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2010-07-14Fix errors where incorrect address information was printed.mmichelson1-9/+9
ast_sockaddr_stringiy_fmt (which is call by all ast_sockaddr_stringify* functions) uses thread-local storage for storing the string that it creates. In cases where ast_sockaddr_stringify_fmt was being called twice within the same statement, the result of one call would be overwritten by the result of the other call. This usually was happening in printf-like statements and was resulting in the same stringified addressed being printed twice instead of two separate addresses. I have fixed this by using ast_strdupa on the result of stringify functions if they are used twice within the same statement. As far as I could tell, there were no instances where a pointer to the result of such a call were saved anywhere, so this is the only situation I could see where this error could occur. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276570 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-14Make compile again.rmudgett1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276531 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-14handle special case were "200 Ok" to pending INVITE never receives ACKdvossel1-1/+19
Unlike most responses, the 200 Ok to a pending INVITE Request is acknowledged by an ACK Request. If the ACK Request for this Response is not received the previous behavior was to immediately destroy the dialog and hangup the channel. Now in an effort to be more RFC compliant, instead of immediately destroying the dialog during this special case, termination is done with a BYE Request as the dialog is technically confirmed when the 200 Ok is sent even if the ACK is never received. The behavior of immediately hanging up the channel remains. This only affects how dialog termination proceeds for this one special case. RFC 3261 section 13.3.1.4 "If the server retransmits the 2xx response for 64*T1 seconds without receiving an ACK, the dialog is confirmed, but the session SHOULD be terminated. This is accomplished with a BYE, as described in Section 15." git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276439 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-14Expand the caller ANI field to an ast_party_idrmudgett15-37/+87
Expand the ani field in ast_party_caller and ast_party_connected_line to an ast_party_id. This is an extension to the ast_callerid restructuring patch in review: https://reviewboard.asterisk.org/r/702/ Review: https://reviewboard.asterisk.org/r/744/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276393 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-14collapse debug code in retrans_pkt into separate linesdvossel1-3/+14
I've been working in this function a bunch lately, and these huge debug strings are getting annoying. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276392 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-14Do not skip sending MWI for a peer if an address is defined. Really just a ↵jpeeler1-1/+1
merge mistake from IPv6 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276389 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-14ast_callerid restructuringrmudgett21-541/+961
The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char *str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char *str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char *str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char *tag; }; struct ast_party_dialed { struct { char *str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char *ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276347 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-13chan_sip: RFC compliant retransmission timeoutdvossel2-12/+40
Retransmission of packets should not be based on how many packets were sent, but instead on a timeout period. Depending on whether or not the packet is for a INVITE or NON-INVITE transaction, the number of packets sent during the retransmission timeout period will be different, so timing out based on the number of packets sent is not accurate. This patch fixes this by removing the retransmit limit and only stopping retransmission after a timeout period is reached. By default this timeout period is 64*(Timer T1) for both INVITE and non-INVITE transactions. For more information on sip timer values refer to RFC3261 Appendix A. Review: https://reviewboard.asterisk.org/r/749/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276219 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-13Revert early destruction of RTP sessionstwilson2-26/+9
Some code improperly assumes that the sessions are still there, so revert the change until I can find all of them and fix them. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276206 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-13Destroy RTP fds when we schedule final dialog destructiontwilson2-9/+26
Since we are only keeping the dialog around for retransmissions at this point and there is no possibility that we are still handling RTP, go ahead and destroy the RTP sessions. Keeping them alive for 32 past when they are used is unnecessary and can lead to problems with having too many open file descriptors, etc. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@275998 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-12Don't try to ref authpeer when it isn't settwilson1-1/+3
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@275816 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-10Add support for devices with less than 3 lines on the LCD.russell1-28/+100
