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2010-06-17fixes some coding guideline issuedvossel1-6/+6
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@271300 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-17retransmit response to BYE requests until timer J expiresdvossel3-7/+34
According to RFC 3261 section 17.2.2, which describes non-INVITE server transaction, when a dialog enters the Completed state it must destroy the dialog after Timer J (T1*64) fires. For a BYE transaction Asterisk terminates the dialog immediately during sip_hangup() when it should be waiting T1*64 ms. This results in some odd behavior. For instance if Asterisk receives a BYE and transmits a 200ok in response, if the endpoint never receives the 200ok it will retransmit the BYE to which Asterisk responds with a "481 Call leg/transaction does not exist" because the dialog is already gone. To resolve this I made a function called sip_scheddestroy_final(). This differs slightly from sip_schedestroy() in that it enables a flag that will prevent the destruction from ever being rescheduled or canceled afterwards. It also prevents the pvt's needdestroy flag from being set which triggers the destruction of the dialog within the do_monitor thread(). By using this function we are guaranteed destruction will not occur until the scheduled time. This allows Asterisk to respond to any possible retransmits for a dialog after we process the initial BYE request for T1*64 ms. Other changes: I removed two instances where sip_cancel_destroy is used right before calling sip_scheddestroy. sip_scheddestroy always calls sip_cancel_destroy before scheduling the new destruction so it is completely unnecessary. Review: https://reviewboard.asterisk.org/r/694/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@271262 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-17Change expected operation from error to debug messagejpeeler1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@271192 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-16addition of more parse_uri test casesdvossel1-0/+54
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@271056 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-16Fix the actual place that was pointed out, for previous commit.qwell1-0/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@270983 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-16Merged revisions 270980 via svnmerge from qwell1-0/+2
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r270980 | qwell | 2010-06-16 16:10:09 -0500 (Wed, 16 Jun 2010) | 4 lines Need to lock the agent chan before access its internal bits. Pointed out by russellb on asterisk-dev mailing list. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@270981 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-16addition of G.719 pass-through supportdvossel2-0/+14
(closes issue #16293) Reported by: malcolmd Patches: g719.passthrough.patch.7 uploaded by malcolmd (license 924) format_g719.c uploaded by malcolmd (license 924) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@270940 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-16Merged revisions 270866 via svnmerge from dvossel1-6/+4
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r270866 | dvossel | 2010-06-16 12:35:29 -0500 (Wed, 16 Jun 2010) | 22 lines fixes chan_iax2 race condition There is code in chan_iax2.c that attempts to guarantee that only a single active thread will handle a call number at a time. This code works once the thread is added to an active_list of threads, but we are not currently guaranteed that a newly activated thread will enter the active_list immediately because it is left up to the thread to add itself after frames have been queued to it. This means that if two frames come in for the same call number at the same time, it is possible for them to grab two separate threads because the first thread did not add itself to the active_list fast enough. This causes some pretty complex problems. This patch resolves this race condition by immediately adding an activated thread to the active_list within the network thread and only depending on the thread to remove itself once it is done processing the frames queued to it. By doing this we are guaranteed that if another frame for the same call number comes in at the same time, that this thread will immediately be found in the active_list of threads. Review: https://reviewboard.asterisk.org/r/720/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@270867 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-16Fix no call waiting caller IDjpeeler1-1/+0
Clearing the callwaitcas flag in analog_call was causing the incoming D digit to be ignored which triggers sending the caller ID. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@270836 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-15Don't blow up if an ast_channel doesn't get allocated.russell1-1/+3
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@270726 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-15Make contactdeny apply to src ip when nat=yestwilson1-21/+22
chan_sip's "contactdeny" feature screens the "to be registered contact". In case of nat=yes it should not use the address information from the Contact header (which is not used at all for routing), but the source IP address of the request. Thus, if nat=yes and a client sends a request from a denied IP address (e.g. by spoofing the src-IP address) it can bypass the screening. This commit makes contactdeny apply to the src ip when nat=yes instead. (closes issue #17276) Reported by: klaus3000 Patches: patch-asterisk-trunk-contactdeny.txt uploaded by klaus3000 (license 65) Tested by: klaus3000 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@270658 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-14Extract sig_ss7_init_linkset() to sig_ss7.rmudgett3-16/+28
