; with the default name of the osp.conf section name followed by 
+    ; "-cacert_0.pem " (for example: default-cacert_0.pem)
+    ;
+    cacert=cacert_0.pem
+    ;
+    ; Configure parameters for OSP communication between Asterisk OSP 
+    ; client and OSP servers. 
+    ;
+    ; maxconnections: Max number of simultaneous connections to the 
+    ;                 provider OSP server (default=20)
+    ; retrydelay:     Extra delay between retries (default=0)
+    ; retrylimit:     Max number of retries before giving up (default=2)
+    ; timeout:        Timeout for response in milliseconds (default=500)
+    ;
+    maxconnections=20
+    retrydelay=0
+    retrylimit=2
+    timeout=500
+    ;
+    ; Set the authentication policy.  
+    ; 0 - NO        - Accept all calls.
+    ; 1 - YES       - Accept calls with valid token or no token.
+    ;                  Block calls with invalid token.  
+    ; 2 - EXCLUSIVE - Accept calls with valid token.
+    ;                  Block calls with invalid token or no token.
+    ; Default is 1,
+    ;
+    authpolicy=1
+    ;
+    ; Set the default destination protocol. The OSP module supports 
+    ; SIP, H323, and IAX protocols.  The default protocol is set to SIP.
+    ;
+    defaultprotocol=SIP
 
-3.1.1 OSPAuth 
+3.1.3 extensions.conf
+  OSP functions are implemented as dial plan functions in the extensions.conf 
+  file.  To add OSP support to your Asterisk server, simply copy and paste the 
+  text box below to your extensions.conf file.  These functions will enable your 
+  Asterisk server to support all OSP call scenarios.  Configuration of your 
+  Asterisk server for OSP is now complete.
+    [globals]
+    DIALOUT=Zap/1
+    
+    [SrcGW]	; OSP Source Gateway
+    exten => _XXXX.,1,NoOp(OSPSrcGW)
+    ; Set calling number if necessary
+    exten => _XXXX.,n,Set(CALLERID(numner)=1234567890)
+    ; OSP lookup using default provider, if fail/error jump to lookup+101
+    exten => _XXXX.,n(lookup),OSPLookup(${EXTEN}||j)
+    ; Deal with outbound call according to protocol
+    exten => _XXXX.,n,Macro(outbound)
+    ; Dial to destination, 60 timeout, with call duration limit
+    exten => _XXXX.,n,Dial(${OSPDIALSTR},60,oL($[${OSPOUTTIMELIMIT}*1000]))
+    ; Wait 1 second
+    exten => _XXXX.,n,Wait,1
+    ; Hangup
+    exten => _XXXX.,n,Hangup
+    ; Deal with OSPLookup fail/error
+    exten => _XXXX.,lookup+101,Hangup
+    exten => h,1,NoOp()
+    ; OSP report usage
+    exten => h,n,OSPFinish(${HANGUPCAUSE})
+    
+    [DstGW]	; OSP Destination Gateway
+    exten => _XXXX.,1,NoOp(OSPDstGW)
+    ; Deal with inbound call according to protocol
+    exten => _XXXX.,n,Macro(inbound)
+    ; Validate token using default provider, if fail/error jump to auth+101
+    exten => _XXXX.,n(auth),OSPAuth(|j)
+    ; Ringing
+    exten => _XXXX.,n,Ringing
+    ; Wait 1 second
+    exten => _XXXX.,n,Wait,1
+    ; Check inbound call duration limit
+    exten => _XXXX.,n,GoToIf($[${OSPINTIMELIMIT}=0]?100:200)
+    ; Without duration limit
+    exten => _XXXX.,100,Dial(${DIALOUT},15,o)
+    exten => _XXXX.,n,Goto(1000)
+    ; With duration limit
+    exten => _XXXX.,200,Dial(${DIALOUT},15,oL($[${OSPINTIMELIMIT}*1000]))
+    exten => _XXXX.,n,Goto(1000)
+    ; Wait 1 second
+    exten => _XXXX.,1000,Wait,1
+    ; Hangup
+    exten => _XXXX.,n,Hangup
+    ; Deal with OSPAuth fail/error
+    exten => _XXXX.,auth+101,Hangup
+    exten => h,1,NoOp()
+    ; OSP report usage
+    exten => h,n,OSPFinish(${HANGUPCAUSE})
+    
+    [GeneralProxy]	; Proxy
+    exten => _XXXX.,1,NoOp(OSP-GeneralProxy)
+    ; Deal with inbound call according to protocol
+    exten => _XXXX.,n,Macro(inbound)
+    ; Validate token using default provider, if fail/error jump to auth+101
+    exten => _XXXX.,n(auth),OSPAuth(|j)
+    ; OSP lookup using default provider, if fail/error jump to lookup+101
