asterisk

Commit Graph

Author	SHA1	Message	Date
David Vossel	fc10fe712b	Merged revisions 215522 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r215522 \| dvossel \| 2009-09-02 12:26:40 -0500 (Wed, 02 Sep 2009) \| 11 lines SIP uri parsing cleanup Now, the scheme passed to parse_uri can either be a single scheme, or a list of schemes ',' delimited. This gets rid of the whole problem of having to create two buffers and calling parse_uri twice to check for separate schemes. Review: https://reviewboard.asterisk.org/r/343/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@215524 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Michiel van Baak	7286161ca0	Merged revisions 215462 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r215462 \| mvanbaak \| 2009-09-02 17:56:46 +0200 (Wed, 02 Sep 2009) \| 12 lines Honor configured parkinglot when parking and retrieving parked calls Thank oej for pointing out the fact that sip_new did not copy parkinglot from the peer into the newly created channel. (closes issue #15538) Reported by: gracedman Patches: 2009090100_sipnewparkinglot-161.diff.txt uploaded by mvanbaak (license 7) With mod by me to also fix callparking as well (this uploaded patch only fixed retrieving a parked call) Tested by: gracedman, mvanbaak ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@215464 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	4b93cae37f	Merged revisions 214199 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r214199 \| tilghman \| 2009-08-26 11:53:03 -0500 (Wed, 26 Aug 2009) \| 6 lines Typo fix ("SIP/2.0 XXX" is 11 chars, not 10) (closes issue #15362) Reported by: klaus3000 Patches: chan_sip.c_logmessagefix_patch.txt uploaded by klaus3000 (license 65) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@214201 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	8e4798e146	Merged revisions 213716 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r213716 \| dvossel \| 2009-08-21 17:22:11 -0500 (Fri, 21 Aug 2009) \| 10 lines Register request line contains wrong address when user domain and register host differ (closes issue #15539) Reported by: Nick_Lewis Patches: chan_sip.c-registraraddr.patch uploaded by Nick (license 657) register_domain_fix_1.6.2 uploaded by dvossel (license 671) Tested by: Nick_Lewis, dvossel ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@213724 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	8be21262e9	Merged revisions 213093 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r213093 \| tilghman \| 2009-08-19 15:29:41 -0500 (Wed, 19 Aug 2009) \| 7 lines If we have realtime caching enabled, 'sip reload' must purge users/peers, even if the config files haven't changed. (closes issue #12869) Reported by: bcnit Patches: 20090819__issue12869__2.diff.txt uploaded by tilghman (license 14) Tested by: lasko ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@213096 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	b6370b2383	Merged revisions 212113 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r212113 \| kpfleming \| 2009-08-13 10:46:25 -0500 (Thu, 13 Aug 2009) \| 3 lines Ensure that T38FaxVersion is put into outgoing SDP in the proper case. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@212115 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Joshua Colp	bedce86b42	Merged revisions 212067 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r212067 \| file \| 2009-08-13 10:51:04 -0300 (Thu, 13 Aug 2009) \| 2 lines Check an actual populated variable when seeing if we need to do video or not. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@212069 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	a26a8a80bd	Merged revisions 211876 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r211876 \| mnicholson \| 2009-08-12 14:53:14 -0500 (Wed, 12 Aug 2009) \| 11 lines Make asterisk handle 423 Interval Too Short messages better. This change uses separate values for the acceptable minimum expiry provided by the 423 error and the expiry value stored in the configuration file. Previously, the value pulled from the configuration file would be overwritten. (closes issue #14366) Reported by: Nick_Lewis Patches: sip-expiry-fix1.diff uploaded by mnicholson (license 96) chan_sip.c-reqexpiry.patch uploaded by Nick (license 657) Tested by: mnicholson ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@211950 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	07e59f290c	AST-2009-005 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@211569 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Joshua Colp	cba1b6e411	Merged revisions 211347 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r211347 \| file \| 2009-08-10 11:07:44 -0300 (Mon, 10 Aug 2009) \| 5 lines Fix retrieval of the port used for the video stream when adding SDP to a SIP message. (closes issue #15121) Reported by: jsmith ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@211349 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Joshua Colp	41af598912	Merged revisions 210817 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r210817 \| file \| 2009-08-06 14:47:04 -0300 (Thu, 06 Aug 2009) \| 11 lines Accept additional T.38 reinvites after an initial