Also found a place where sig_pri_init_pri() was inlined and called it instead. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@270298 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-14Add option to get untruncated channel name from AGENT function.qwell1-0/+7
The "channel" option would chop the channel name at the last '-', which made it useless for something like a channel transfer from the dialplan. The "fullchannel" option will return the channel name as-is. ABE-2218 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@270260 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-14Add digit manipulation tag support to chan_dahdi/sig_pri like chan_misdn.rmudgett3-1/+77
Add the append_msn_to_cid_tag option to chan_dahdi like chan_misdn. Review: https://reviewboard.asterisk.org/r/696/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@270219 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-10Attempt to fix a FreeBSD build error by including sys/stat.h.russell1-0/+1
http://bamboo.asterisk.org/download/AST-TRUNKFREEBSD/build_logs/AST-TRUNKFREEBSD-187.log git-svn-id: http://svn.digium.com/svn/asterisk/trunk@269602 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-09Merged revisions 269495 via svnmerge from russell1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r269495 | russell | 2010-06-09 17:18:37 -0500 (Wed, 09 Jun 2010) | 2 lines Don't stop Asterisk if chan_oss fails to register 'Console' (due to another channel driver already claiming it). ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@269497 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-09Add missing API function to sig_ss7: sig_ss7_fixup().rmudgett3-7/+32
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@269308 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-09Eliminate deadlock potential in dahdi_fixup().rmudgett1-3/+4
Calling dahdi_indicate() within dahdi_fixup() while the owner pointers are in a potentially inconsistent state is a potentially bad thing in principle. However, calling dahdi_indicate() when the channel private lock is already held can cause a deadlock if the PRI lock is needed because dahdi_indicate() will also get the channel private lock. The pri_grab() function assumes that the channel private lock is held once to avoid deadlock. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@269307 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-09dial by name in chan_dahditzafrir1-10/+122
* chan_dahdi supports dialing configuring and dialing by device file name. DAHDI/span-name!local!1 will use /dev/dahdi/span-name/local/1 . Likewise it may appear in chan_dahdi.conf as 'channel => span-name!local!1'. * A new options for chan_dahdi.conf: 'ignore_failed_channels'. Boolean. False by default. If set, chan_dahdi will ignore failed 'channel' entries. Handy for the above name-based syntax as it does not depend on initialization order. * have my_pri_make_cc_dialstring() only manupulate dial-strings of group (gGrR) dialing, which make it lsightly more complicated. https://reviewboard.asterisk.org/r/535/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@269238 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-08Fix some doxygen warnings.lmadsen3-10/+11
(closes issue #17336) Reported by: snuffy Patches: doxygen-fixes1.diff uploaded by snuffy (license 35) Tested by: russell git-svn-id: http://svn.digium.com/svn/asterisk/trunk@268969 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-08Add SRTP support for Asterisktwilson8-18/+781
After 5 years in mantis and over a year on reviewboard, SRTP support is finally being comitted. This includes generic CHANNEL dialplan functions that work for getting the status of whether a call has secure media or signaling as defined by the underlying channel technology and for setting whether or not a new channel being bridged to a calling channel should have secure signaling or media. See doc/tex/secure-calls.tex for examples. Original patch by mikma, updated for trunk and revised by me. (closes issue #5413) Reported by: mikma Tested by: twilson, notthematrix, hemanshurpatel Review: https://reviewboard.asterisk.org/r/191/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@268894 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-08Make SIP tests compile again.rmudgett1-3/+3
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@268857 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-07Use the mailbox destructor function, instead.tilghman1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@268818 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-07Mailbox list would previously grow at each reload, containing duplicates.tilghman2-12/+45
Also, optimize the allocation of mailboxes to avoid additional memory structures. (closes issue #16320) Reported by: Marquis Patches: 20100525__issue16320.diff.txt uploaded by tilghman (license 14) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@268817 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-07Extract sig_ss7 out of chan_dahdi.rmudgett6-1581/+2623
Extract the SS7 specific code out of chan_dahdi like what was done to ISDN/PRI and analog signaling. The new SS7 structures were modeled on sig_pri. The changes to sig_pri are an enhancement and a bug fix made possible because SS7 was extracted. 1) The sig_pri TRANSFERCAPABILITY channel variable should have been set unconditionally in sig_pri_new_ast_channel(). 2) SS7/PRI transfer capability interaction in dahdi_new() fixed because of SS7 extraction. 3) Module ref count error in dahdi_new() if startpbx failed to start the PBX for some reason. Review: https://reviewboard.asterisk.org/r/661/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@268774 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-07Seems strange (and the code backs up) that if the max and min of a statistic ↵tilghman2-4/+4