+    exten => _XXXX.,n(lookup),OSPLookup(${EXTEN}||j)
+    ; Deal with outbound call according to protocol
+    exten => _XXXX.,n,Macro(outbound)
+    ; Dial to destination, 14 timeout, with call duration limit
+    exten => _XXXX.,n,Dial(${OSPDIALSTR},14,oL($[${OSPOUTTIMELIMIT}*1000]))
+    ; OSP lookup next destination using default provider, if fail/error jump to next1+101
+    exten => _XXXX.,n(next1),OSPNext(${HANGUPCAUSE}||j)
+    ; Deal with outbound call according to protocol
+    exten => _XXXX.,n,Macro(outbound)
+    ; Dial to destination, 15 timeout, with call duration limit
+    exten => _XXXX.,n,Dial(${OSPDIALSTR},15,oL($[${OSPOUTTIMELIMIT}*1000]))
+    ; OSP lookup next destination using default provider, if fail/error jump to next2+101
+    exten => _XXXX.,n(next2),OSPNext(${HANGUPCAUSE}||j)
+    ; Deal with outbound call according to protocol
+    exten => _XXXX.,n,Macro(outbound)
+    ; Dial to destination, 16 timeout, with call duration limit
+    exten => _XXXX.,n,Dial(${OSPDIALSTR},16,oL($[${OSPOUTTIMELIMIT}*1000]))
+    ; Hangup
+    exten => _XXXX.,n,Hangup
+    ; Deal with OSPAuth fail/error
+    exten => _XXXX.,auth+101,Hangup
+    ; Deal with OSPLookup fail/error
+    exten => _XXXX.,lookup+101,Hangup
+    ; Deal with OSPNext fail/error
+    exten => _XXXX.,next1+101,Hangup
+    ; Deal with OSPNext fail/error
+    exten => _XXXX.,next2+101,Hangup
+    exten => h,1,NoOp()
+    ; OSP report usage
+    exten => h,n,OSPFinish(${HANGUPCAUSE})
+    
+    [macro-inbound]
+    exten => s,1,NoOp(inbound)
+    ; Get inbound protocol
+    exten => s,n,Set(CHANTECH=${CUT(CHANNEL,/,1)})
+    exten => s,n,GoToIf($["${CHANTECH}"="H323"]?100)
+    exten => s,n,GoToIf($["${CHANTECH}"="IAX2"]?200)
+    exten => s,n,GoToIf($["${CHANTECH}"="SIP"]?300)
+    exten => s,n,GoTo(1000)
+    ; H323 --------------------------------------------------------
+    ; Get peer IP
+    exten => s,100,Set(OSPPEERIP=${H323CHANINFO(peerip)})
+    ; Get OSP token
+    exten => s,n,Set(OSPINTOKEN=${H323CHANINFO(osptoken)})
+    exten => s,n,GoTo(1000)
+    ; IAX ----------------------------------------------------------
+    ; Get peer IP
+    exten => s,200,Set(OSPPEERIP=${IAXPEER(CURRENTCHANNEL)})
+    ; Get OSP token
+    exten => s,n,Set(OSPINTOKEN=${IAXCHANINFO(osptoken)})
+    exten => s,n,GoTo(1000)
+    ; SIP ----------------------------------------------------------
+    ; Get peer IP
+    exten => s,300,Set(OSPPEERIP=${SIPCHANINFO(peerip)})
+    ; Get OSP token
+    exten => s,n,Set(OSPINTOKEN=${SIP_HEADER(P-OSP-Auth-Token)})
+    exten => s,n,GoTo(1000)
+    ; --------------------------------------------------------------
+    exten => s,1000,MacroExit
+    
+    [macro-outbound]
+    exten => s,1,NoOp(outbound)
+    ; Set calling number which may be translated
+    exten => s,n,Set(CALLERID(number)=${OSPCALLING})
+    ; Check destinatio protocol
+    exten => s,n,GoToIf($["${OSPTECH}"="H323"]?100)
+    exten => s,n,GoToIf($["${OSPTECH}"="IAX2"]?200)
+    exten => s,n,GoToIf($["${OSPTECH}"="SIP"]?300)
+    ; Something wrong
+    exten => s,n,Hangup
+    exten => s,n,GoTo(1000)
+    ; H323 --------------------------------------------------------
+    ; Set call id
+    exten => s,100,Set(H323CHANINFO(callid)=${OSPOUTCALLID})
+    ; Set OSP token
+    exten => s,n,Set(H323CHANINFO(osptoken)=${OSPOUTTOKEN})
+    exten => s,n,GoTo(1000)
+    ; IAX ----------------------------------------------------------
+    ; Set OSP token
+    exten => s,200,Set(IAXCHANINFO(osptoken)=${OSPOUTTOKEN})
+    exten => s,n,GoTo(1000)
+    ; SIP ----------------------------------------------------------
+    exten => s,300,GoTo(1000)
+    ; --------------------------------------------------------------
+    exten => s,1000,MacroExit
+
+3.1.4 zapata/sip/iax/h323/ooh323.conf
+  There is no configuration required for OSP.