(closes issue #17600) Reported by: minaguib Patches: ast_unistim_height_v2.patch uploaded by minaguib (license 1078) Tested by: minaguib git-svn-id: http://svn.digium.com/svn/asterisk/trunk@275466 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-09Fix error in parsing SIP registry strings from ASTdb.mmichelson1-18/+22
It was essentially an off-by-one error. The easiest way to fix this was to use the handy-dandy AST_NONSTANDARD_RAW_ARGS macro to parse the pieces of the registration string out. Tested and it works wonderfully. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@275385 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-09Get more information about the Bamboo test failurestilghman1-3/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@275312 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-09Fix port parsing in check_via.mmichelson1-5/+7
If a Via header contained an IPv6 address, we would not properly parse the port. We would instead get the information after the first colon in the address. (closes issue #17614) Reported by: oej Patches: diff uploaded by sperreault (license 252) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@275308 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-09Fix an issue where the port for p->ourip was being set to 0.mmichelson1-0/+3
This should fix all the CDR tests that were not passing. When they would originate a call, all fields in the INVITE that contained the source port would have the port set to 0. Most troubling of these was the Contact header. Tests are passing locally now and should also pass on the bamboo build agents. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@275294 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-09Merged revisions 275241 via svnmerge from pabelanger1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r275241 | pabelanger | 2010-07-09 15:20:00 -0400 (Fri, 09 Jul 2010) | 8 lines Fix logging message for stale nonce. (closes issue #17582) Reported by: kenner Patches: chan_sip.c.diff uploaded by kenner (license 1040) Tested by: lmadsen ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@275249 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-09Weird, no output and Bamboo still fails...tilghman1-0/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@275227 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-09Add some diagnostic feedback to our data teststilghman1-0/+5
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@275172 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-09Kill some startup warnings and errors and make some messages more helpful in ↵tilghman5-14/+14
tracking down the source. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@275105 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-09Return logic of sip_debug_test_addr() to its original functionality.mmichelson1-1/+18
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@275104 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-09Fix sip_uri_parse test comparison.mmichelson1-2/+2
Part of the change with the IPv6 changes is to treat a host:port as a single 'domain' entity. This test was not updated to have the correct expectation after calling parse_uri(). git-svn-id: http://svn.digium.com/svn/asterisk/trunk@274984 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-09Copy the address into the peer structure after we set the default portsimon.perreault1-1/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@274947 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-08Fix calls of ast_sockaddr_from_sin() from IPv6 integration.rmudgett3-9/+9
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@274828 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-08Add IPv6 to Asterisk.mmichelson15-1081/+1372
This adds a generic API for accommodating IPv6 and IPv4 addresses within Asterisk. While many files have been updated to make use of the API, chan_sip and the RTP code are the files which actually support IPv6 addresses at the time of this commit. The way has been paved for easier upgrading for other files in the near future, though. Big thanks go to Simon Perrault, Marc Blanchet, and Jean-Philippe Dionne for their hard work on this. (closes issue #17565) Reported by: russell Patches: asteriskv6-test-report.pdf uploaded by russell (license 2) Review: https://reviewboard.asterisk.org/r/743 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@274783 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-08Implement AstData API data providers as part of the GSOC 2010 project,eliel4-2/+890
midterm evaluation. Review: https://reviewboard.asterisk.org/r/757/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@274727 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-07Fixes some ref count issues introduced by r274539dvossel1-2/+8
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@274686 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-07Add missing conditional around chan_dahdi mfcr2_skip_category config parameter.rmudgett1-0/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@274639 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-07Merged revisions 274579 via svnmerge from rmudgett1-0/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r274579 | rmudgett | 2010-07-07 13:12:41 -0500 (Wed, 07 Jul 2010) | 1 line Close the DAHDI FD on error when processing chan_dahdi toneduration config parameter. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@274595 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-07Use the relatedpeer field of a sip_pvt during INVITE processing.mmichelson1-49/+35