is expressed as a double, the last value would not also need to be a double. (closes issue #15807) Reported by: klaus3000 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@268773 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-07Moved AOC request code out of the middle of code parsing the dialed number.rmudgett1-15/+14
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@268734 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-06Finally track down and eliminate the "FRACK! warnings from chan_iax2".tilghman1-0/+9
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@268495 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-04fixes compile error from uninitialized variabledvossel1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@268281 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-04RFC3261 compliant sip unreliable retransmit timing + 'registerattempts' ↵dvossel1-6/+9
option tweak Changes. 1. RFC 3261 states in section 17.1.2.2 and 17.1.1.2 that retransmission timers should initially be set to timer T1. T1 by default is 500ms. Asterisk was starting the retransmission timers at T1*2 which shouldn't cause any problems, but is not RFC compliant. 2. RFC 3261 states in section 17.1.2.2 that for a non-INVITE client transaction, if the retransmit timer fires while in the proceeding state that the request must be retransmitted. Asterisk currently ack's requests for both INVITE and non-INVITE transactions when a 1XX response is received, this patch changes this for non-INVITE requests. 3. The 'registerattempts' option in sip.conf is supposed to set how many registry attempts will be made before giving up. When this option is set to 1, I would expect only one registry attempt to be made before stopping because of a failure, but instead two are made. In my opinion this is not expected behavior. This option does not indicate that these are re-attempts. The logic behind this option has been changed to only attempt registers the exact number of times this option is set to. If this option is 0, it still continues to re-attempt the registration forever. Review: https://reviewboard.asterisk.org/r/687/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@268205 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-04Incoming overlap dialing no longer works after sig_pri extraction.rmudgett1-8/+1
The problem would manifest itself if your dialplan matching could accept more digits to match than were actually dialed. The time out waiting for overlap digits disconnected the call instead of matching any accumulated digits to the dialplan. Accidental conversion of a break out of loop as a break out of switch. (closes issue #17401) Reported by: avalentin Patches: issue17401_digit_timeout.patch uploaded by rmudgett (license 664) Tested by: avalentin, rmudgett git-svn-id: http://svn.digium.com/svn/asterisk/trunk@267928 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-04Send an ACK for every final response received for an INVITEtwilson1-0/+5
From issue ABE-2247. RFC 3261 compliance for sections 13.2.24 and 17.1.1.2. Review: https://reviewboard.asterisk.org/r/692/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@267863 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-03Don't stop Asterisk if chan_usbradio isn't configured.russell1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@267537 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-03Remove a line that was killing Asterisk on startup.russell1-1/+0
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@267490 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-03Comment out a rule that likes to run implicitly unnecessarily, breaking buildsrussell1-2/+6
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@267445 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-03Add ETSI Message Waiting Indication (MWI) support.rmudgett3-1/+252
Add the ability to report waiting messages to ISDN endpoints (phones). Relevant specification: EN 300 650 and EN 300 745 Review: https://reviewboard.asterisk.org/r/599/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@267399 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-02try to fix some random chan_h323 compilation failuresrussell2-2/+1
After some debugging, the random chan_h323 build failures appear to be due to complications introduced by some chan_h323 specific build stuff getting triggered during a clean. Simplify this by moving the h323 clean commands down into channels/makefile. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@267352 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-02Add ETSI Malicious Call ID support.rmudgett1-0/+110
Add the ability to report malicious callers as an AMI event in the call event class. Relevant specification: EN 300 180 Review: https://reviewboard.asterisk.org/r/576/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@267350 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-02Add ETSI Call Waiting support.rmudgett3-153/+739
Add the ability to announce a call to an endpoint when there are no B channels available. A call waiting call is a SETUP message with no B channel selected. Relevant specification: EN 300 056, EN 300 057, EN 300 058 For DAHDI/ISDN channels, the CHANNEL() dialplan function now supports the "no_media_path" option. * Returns "0" if there is a B channel associated with the call. * Returns "1" if no B channel is associated with the call. The call is either on hold or is a call waiting call. If you are going to allow incoming call waiting calls then you need to use CHANNEL(no_media_path) do determine if you must drop a call to accept the new call. Review: https://reviewboard.asterisk.org/r/568/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@267261 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-02Prevent use of uninitialized values.mmichelson1-1/+1