+
+3.2 OSP Dial Plan Functions
+  This section provides a description of each OSP dial plan function.
+
+3.2.1 OSPAuth 
   OSP token validation function.
   Input:
   * OSPPEERIP: last hop IP address
   * OSPINTOKEN: inbound OSP token
-  * provider: OSP service provider configured in osp.conf. If it is empty, 
-  default provider is used.
+  * provider: OSP service provider configured in osp.conf. If it is empty, default provider is used.
   * priority jump
   Output:
   * OSPINHANDLE: inbound OSP transaction handle
   * OSPINTIMELIMIT: inbound call duration limit
   * OSPAUTHSTATUS: OSPAuth return value. SUCCESS/FAILED/ERROR
 
-3.1.2 OSPLookup
+3.2.2 OSPLookup
   OSP lookup function.
   Input:
   * OSPPEERIP: last hop IP address
   * OSPINHANDLE: inbound OSP transaction handle
   * OSPINTIMELIMIT: inbound call duration limit 
   * exten: called number
-  * provider: OSP service provider configured in osp.conf. If it is empty, 
-  default provider is used.
+  * provider: OSP service provider configured in osp.conf. If it is empty, default provider is used.
   * priority jump
+  * callidtypes: Generate call ID for the outbound call. h: H.323; s: SIP; i: IAX. Only h, H.323, has been implemented.
   Output:
   * OSPOUTHANDLE: outbound transaction handle
   * OSPTECH: outbound protocol
-  * OSPDEST: outbound destination
+  * OSPDEST: outbound destination IP address
+  * OSPCALLED: outbound called nummber
   * OSPCALLING: outbound calling number
   * OSPOUTTOKEN: outbound OSP token
-  * OSPRESULTS: number of remain destinations
+  * OSPRESULTS: number of remaining destinations
   * OSPOUTTIMELIMIT: outbound call duration limit
+  * OSPOUTCALLIDTYPES: same as input callidtypes
+  * OSPOUTCALLID: outbound call ID. Only for H.323
+  * OSPDIALSTR: outbound dial string
   * OSPLOOKUPSTATUS: OSPLookup return value. SUCCESS/FAILED/ERROR
 
-3.1.3 OSPNext
+3.2.3 OSPNext
   OSP lookup next function.
   Input:
   * OSPINHANDLE: inbound transaction handle
   * OSPOUTHANDLE: outbound transaction handle
   * OSPINTIMELIMIT: inbound call duration limit 
+  * OSPOUTCALLIDTYPES: types of call ID generated by Asterisk.
   * OSPRESULTS: number of remain destinations
   * cause: last destination disconnect cause
   * priority jump
   Output:
   * OSPTECH: outbound protocol
-  * OSPDEST: outbound destination
+  * OSPDEST: outbound destination IP address
+  * OSPCALLED: outbound called number
   * OSPCALLING: outbound calling number
   * OSPOUTTOKEN: outbound OSP token
   * OSPRESULTS: number of remain destinations
   * OSPOUTTIMELIMIT: outbound call duration limit
+  * OSPOUTCALLID: outbound call ID. Only for H.323
+  * OSPDIALSTR: outbound dial string
   * OSPNEXTSTATUS: OSPLookup return value. SUCCESS/FAILED/ERROR
 
-3.1.4 OSPFinish
+3.2.4 OSPFinish
   OSP report usage function.
   Input:
   * OSPINHANDLE: inbound transaction handle
@@ -288,176 +572,233 @@ TransNexus and OSP Secured are trademarks of TransNexus, Inc.