Review: https://reviewboard.asterisk.org/r/629 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@274539 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-06Merged revisions 274280 via svnmerge from twilson1-10/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r274280 | twilson | 2010-07-06 17:08:20 -0500 (Tue, 06 Jul 2010) | 9 lines Add option to not do a call forward on 482 Loop Detected Asterisk has always set up a forwarded call when receiving a 482 Loop Detected. This prevents handling the call failure by just continuing on in the dialplan. Since this would be a change in behavior, the new option to disable this behavior is forwardloopdetected which defaults to 'yes'. Review: https://reviewboard.asterisk.org/r/764/ ........ (no option for trunk, just changing the behavior) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@274284 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-06Status shows all non-CRC4 lines as "yellow", even if "yellow" was not in the ↵tilghman1-1/+2
bitfield. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@274281 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-03Merged revisions 273793 via svnmerge from tilghman3-3/+12
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r273793 | tilghman | 2010-07-02 16:36:39 -0500 (Fri, 02 Jul 2010) | 9 lines Have the DEADLOCK_AVOIDANCE macro warn when an unlock fails, to help catch potentially large software bugs. (closes issue #17407) Reported by: pdf Patches: 20100527__issue17407.diff.txt uploaded by tilghman (license 14) Review: https://reviewboard.asterisk.org/r/751/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@273830 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-02Fix various typos reported by Lintiantzafrir6-24/+24
(Also fix the typos in the comments) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@273641 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-01correct handling of get_destination return valuesdvossel2-29/+55
A failure when calling the get_destination can mean multiple things. If the extension is not found, a 404 error is appropriate, but if the URI scheme is incorrect, a 404 is not approperiate. This patch adds the get_destination_result enum to differentiate between these and other failure types. The only logical difference in this patch is that we now send a "416 Unsupported URI scheme" response instead of a "404" when the scheme is not recognized. This indicates to the initiator of the INVITE to retry the request with a correct URI. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@273427 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-29Merged revisions 273060 via svnmerge from tilghman1-0/+2
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r273060 | tilghman | 2010-06-29 18:15:28 -0500 (Tue, 29 Jun 2010) | 10 lines Allow the "useragent" value to be restored into memory from the realtime backend. This value is purely informational. It does not alter configuration at all. (closes issue #16029) Reported by: Guggemand Patches: realtime-useragent.patch uploaded by Guggemand (license 897) Tested by: Guggemand ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@273078 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-29send a 400 Bad Request on malformed sip requestdvossel1-1/+7
RFC 2361 section 24.4.1 send a 400 Bad Request if the request can not be understood due to malformed syntax. Currently we simply ignore a packet with a missing callid, to, from, or via header. Instead of ignoring we now send the 400 Bad request. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@272981 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-28rfc compliant sip option parsing + new unit testdvossel4-113/+308
RFC 3261 section 8.2.2.3 states that if any unsupported options are found in the Require header field, a "420 (Bad Extension)" response should be sent with an Unsupported header field containing only the unsupported options. This is not currently being done correctly. Right now, if Asterisk detects any unsupported sip options in a Require header the entire list of options are returned in the Unsupported header even if some of those options are in fact supported. This patch fixes that by building an unsupported options character buffer when parsing the options that can be sent with the 420 response. A unit test verifying this functionality has been created. Some code refactoring was required. Review: https://reviewboard.asterisk.org/r/680/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@272880 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-28Merged revisions 272804 via svnmerge from mmichelson1-0/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r272804 | mmichelson | 2010-06-28 12:31:40 -0500 (Mon, 28 Jun 2010) | 5 lines Decode URI in contact header of 302 response. ABE-2352 ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@272805 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-28code guidelines cleanup for retrans_pkt() functiondvossel1-33/+42
I am doing work in this function. I noticed a large number of coding guidline fixes that needed to be made. Rather than have those changes distract from my functional changes I decided to separate these into a separate patch. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@272652 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-25chan_sip: more accurate retransmissionsdvossel1-0/+5