one has been handled. Discussion of this subject has yielded that it is not actually acceptable to change T.38 parameters after the initial reinvite but declining is harsh and can cause the fax to fail when it may be possible to allow it to continue. This patch changes things so that additional T.38 reinvites are accepted but parameter changes ignored. This gives the fax a fighting chance. (closes issue #15610) Reported by: huangtx2009 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@210819 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	2eea5e2553	Merged revisions 209516 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r209516 \| mmichelson \| 2009-07-30 09:38:21 -0500 (Thu, 30 Jul 2009) \| 8 lines Fix a crash that can result if text codecs are allowed but textsupport is disabled. (closes issue #15596) Reported by: fabled Patches: sip-red.patch uploaded by fabled (license 448) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@209517 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	623e055a28	Merged revisions 208588 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r208588 \| mmichelson \| 2009-07-24 13:31:04 -0500 (Fri, 24 Jul 2009) \| 16 lines Merged revisions 208587 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r208587 \| mmichelson \| 2009-07-24 13:26:50 -0500 (Fri, 24 Jul 2009) \| 10 lines Only send a BYE when hanging up a channel that is up. For cases where Asterisk sends an INVITE and receives a non 2XX final response, Asterisk would follow the INVITE transaction by immediately sending a BYE, which was unnecessary. (closes issue #14575) Reported by: chris-mac ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@208590 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	72c88bd434	Merged revisions 208548 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r208548 \| kpfleming \| 2009-07-24 10:02:53 -0500 (Fri, 24 Jul 2009) \| 8 lines Resolve a T.38 negotiation issue left over from the udptl-updates merge. The udptl-updates branch that was merged yesterday failed to properly send back T.38 SDP responses with the correct error correction mode, if the incoming SDP from the other end caused us to change error correction modes. This patch corrects that situation. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@208550 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	f4d55039dc	Merged revisions 208464 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r208464 \| kpfleming \| 2009-07-23 16:57:24 -0500 (Thu, 23 Jul 2009) \| 46 lines Rework of T.38 negotiation and UDPTL API to address interoperability problems Over the past couple of months, a number of issues with Asterisk negotiating (and successfully completing) T.38 sessions with various endpoints have been found. This patch attempts to address many of them, primarily focused around ensuring that the endpoints' MaxDatagram size is honored, and in addition by ensuring that T.38 session parameter negotiation is performed correctly according to the ITU T.38 Recommendation. The major changes here are: 1) T.38 applications in Asterisk (app_fax) only generate/receive IFP packets, they do not ever work with UDPTL packets. As a result of this, they cannot be allowed to generate packets that would overflow the other endpoints' MaxDatagram size after the UDPTL stack adds any error correction information. With this patch, the application is told the maximum IFP size it can generate, based on a calculation using the far end MaxDatagram size and the active error correction mode on the T.38 session. The same is true for sending our MaxDatagram size to the remote endpoint; it is computed from the value that the application says it can accept (for a single IFP packet) combined with the active error correction mode. 2) All treatment of T.38 session parameters as 'capabilities' in chan_sip has been removed; these parameters are not at all like audio/video stream capabilities. There are strict rules to follow for computing an answer to a T.38 offer, and chan_sip now follows those rules, using the desired parameters from the application (or channel) that wants to accept the T.38 negotiation. 3) chan_sip now stores and forwards ast_control_t38_parameters structures for tracking 'our' and 'their' T.38 session parameters; this greatly simplifies negotiation, especially for pass-through calls. 4) Since T.38 negotiation without specifying parameters or receiving the final negotiated parameters is not very worthwhile, the AST_CONTROL_T38 control frame has been removed. A note has been added to UPGRADE.txt about this removal, since any out-of-tree applications that use it will no longer function properly until they are upgraded to use AST_CONTROL_T38_PARAMETERS. Review: https://reviewboard.asterisk.org/r/310/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@208484 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	4642b45802	Merged revisions 208388 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r208388 \| mmichelson \| 2009-07-23 14:34:49 -0500 (Thu, 23 Jul 2009) \| 24 lines Merged revisions 208386 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r208386 \| mmichelson \| 2009-07-23 14:24:21 -0500 (Thu, 23 Jul 2009) \| 17 lines Fix a