   Output:
   * OSPFINISHSTATUS: OSPLookup return value. SUCCESS/FAILED/ERROR
 
-3.2 Build with OSP Support
-  If the OSP Toolkit is installed in the default install directory, /usr/local,
-  no additional configuration is required. If the OSP Toolkit is installed in 
-  another directory, say /myosp, Asterisk must be configured with the location 
-  of the OSP Toolkit.
-    --with-osptk=/myosp
-  Note: Please change the install path only if you familiar with both the OSP 
-  Toolkit and the Asterisk. Otherwise, the change may results Asterisk not 
-  supporting the OSP protocol.
-  Now, you can compile Asterisk according to the instructions provided with the
-  Asterisk distribution.
-
-3.3 Configure with OSP Support
+3.3 extensions.conf Examples
+  The extensions.conf file example provided in Section 3.1 is designed to 
+  handle all OSP call scenarios when Asterisk is used as a source or destination 
+  gateway to the PSTN or as a proxy between VoIP networks.  The extenstion.conf 
+  examples in this section are designed for specific use cases only.
 
-3.3.1 osp.conf
-  ;
-  ; Open Settlement Protocol Sample Configuration File
-  ;
-  ; This file contains configuration of providers that
-  ; are used by the OSP subsystem of Asterisk.  The section
-  ; "general" is reserved for global options.  Each other 
-  ; section declares an OSP Provider.  The provider "default"
-  ; is used when no provider is otherwise specified.
-  ;
-  [general]
-  ;
-  ; Should hardware acceleration be enabled?  May not be changed
-  ; on a reload.
-  ;
-  accelerate=no
-  ;
-  ; Defines the token format that Asterisk can validate. 
-  ; 0 - signed tokens only 
-  ; 1 - unsigned tokens only 
-  ; 2 - both signed and unsigned
-  ; The defaults to 0, i.e. the Asterisk can validate signed tokens only.
-  ;
-  tokenformat=0
-  ;
-  [default]
-  ;
-  ; All paths are presumed to be under /var/lib/asterisk/keys unless
-  ; the path begins with '/'
-  ;
-  ; Specify the private keyfile.  If unspecified, defaults to the name
-  ; of the section followed by "-privatekey.pem" (e.g. default-privatekey.pem)
-  ;
-  privatekey=pkey.pem
-  ;
-  ; Specify the local certificate file.  If unspecified, defaults to
-  ; the name of the section followed by "-localcert.pem"
-  ;
-  localcert=localcert.pem
-  ;
-  ; Specify one or more Certificate Authority keys.  If none are listed,
-  ; a single one is added with the name "-cacert.pem"
-  ;
-  cacert=cacert_0.pem
-  ;
-  ; Specific parameters can be tuned as well:
-  ;
-  ; maxconnections: Max number of simultaneous connections to the provider (default=20)
-  ; retrydelay:     Extra delay between retries (default=0)
-  ; retrylimit:     Max number of retries before giving up (default=2)
-  ; timeout:        Timeout for response in milliseconds (default=500)
-  ;
-  maxconnections=20
-  retrydelay=0
-  retrylimit=2
-  timeout=500
-  ;
-  ; List all service points for this provider
-  ;
-  ;servicepoint=http://osptestserver.transnexus.com:1080/osp
-  servicepoint=http://OSP server IP:1080/osp
-  ;
-  ; Set the "source" for requesting authorization
-  ;
-  ;source=foo
-  source=[host IP]
-  ;
-  ; Set the authentication policy.  
-  ; 0 - NO 
-  ; 1 - YES
-  ; 2 - EXCLUSIVE
-  ; Default is 1, validate token but allow no token.
-  ;
-  authpolicy=1
-
-3.3.2 zapata/sip/iax.conf
-  There is no configuration required for OSP.
+3.3.1 Source Gateway
+  The examples in this section apply when the Asterisk server is being used as 
+  a TDM to VoIP gateway.  Calls originate on the TDM network and are converted 
+  to VoIP by Asterisk.  In these cases, the Asterisk server queries an OSP 
+  server to find a route to a VoIP destination.  When the call ends, Asterisk 
+  sends a CDR to the OSP server.
+  For SIP protocol.