RFC3261 states that Timer A should start at 500ms (T1) by default. In chan_sip this value initially started at 1000ms and I changed it to 500ms recently. After doing that I noticed in my packet captures that it still occasionally retransmitted starting at 1000ms instead of 500ms like I told it to. This occurs because the scheduler runs in the do_monitor thread. If a new retransmission is added while the do_monitor thread is sleeping then it may not detect that retransmission for nearly 1000ms. To fix this I just poke the do_monitor thread to wake up when a new packet is sent reliably requiring retransmits. The thread then detects the new scheduler entry and adjusts its sleep time to account for it. Review: https://reviewboard.asterisk.org/r/747 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@272557 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-24Merged revisions 272446 via svnmerge from rmudgett1-0/+2
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r272446 | rmudgett | 2010-06-24 16:58:49 -0500 (Thu, 24 Jun 2010) | 10 lines ss_thread calls pri_grab without lock during overlap dial Recent changes to chan_dahdi with relation to overlap dialing call pri_grab without first obtaining a lock. (closes issue #17414) Reported by: pdf Patches: bug17414.patch uploaded by jpeeler (license 325) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@272447 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-23Resolve some errors produced during module unload of chan_iax2.russell1-41/+62
The external test suite stops Asterisk using the "core stop gracefully" command. The logs from the tests show that there are a number of problems with Asterisk trying to cleanly shut down. This patch addresses the following type of error that comes from chan_iax2: [Jun 22 16:58:11] ERROR[29884]: lock.c:129 __ast_pthread_mutex_destroy: chan_iax2.c line 11371 (iax2_process_thread_cleanup): Error destroying mutex &thread->lock: Device or resource busy For an example in the context of a build, see: http://bamboo.asterisk.org/browse/AST-TRUNK-739/log The primary purpose of this patch is to change the thread pool shutdown procedure to be more explicit to ensure that the thread exits from a point where it is not holding a lock. While testing that, I encountered various crashes due to the order of operations in unload_module() being problematic. I reordered some things there, as well. Review: https://reviewboard.asterisk.org/r/736/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@272370 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-23Add new AMI command LocalOptimizeAway.tringenbach1-0/+73
This command lets you request a "/n" local channel optimize itself out of the way anyway. Review: https://reviewboard.asterisk.org/r/732/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@272218 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-23D'oh! Defaultenabled FTL.tilghman1-2/+0
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@272150 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-23Load all lines from realtime, not just the first one.tilghman1-26/+35
(closes issue #17144) Reported by: nahuelgreco Patches: 20100513__issue17144__trunk.diff.txt uploaded by tilghman (license 14) Tested by: tilghman git-svn-id: http://svn.digium.com/svn/asterisk/trunk@272145 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-23Add extra protection for reinvite glare scenario.mmichelson1-1/+1
Testing proved that if Asterisk sent a connected line reinvite, and the endpoint to which the reinvite were being sent sent a reinvite, Asterisk would not properly respond with a 491 response. The reason is that on connected line reinvites, we set the dialog's invitestate to INV_CALLING to prevent Asterisk from sending a rapid flurry of connected line reinvites. For other reinvites we do not do this. Because of the current invitestate, when Asterisk received the reinvite, we interpreted this as a spiraled INVITE, and thus did not behave properly. The fix for this is to not enter the loop detection or spiral logic in handle_request_invite if the channel state is currently up. This way, no mid-call reinvites will be misinterpreted, no matter what the nature of the reinvite may have been. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@272090 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-22Don't try to lock/unlock an uninitialized lock on a dahdi_pri.russell1-0/+3
This small changes prevents destroy_all_channels() from accessing a lock on an unused dahdi_pri struct, resolving a ton of ERRORs that get spewed out when shutting Asterisk down gracefully. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@272052 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-22ignore CANCEL request after having already received final response to INVITEdvossel1-1/+8
RFC 3261 section 9 states that a CANCEL has no effect on a request to a UAS that has already given a final response. This patch checks to make sure there is a pending invite before allowing a CANCEL request to be processed, otherwise it responds to the CANCEL with a "481 Call/Transaction Does Not Exist". Review: https://reviewboard.asterisk.org/r/697/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@271977 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-22Merged revisions 271902 via svnmerge from mnicholson1-0/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r271902 | mnicholson | 2010-06-22 12:31:57 -0500 (Tue, 22 Jun 2010) | 8 lines Decrease the module ref count in sip_hangup when SIP_DEFER_BYE_ON_TRANSFER is set. This is necessary to keep the ref count correct. (closes issue #16815) Reported by: rain Patches: chan_sip-unref-fix.diff uploaded by rain (license 327) (modified) Tested by: rain ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@271903 f38db490-d61c-443f-a65b-d21fe96a405b
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