Two struct sockaddr_ins are created when applying directmedia host access rules. The addresses of these are passed to the RTP engine to be filled in. However, the RTP engine inspects the fields of the structs before actually taking action. This inspection caused valgrind to be a bit unhappy. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@267097 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-02Generic Advice of Charge.rmudgett5-432/+1058
Asterisk Generic AOC Representation - Generic AOC encode/decode routines. (Generic AOC must be encoded to be passed on the wire in the AST_CONTROL_AOC frame) - AST_CONTROL_AOC frame type to represent generic encoded AOC data - Manager events for AOC-S, AOC-D, and AOC-E messages Asterisk App Support - app_dial AOC-S pass-through support on call setup - app_queue AOC-S pass-through support on call setup AOC Unit Tests - AOC Unit Tests for encode/decode routines - AOC Unit Test for manager event representation. SIP AOC Support - Pass-through of generic AOC-D and AOC-E messages to snom phones via the snom AOC specification. - Creation of chan_sip page3 flags for the addition of the new 'snom_aoc_enabled' sip.conf option. IAX AOC Support - Natively supports AOC pass-through through the use of the new AST_CONTROL_AOC frame type DAHDI AOC Support - ETSI PRI full AOC Pass-through support - 'aoc_enable' chan_dahdi.conf option for independently enabling pass-through of AOC-S, AOC-D, AOC-E. - 'aoce_delayhangup' option for retrieving AOC-E on disconnect. - DAHDI A() dial string option for requesting AOC services. example usage: ;requests AOC-S, AOC-D, and AOC-E on call setup exten=>1111,1,Dial(DAHDI/g1/1112/A(s,d,e)) Review: https://reviewboard.asterisk.org/r/552/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@267096 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-02Add ETSI Advice Of Charge (AOC) event reporting.rmudgett1-0/+683
This feature generates AMI events in the new aoc event class from the events passed up by libpri. Review: https://reviewboard.asterisk.org/r/537/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@267008 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-02Add ETSI Explicit Call Transfer (ECT) support.rmudgett3-75/+130
Added ability to send and receive ETSI Explicit Call Transfer (ECT) messages to eliminate tromboned calls. Note: Asterisk already supported initiating the transfer of calls to eliminate tromboned calls to libpri so there was nothing to do for the asterisk portion. Review: https://reviewboard.asterisk.org/r/520/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@266926 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-28fixes crash when creation of UDPTL failsdvossel1-12/+27
(closes issue #17264) Reported by: falves11 Patches: issue_17264_reviewboard_fix.diff uploaded by dvossel (license 671) issue_17264_1.6.2_reviewboard_fix.diff uploaded by dvossel (license 671) Tested by: falves11 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@266292 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-26do all sip registry parsing before transmit_registerdvossel3-95/+239
This patch breaks up every part of the sip registry string during config parsing and removes all parsing from transmit_register(). Thanks to Nick_Lewis for contributing this patch! (closes issue #14331) Reported by: Nick_Lewis Patches: chan_sip.c-domparse.patch uploaded by Nick Lewis (license 657) chan_sip.c.patch uploaded by Nick Lewis (license 657) chan_sip.c.domainparse3.patch uploaded by Nick Lewis (license 657) chan_sip.c-domparse4.patch uploaded by Nick Lewis (license 657) chan_sip.c-domparse5.patch uploaded by Nick Lewis (license 657) nicklewispatch.diff uploaded by dvossel (license 671) Tested by: Nick_Lewis, dvossel Review: https://reviewboard.asterisk.org/r/628/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@266090 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-26fixes failed SIP Directed pickup resulting in dead channeldvossel1-0/+1
(closes issue #17339) Reported by: one47 Patches: sip_magic_pickup2 uploaded by one47 (license 23) Tested by: one47, dvossel git-svn-id: http://svn.digium.com/svn/asterisk/trunk@266006 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-26.......mmichelson1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@265844 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-26Re-enable "always" option for videosupport option in sip.conf.mmichelson1-2/+3
(closes issue #17016) Reported by: twilson Patches: 17016.patch uploaded by mmichelson (license 60) Tested by: devmod git-svn-id: http://svn.digium.com/svn/asterisk/trunk@265842 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-25Properly use peer's outboundproxy for outbound REGISTERs.mmichelson1-16/+5
The logic used in transmit_register to get the outboundproxy for a peer was flawed since this value would be overridden shortly afterwards when create_addr was called. In addition, this also fixes some logic used when parsing users.conf so that the peer name is placed in the internally-generated register string so that an outboundproxy set in the Asterisk GUI will be used for outbound REGISTERs. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@265698 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-24Print openh323 log to the Asterisk console.mmichelson1-0/+2
(closes issue #17109) Reported by: under Patches: logstream.diff uploaded by under (license 914) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@265451 f38db490-d61c-443f-a65b-d21fe96a405b
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