+    [SIPSrcGW]
+    exten => _XXXX.,1,NoOp(SIPSrcGW)
+    ; Set calling number if necessary
+    exten => _XXXX.,n,Set(CALLERID(numner)=CallingNumber)
+    ; OSP lookup using default provider, if fail/error jump to lookup+101
+    exten => _XXXX.,n(lookup),OSPLookup(${EXTEN}||j)
+    ; Set calling number which may be translated 
+    exten => _XXXX.,n,Set(CALLERID(number)=${OSPCALLING})
+    ; Dial to destination, 60 timeout, with call duration limit
+    exten => _XXXX.,n,Dial(${OSPDIALSTR},60,oL($[${OSPOUTTIMELIMIT}*1000]))
+    ; Wait 3 seconds
+    exten => _XXXX.,n,Wait,3
+    ; Hangup
+    exten => _XXXX.,n,Hangup
+    ; Deal with OSPLookup fail/error
+    exten => _XXXX.,lookup+101,Hangup
+    exten => h,1,NoOp()
+    ; OSP report usage
+    exten => h,n,OSPFinish(${HANGUPCAUSE})
+  For IAX protocol.
+    [IAXSrcGW]
+    exten => _XXXX.,1,NoOp(IAXSrcGW)
+    ; Set calling number if necessary
+    exten => _XXXX.,n,Set(CALLERID(numner)=CallingNumber)
+    ; OSP lookup using default provider, if fail/error jump to lookup+101
+    exten => _XXXX.,n(lookup),OSPLookup(${EXTEN}||j)
+    ; Set outbound OSP token
+    exten => _XXXX.,n,Set(IAXCHANINFO(osptoken)=${OSPOUTTOKEN})
+    ; Set calling number which may be translated 
+    exten => _XXXX.,n,Set(CALLERID(number)=${OSPCALLING})
+    ; Dial to destination, 60 timeout, with call duration limit
+    exten => _XXXX.,n,Dial(${OSPDIALSTR},60,oL($[${OSPOUTTIMELIMIT}*1000]))
+    ; Wait 3 seconds
+    exten => _XXXX.,n,Wait,3
+    ; Hangup
+    exten => _XXXX.,n,Hangup
+    ; Deal with OSPLookup fail/error
+    exten => _XXXX.,lookup+101,Hangup
+    exten => h,1,NoOp()
+    ; OSP report usage
+    exten => h,n,OSPFinish(${HANGUPCAUSE})
+  For H.323 protocol.
+    [H323SrcGW]
+    exten => _XXXX.,1,NoOp(H323SrcGW)
+    ; Set calling number if necessary
+    exten => _XXXX.,n,Set(CALLERID(numner)=CallingNumber)
+    ; OSP lookup using default provider, if fail/error jump to lookup+101
+    ; "h" parameter is used to generate a call id
+    ; Cisco OSP gateways use this call id to validate OSP token
+    exten => _XXXX.,n(lookup),OSPLookup(${EXTEN}||jh)
+    ; Set outbound call id
+    exten => _XXXX.,n,Set(OH323CHANINFO(callid)=${OSPOUTCALLID})
+    ; Set outbound OSP token
+    exten => _XXXX.,n,Set(OH323CHANINFO(osptoken)=${OSPOUTTOKEN})
+    ; Set calling number which may be translated 
+    exten => _XXXX.,n,Set(CALLERID(number)=${OSPCALLING})
+    ; Dial to destination, 60 timeout, with call duration limit
+    exten => _XXXX.,n,Dial(${OSPDIALSTR},60,oL($[${OSPOUTTIMELIMIT}*1000]))
+    ; Wait 3 seconds
+    exten => _XXXX.,n,Wait,3
+    ; Hangup
+    exten => _XXXX.,n,Hangup
+    ; Deal with OSPLookup fail/error
+    exten => _XXXX.,lookup+101,Hangup
+    exten => h,1,NoOp()
+    ; OSP report usage
+    exten => h,n,OSPFinish(${HANGUPCAUSE})
 
-3.3.3 extensions.conf
-  An Asterisk box can be configured as OSP source/destination gateway or OSP proxy.
+3.3.2 Destination Gateway
+  The examples in this section apply when Asterisk is being used as a VoIP to 
+  TDM gateway.  VoIP calls are received by Asterisk which validates the OSP 
+  peering token and completes to the TDM network.  After the call ends, 
+  Asterisk sends a CDR to the OSP server.
+  For SIP protocol
+    [SIPDstGW]
+    exten => _XXXX.,1,NoOp(SIPDstGW)
+    ; Get peer IP
+    exten => _XXXX.,n,Set(OSPPEERIP=${SIPCHANINFO(peerip)})
+    ; Get OSP token
+    exten => _XXXX.,n,Set(OSPINTOKEN=${SIP_HEADER(P-OSP-Auth-Token)})
+    ; Validate token using default provider, if fail/error jump to auth+101
+    exten => _XXXX.,n(auth),OSPAuth(|j)
+    ; Ringing
+    exten => _XXXX.,n,Ringing
+    ; Wait 1 second
+    exten => _XXXX.,n,Wait,1
+    ; Dial phone, timeout 15 seconds, with call duration limit
+    exten => _XXXX.,n,Dial(${DIALOUTANALOG}/${EXTEN:1},15,oL($[${OSPINTIMELIMIT}*1000]))
+    ; Wait 3 seconds
+    exten => _XXXX.,n,Wait,3
+    ; Hangup
+    exten => _XXXX.,n,Hangup
+    ; Deal with OSPAuth fail/error
+    exten => _XXXX.,auth+101,Hangup
+    exten => h,1,NoOp()
+    ; OSP report usage
+    exten => h,n,OSPFinish(${HANGUPCAUSE})
+  For IAX protocol
+    [IAXDstGW]
+    exten => _XXXX.,1,NoOp(IAXDstGW)
+    ; Get peer IP
+    exten => _XXXX.,n,Set(OSPPEERIP=${IAXPEER(CURRENTCHANNEL)})
+    ; Get OSP token
+    exten => _XXXX.,n,Set(OSPINTOKEN=${IAXCHANINFO(osptoken)})
+    ; Validate token using default provider, if fail/error jump to auth+101
+    exten => _XXXX.,n(auth),OSPAuth(|j)
+    ; Ringing
+    exten => _XXXX.,n,Ringing
+    ; Wait 1 second
+    exten => _XXXX.,n,Wait,1
+    ; Dial phone, timeout 15 seconds, with call duration limit
+    exten => _XXXX.,n,Dial(${DIALOUTANALOG}/${EXTEN:1},15,oL($[${OSPINTIMELIMIT}*1000]))
+    ; Wait 3 seconds
+    exten => _XXXX.,n,Wait,3
+    ; Hangup
+    exten => _XXXX.,n,Hangup
+    ; Deal with OSPAuth fail/error
+    exten => _XXXX.,auth+101,Hangup
+    exten => h,1,NoOp()
+    ; OSP report usage
+    exten => h,n,OSPFinish(${HANGUPCAUSE})
+  For H.323 protocol
+    [H323DstGW]
+    exten => _XXXX.,1,NoOp(H323DstGW)
+    ; Get peer IP
+    exten => _XXXX.,n,Set(OSPPEERIP=${H323CHANINFO(peerip)})
+    ; Get OSP token
+    exten => _XXXX.,n,Set(OSPINTOKEN=${H323CHANINFO(osptoken)})
+    ; Validate token using default provider, if fail/error jump to auth+101
+    exten => _XXXX.,n(auth),OSPAuth(|j)
+    ; Ringing
+    exten => _XXXX.,n,Ringing
+    ; Wait 1 second
+    exten => _XXXX.,n,Wait,1
+    ; Dial phone, timeout 15 seconds, with call duration limit
+    exten => _XXXX.,n,Dial(${DIALOUTANALOG}/${EXTEN:1},15,oL($[${OSPINTIMELIMIT}*1000]))
+    ; Wait 3 seconds
+    exten => _XXXX.,n,Wait,3
+    ; Hangup
+    exten => _XXXX.,n,Hangup
+    ; Deal with OSPAuth fail/error
+    exten => _XXXX.,auth+101,Hangup
+    exten => h,1,NoOp()
+    ; OSP report usage
+    exten => h,n,OSPFinish(${HANGUPCAUSE})
 
-3.3.3.1 OSP Source Gateway
-  [PhoneSrcGW]
-  ; Set calling number if necessary
-  exten => _XXXX.,1,Set(CALLERID(numner)=CallingNumber)
-  ; OSP lookup using default provider, if fail/error jump to 2+101
-  exten => _XXXX.,2,OSPLookup(${EXTEN}||j)
-  ; Set calling number which may be translated 
-  exten => _XXXX.,3,Set(CALLERID(number)=${OSPCALLING})
-  ; Dial to destination, 60 timeout, with call duration limit
-  exten => _XXXX.,4,Dial(${OSPTECH}/${OSPDEST},60,oL($[${OSPOUTTIMELIMIT}*1000]))
-  ; Wait 3 seconds
-  exten => _XXXX.,5,Wait,3
-  ; Hangup
-  exten => _XXXX.,6,Hangup
-  ; Deal with OSPLookup fail/error
-  exten => _XXXX.,2+101,Hangup
-  ; OSP report usage
-  exten => h,1,OSPFinish(${HANGUPCAUSE})
-  3.3.3.2 OSP Destination Gateway
-  [PhoneDstGW]
-  ; Get peer IP
-  exten => _XXXX.,1,Set(OSPPEERIP=${SIPCHANINFO(peerip)})
-  ; Get OSP token
-  exten => _XXXX.,2,Set(OSPINTOKEN=${SIP_HEADER(P-OSP-Auth-Token)})
-  ; Validate token using default provider, if fail/error jump to 3+101
-  exten => _XXXX.,3,OSPAuth(|j)
-  ; Ringing
-  exten => _XXXX.,4,Ringing
-  ; Wait 1 second
-  exten => _XXXX.,5,Wait,1
-  ; Dial phone, timeout 15 seconds, with call duration limit
-  exten => _XXXX.,6,Dial(${DIALOUTANALOG}/${EXTEN:1},15,oL($[${OSPINTIMELIMIT}*1000]))
-  ; Wait 3 seconds
-  exten => _XXXX.,7,Wait,3
-  ; Hangup
-  exten => _XXXX.,8,Hangup
-  ; Deal with OSPAuth fail/error
-  exten => _XXXX.,3+101,Hangup
-  ; OSP report usage
-  exten => h,1,OSPFinish(${HANGUPCAUSE})
-  3.3.3.3 Proxy
-  [GeneralProxy]
-  ; Get peer IP
-  exten => _XXXX.,1,Set(OSPPEERIP=${SIPCHANINFO(peerip)})
-  ; Get OSP token
-  exten => _XXXX.,2,Set(OSPINTOKEN=${SIP_HEADER(P-OSP-Auth-Token)})
-  ; Validate token using default provider, if fail/error jump to 3+101
-  exten => _XXXX.,3,OSPAuth(|j)
-  ; OSP lookup using default provider, if fail/error jump to 4+101
-  exten => _XXXX.,4,OSPLookup(${EXTEN}||j)
-  ; Set calling number which may be translated 
-  exten => _XXXX.,5,Set(CALLERID(number)=${OSPCALLING})
-  ; Dial to 1st destination, 60 timeout, with call duration limit
-  exten => _XXXX.,6,Dial(${OSPTECH}/${OSPDEST},24,oL($[${OSPOUTTIMELIMIT}*1000]))
-  ; OSP lookup next, if fail/error jump to 7+101
-  exten => _XXXX.,7,OSPNext(${HANGUPCAUSE}||j)
-  ; Set calling number which may be translated 
-  exten => _XXXX.,8,Set(CALLERID(number)=${OSPCALLING})
-  ; Dial to 2nd destination, 60 timeout, with call duration limit
-  exten => _XXXX.,9,Dial(${OSPTECH}/${OSPDEST},25,oL($[${OSPOUTTIMELIMIT}*1000]))
-  ; OSP lookup next, if fail/error jump to 10+101
-  exten => _XXXX.,10,OSPNext(${HANGUPCAUSE}||j)
-  ; Set calling number which may be translated 
-  exten => _XXXX.,11,Set(CALLERID(number)=${OSPCALLING})
-  ; Dial to 3rd destination, 60 timeout, with call duration limit
-  exten => _XXXX.,12,Dial(${OSPTECH}/${OSPDEST},26,oL($[${OSPOUTTIMELIMIT}*1000]))
-  ; Hangup
-  exten => _XXXX.,13,Hangup
-  ; Deal with OSPAuth fail/error
-  exten => _XXXX.,3+101,Hangup
-  ; Deal with OSPLookup fail/error
-  exten => _XXXX.,4+101,Hangup
-  ; Deal with 1st OSPNext fail/error
-  exten => _XXXX.,7+101,Hangup
-  ; Deal with 2nd OSPNext fail/error
-  exten => _XXXX.,10+101,Hangup
-  ; OSP report usage
-  exten => h,1,OSPFinish(${HANGUPCAUSE})
+3.3.3 Proxy
+  The example in this section applies when Asterisk is a proxy between two VoIP networks.
+    [GeneralProxy]
+    exten => _XXXX.,1,NoOp(GeneralProxy)
+    ; Get peer IP and inbound OSP token
+    ; SIP, un-comment the following two lines.
+    ;exten => _XXXX.,n,Set(OSPPEERIP=${SIPCHANINFO(peerip)})
+    ;exten => _XXXX.,n,Set(OSPINTOKEN=${SIP_HEADER(P-OSP-Auth-Token)})
+    ; IAX, un-comment the following 2 lines
+    ;exten => _XXXX.,n,Set(OSPPEERIP=${IAXPEER(CURRENTCHANNEL)})
+    ;exten => _XXXX.,n,Set(OSPINTOKEN=${IAXCHANINFO(osptoken)})
+    ; H323, un-comment the following two lines.
+    ;exten => _XXXX.,n,Set(OSPPEERIP=${OH323CHANINFO(peerip)})
+    ;exten => _XXXX.,n,Set(OSPINTOKEN=${OH323CHANINFO(osptoken)})
+    ;---------------------------------------------------------------
+    ; Validate token using default provider, if fail/error jump to auth+101
+    exten => _XXXX.,n(auth),OSPAuth(|j)
+    ; OSP lookup using default provider, if fail/error jump to lookup+101
+    ; "h" parameter is used to generate a call id for H.323 destinations
+    ; Cisco OSP gateways use this call id to validate OSP token
+    exten => _XXXX.,n(lookup),OSPLookup(${EXTEN}||jh)
+    ; Set outbound call id and OSP token
+    ; IAX, un-comment the following line. 
+    ;exten => _XXXX.,n,Set(IAXCHANINFO(osptoken)=${OSPOUTTOKEN})
+    ; H323, un-comment the following two lines. 
+    ;exten => _XXXX.,n,Set(OH323CHANINFO(callid)=${OSPOUTCALLID})
+    ;exten => _XXXX.,n,Set(OH323CHANINFO(osptoken)=${OSPOUTTOKEN})
+    ;---------------------------------------------------------------
+    ; Set calling number which may be translated 
+    exten => _XXXX.,n,Set(CALLERID(number)=${OSPCALLING})
+    ; Dial to destination, 14 timeout, with call duration limit
+    exten => _XXXX.,n,Dial(${OSPDIALSTR},14,oL($[${OSPOUTTIMELIMIT}*1000]))
+    ; OSP lookup next destination using default provider, if fail/error jump to next1+101
+    exten => _XXXX.,n(next1),OSPNext(${HANGUPCAUSE}||j)
+    ; Set outbound call id and OSP token
+    ; IAX, un-comment the following line. 
+    ;exten => _XXXX.,n,Set(IAXCHANINFO(osptoken)=${OSPOUTTOKEN})
+    ; H323, un-comment the following two lines.
+    ;exten => _XXXX.,n,Set(OH323CHANINFO(callid)=${OSPOUTCALLID})
+    ;exten => _XXXX.,n,Set(OH323CHANINFO(osptoken)=${OSPOUTTOKEN})
+    ;---------------------------------------------------------------
+    ; Set calling number which may be translated 
+    exten => _XXXX.,n,Set(CALLERID(number)=${OSPCALLING})
+    ; Dial to destination, 15 timeout, with call duration limit
+    exten => _XXXX.,n,Dial(${OSPDIALSTR},15,oL($[${OSPOUTTIMELIMIT}*1000]))
+    ; OSP lookup next destination using default provider, if fail/error jump to next2+101
+    exten => _XXXX.,n(next2),OSPNext(${HANGUPCAUSE}||j)
+    ; Set outbound call id and OSP token
+    ; IAX, un-comment the following line. 
+    ;exten => _XXXX.,n,Set(IAXCHANINFO(osptoken)=${OSPOUTTOKEN})
+    ; H323, un-comment the following two lines.
+    ;exten => _XXXX.,n,Set(OH323CHANINFO(callid)=${OSPOUTCALLID})
+    ;exten => _XXXX.,n,Set(OH323CHANINFO(osptoken)=${OSPOUTTOKEN})
+    ;---------------------------------------------------------------
+    ; Set calling number which may be translated 
+    exten => _XXXX.,n,Set(CALLERID(number)=${OSPCALLING})
+    ; Dial to destination, 16 timeout, with call duration limit
+    exten => _XXXX.,n,Dial(${OSPDIALSTR},16,oL($[${OSPOUTTIMELIMIT}*1000]))
+    ; Hangup
+    exten => _XXXX.,n,Hangup
+    ; Deal with OSPAuth fail/error
+    exten => _XXXX.,auth+101,Hangup
+    ; Deal with OSPLookup fail/error
+    exten => _XXXX.,lookup+101,Hangup
+    ; Deal with 1st OSPNext fail/error
+    exten => _XXXX.,next1+101,Hangup
+    ; Deal with 2nd OSPNext fail/error
+    exten => _XXXX.,next2+101,Hangup
+    exten => h,1,NoOp()
+    ; OSP report usage
+    exten => h,n,OSPFinish(${HANGUPCAUSE})