problem where a 491 response could be sent out of dialog. This generalizes the fix for issue 13849. The initial fix corrected the problem that Asterisk would reply with a 491 if a reinvite were received from an endpoint and we had not yet received an ACK from that endpoint for the initial INVITE it had sent us. This expansion also allows Asterisk to appropriately handle an INVITE with authorization credentials if Asterisk had not received an ACK from the previous transaction in which Asterisk had responded to an unauthorized INVITE with a 407. (closes issue #14239) Reported by: klaus3000 Patches: 14239.patch uploaded by mmichelson (license 60) Tested by: klaus3000 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@208390 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	c8982d075e	Merged revisions 208314 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r208314 \| mmichelson \| 2009-07-23 11:29:37 -0500 (Thu, 23 Jul 2009) \| 9 lines Merged revisions 208312 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r208312 \| mmichelson \| 2009-07-23 11:29:18 -0500 (Thu, 23 Jul 2009) \| 3 lines Remove inaccurate XXX comment. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@208318 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	b2cd6bc4f3	Merged revisions 208263 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r208263 \| mmichelson \| 2009-07-23 10:46:34 -0500 (Thu, 23 Jul 2009) \| 15 lines Merged revisions 208262 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r208262 \| mmichelson \| 2009-07-23 10:43:07 -0500 (Thu, 23 Jul 2009) \| 8 lines Properly handle 183 responses which do not contain an SDP. (closes issue #15442) Reported by: ffloimair Patches: 15442.patch uploaded by mmichelson (license 60) Tested by: tkarl, ffloimair ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@208265 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	52bfea4da6	Merged revisions 207424 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r207424 \| mmichelson \| 2009-07-20 14:48:12 -0500 (Mon, 20 Jul 2009) \| 39 lines Merged revisions 207423 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r207423 \| mmichelson \| 2009-07-20 14:39:59 -0500 (Mon, 20 Jul 2009) \| 33 lines Answer video SDP offers properly when videosupport is not enabled. Copied from Review board: In issue 12434, the reporter describes a situation in which audio and video is offered on the call, but because videosupport is disabled in sip.conf, Asterisk gives no response at all to the video offer. According to RFC 3264, all media offers should have a corresponding answer. For offers we do not intend to actually reply to with meaningful values, we should still reply with the port for the media stream set to 0. In this patch, we take note of what types of media have been offered and save the information on the sip_pvt. The SDP in the response will take into account whether media was offered. If we are not otherwise going to answer a media offer, we will insert an appropriate m= line with the port set to 0. It is important to note that this patch is pretty much a bandage being applied to a broken bone. The patch only helps for situations where video is offered but videosupport is disabled and when udptl_pt is disabled but T.38 is offered. Asterisk is not guaranteed to respond to every media offer. Notable cases are when multiple streams of the same type are offered. The 2 media stream limit is still present with this patch, too. In trunk and the 1.6.X branches, things will be a bit different since Asterisk also supports text in SDPs as well. (closes issue #12434) Reported by: mnnojd Review: https://reviewboard.asterisk.org/r/311 Review: https://reviewboard.asterisk.org/r/313 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@207426 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	f1fdcb317f	Merged revisions 207029 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r207029 \| dvossel \| 2009-07-17 12:51:44 -0500 (Fri, 17 Jul 2009) \| 6 lines sip option flags handled incorrectly (closes issue #15376) Reported by: Takehiko Ooshima Tested by: dvossel, Takehiko_Ooshima ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@207031 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	88dc0e47d7	Merged revisions 206939 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r206939 \| dvossel \| 2009-07-17 11:13:22 -0500 (Fri, 17 Jul 2009) \| 20 lines Merged revisions 206938 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r206938 \| dvossel \| 2009-07-17 11:05:06 -0500 (Fri, 17 Jul 2009) \| 14 lines SIP incorrect From: header information when callpres is prohib Some ITSP make use of the "Anonymous" display name to detect a requirement to withhold caller id across the PSTN. This does not work if the display name is "Unknown". (closes issue #14465) Reported by: Nick_Lewis Patches: chan_sip.c-callerpres.patch uploaded by Nick (license 657) chan_sip.c-callerpres_trunk.patch uploaded by dvossel (license 671) Tested by: Nick_Lewis, dvossel ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@206941 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	19b741deb0	Merged revisions 206768 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r206768 \| dvossel \| 2009-07-15 17:04:13 -0500 (Wed, 15 Jul 2009) \| 8 lines Session timer were not activated if Supported header field in INVITE had both "timer" and other options. (closes issue #15403) Reported by: makoto Patches: sip-session-timer.patch uploaded by makoto (license ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@206774 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	44fa844576	Merged revisions 206702 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r206702 \| dvossel \| 2009-07-15 15:20:01 -0500 (Wed, 15 Jul 2009) \| 10 lines callerid(num) is wrong when username is missing A domain only sip uri <sip:123.123.123.123> would return 123.123.123.123 as callid num. Now, if the username is missing from a uri, the callerid num field is left empty. (closes issue #15476) Reported by: viraptor ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@206704 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	6de099e16c	Merged revisions 206280 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r206280 \| dvossel \| 2009-07-13 18:26:51 -0500 (Mon, 13 Jul 2009) \| 9 lines dns lookup of peername rather than peer's host in transmit_register() (closes issue #15052) Reported by: fsantulli Patches: chan_sip_bug_15052_[20090626204511].patch uploaded by fsantulli (license 818) Tested by: fsantulli ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@206282 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	31728d23ea	Merged revisions 205985 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r205985 \| dvossel \| 2009-07-10 16:42:10 -0500 (Fri, 10 Jul 2009) \| 16 lines SIP register not using peer's outbound proxy If callbackextension is defined for a peer it successfully causes a registration to occur, but the registration ignores the outboundproxy settings for the peer. This patch allows the peer to be passed to obproxy_get() in transmit_register(). (closes issue #14344) Reported by: Nick_Lewis Patches: callbackextension_peer_trunk.diff uploaded by dvossel (license 671) Tested by: dvossel Review: https://reviewboard.asterisk.org/r/294/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@205987 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	74b383157e	Merged revisions 205878 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r205878 \| mmichelson \| 2009-07-10 12:39:57 -0500 (Fri, 10 Jul 2009) \| 30 lines Merged revisions 205877 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r205877 \| mmichelson \| 2009-07-10 12:39:13 -0500 (Fri, 10 Jul 2009) \| 23 lines Merged revisions 205776 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r205776 \| mmichelson \| 2009-07-10 10:56:45 -0500 (Fri, 10 Jul 2009) \| 16 lines Merged revisions 205775 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205775 \| mmichelson \| 2009-07-10 10:51:36 -0500 (Fri, 10 Jul 2009) \| 10 lines Ensure that outbound NOTIFY requests are properly routed through stateful proxies. With this change, we make note of Record-Route headers present in any SUBSCRIBE request that we receive so that our outbound NOTIFY requests will have the proper Route headers in them. (closes issue #14725) Reported by: ibc ........ ................ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@205881 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	f3b9afe34d	Merged revisions 205840 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r205840 \| dvossel \| 2009-07-10 11:42:04 -0500 (Fri, 10 Jul 2009) \| 37 lines Merged revisions 205804 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205804 \| dvossel \| 2009-07-10 11:23:59 -0500 (Fri, 10 Jul 2009) \| 31 lines SIP registration auth loop caused by stale nonce If an endpoint sends two registration requests in a very short period of time with the same nonce, both receive 401 responses from Asterisk, each with a different nonce (the second 401 containing the current nonce and the first one being stale). If the endpoint responds to the first 401, it does not match the current nonce so Asterisk sends a third 401 with a newly generated nonce (which updates the current nonce)... Now if the endpoint responds to the second 401, it does not match the current nonce either and Asterisk sends a fourth 401 with a newly generated nonce... This loop goes on and on. There appears to be a simple fix for this. If the nonce from the request does not match our nonce, but is a good response to a previous nonce, instead of sending a 401 with a newly generated nonce, use the current one instead. This breaks the loop as the nonce is not updated until a response is received. Additional logic has been added to make sure no nonce can be responded to twice though. (closes issue #15102) Reported by: Jamuel Patches: patch-bug_0015102 uploaded by Jamuel (license 809) nonce_sip.diff uploaded by dvossel (license 671) Tested by: Jamuel Review: https://reviewboard.asterisk.org/r/289/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@205842 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	2e6570186a	Merged revisions 205776 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r205776 \| mmichelson \| 2009-07-10 10:56:45 -0500 (Fri, 10 Jul 2009) \| 16 lines Merged revisions 205775 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205775 \| mmichelson \| 2009-07-10 10:51:36 -0500 (Fri, 10 Jul 2009) \| 10 lines Ensure that outbound NOTIFY requests are properly routed through stateful proxies. With this change, we make note of Record-Route headers present in any SUBSCRIBE request that we receive so that our outbound NOTIFY requests will have the proper Route headers in them. (closes issue #14725) Reported by: ibc ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@205778 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	746eb38a12	Merged revisions 205696 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r205696 \| kpfleming \| 2009-07-09 16:20:23 -0500 (Thu, 09 Jul 2009) \| 16 lines Repair ability of SendFAX/ReceiveFAX to respond to T.38 switchover. Recent changes in T.38 negotiation in Asterisk caused these applications to not respond when the other endpoint initiated a switchover to T.38; this resulted in the T.38 switchover failing, and the FAX attempt to be made using an audio connection, instead of T.38 (which would usually cause the FAX to fail completely). This patch corrects this problem, and the applications will now correctly respond to the T.38 switchover request. In addition, the response will include the appopriate T.38 session parameters based on what the other end offered and what our end is capable of. (closes issue #14849) Reported by: afosorio ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@205698 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	17f8c7a354	Merged revisions 204301 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r204301 \| mmichelson \| 2009-06-29 17:50:35 -0500 (Mon, 29 Jun 2009) \| 15 lines Merged revisions 204300 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r204300 \| mmichelson \| 2009-06-29 17:45:34 -0500 (Mon, 29 Jun 2009) \| 9 lines Add error message so that it is clear why a SIP peer was not processed when a DNS lookup fails on a host or outboundproxy. (closes issue #13432) Reported by: p_lindheimer Patches: outboundproxy.patch uploaded by p (license 558) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@204303 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	e5706ee847	Merged revisions 204247 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r204247 \| mmichelson \| 2009-06-29 16:48:54 -0500 (Mon, 29 Jun 2009) \| 32 lines Merged revisions 204243,204246 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r204243 \| mmichelson \| 2009-06-29 16:23:43 -0500 (Mon, 29 Jun 2009) \| 22 lines Fix a problem where chan_sip would ignore "old" but valid responses. chan_sip has had a problem for quite a long time that would manifest when Asterisk would send multiple SIP responses on the same dialog before receiving a response. The problem occurred because chan_sip only kept track of the highest outgoing sequence number used on the dialog. If Asterisk sent two requests out, and a response arrived for the first request sent, then Asterisk would ignore the response. The result was that Asterisk would continue retransmitting the requests and ignoring the responses until the maximum number of retransmissions had been reached. The fix here is to rearrange the code a bit so that instead of simply comparing the sequence number of the response to our latest outgoing sequence number, we walk our list of outstanding packets and determine if there is a match. If there is, we continue. If not, then we ignore the response. In doing this, I found a few completely useless variables that I have now removed. (closes issue #11231) Reported by: flefoll Review: https://reviewboard.asterisk.org/r/298 ........ r204246 \| mmichelson \| 2009-06-29 16:37:05 -0500 (Mon, 29 Jun 2009) \| 3 lines Fix build oops. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@204249 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	41c332513f	Merged revisions 203779 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r203779 \| russell \| 2009-06-26 15:45:00 -0500 (Fri, 26 Jun 2009) \| 5 lines Ensure the TCP read buffer is fully initialized before handling each packet. (closes issue #14452) Reported by: umberto71 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@203781 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Joshua Colp	642b571683	Merged revisions 203699 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r203699 \| file \| 2009-06-26 16:27:24 -0300 (Fri, 26 Jun 2009) \| 2 lines Improve T.38 negotiation by exchanging session parameters between application and channel. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@203703 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	8ac0deae26	Merged revisions 203116 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r203116 \| russell \| 2009-06-25 11:04:10 -0500 (Thu, 25 Jun 2009) \| 18 lines Merged revisions 203115 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r203115 \| russell \| 2009-06-25 11:02:16 -0500 (Thu, 25 Jun 2009) \| 11 lines Resolve a crash related to a T.38 reinvite race condition. This change resolves a crash observed locally during some T.38 testing. A call was set up using a call file, and when the T.38 reinvite came in, the channel state was still AST_STATE_DOWN. The reason is explained by a comment in the code that previously lived in the handling of AST_STATE_RINGING. This change modifies the logic to handle the same race condition for any channel state that is not UP. (closes ABE-1895) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@203118 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	9d35f9503b	Merged revisions 202967 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r202967 \| mmichelson \| 2009-06-24 13:29:10 -0500 (Wed, 24 Jun 2009) \| 9 lines Merged revisions 202966 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r202966 \| mmichelson \| 2009-06-24 13:28:47 -0500 (Wed, 24 Jun 2009) \| 3 lines Use the handy UNLINK macro instead of hand-coding the same thing in-line. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@202969 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Joshua Colp	10d49a7cc8	Merged revisions 202925 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r202925 \| file \| 2009-06-24 15:08:17 -0300 (Wed, 24 Jun 2009) \| 2 lines Ensure the default settings are applied for T.38 when we set it up for a peer. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@202927 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	f2441e1d3d	Merged revisions 202672 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r202672 \| dvossel \| 2009-06-23 11:31:30 -0500 (Tue, 23 Jun 2009) \| 18 lines Merged revisions 202671 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r202671 \| dvossel \| 2009-06-23 11:28:46 -0500 (Tue, 23 Jun 2009) \| 12 lines MWI NOTIFY contains a wrong URI if Asterisk listens to non-standard port and transport (closes issue #14659) Reported by: klaus3000 Patches: patch_chan_sip_fixMWIuri_1.4.txt uploaded by klaus3000 (license 65) mwi_port-transport_trunk.diff uploaded by dvossel (license 671) Tested by: dvossel, klaus3000 Review: https://reviewboard.asterisk.org/r/288/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@202674 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	9bce657f84	Merged revisions 202415 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r202415 \| russell \| 2009-06-22 11:05:08 -0500 (Mon, 22 Jun 2009) \| 9 lines Merged revisions 202414 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r202414 \| russell \| 2009-06-22 11:00:00 -0500 (Mon, 22 Jun 2009) \| 2 lines Make Polycom subscription type override check more explicit. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@202418 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	1606795a78	Merged revisions 202343 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r202343 \| mmichelson \| 2009-06-22 09:58:24 -0500 (Mon, 22 Jun 2009) \| 36 lines Merged revisions 202341-202342 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r202341 \| mmichelson \| 2009-06-22 09:42:55 -0500 (Mon, 22 Jun 2009) \| 26 lines Fix a situation in which Asterisk would not stop retransmitting 487s. If a CANCEL were received by Asterisk, we would send a 487 in response to the original INVITE and a 200 OK for the CANCEL. If there were a network hiccup which caused the 200 OK and the 487 to be lost, then the UA communicating with Asterisk may try to retransmit its CANCEL. Asterisk's response to this used to be to try sending another 487 to the canceled INVITE and another 200 OK to the CANCEL. The problem here is that the originally-sent 487 was sent "reliably" meaning that it will be retransmitted until it is received properly. So when we receive the second CANCEL it is likely that the first batch of 487s we sent is still going strong and reaches the UA. The result was that the second set of 487s would be retransmitted constantly until the maximum number of retries had been reached. The fix for this is that if we receive a second CANCEL for an INVITE, then we cancel the retransmission of the first set of 487s and start a second set. This causes the dialog to be terminated reasonably. (closes issue #14584) Reported by: klaus3000 Patches: 14584_v2.patch uploaded by mmichelson (license 60) Tested by: klaus3000 ........ r202342 \| mmichelson \| 2009-06-22 09:44:58 -0500 (Mon, 22 Jun 2009) \| 3 lines Remove an extra debug line left from previous commit. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@202345 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	ee91773ea8	Merged revisions 202337 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r202337 \| mmichelson \| 2009-06-22 09:35:09 -0500 (Mon, 22 Jun 2009) \| 31 lines Merged revisions 202336 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r202336 \| mmichelson \| 2009-06-22 09:34:05 -0500 (Mon, 22 Jun 2009) \| 25 lines Fix a possible infinite loop in SDP parsing during glare situation. There was a while loop in get_ip_and_port_from_sdp which was controlled by a call to get_sdp_iterate. The loop would exit either if what we were searching for was found or if the return was NULL. The problem is that get_sdp_iterate never returns NULL. This means that if what we were searching for was not present, the loop would run infinitely. This modification of the loop fixes the problem. (closes issue #15213) Reported by: schmidts (closes issue #15349) Reported by: samy (closes issue #14464) Reported by: pj (closes issue #15345) Reported by: aragon Patches: sip_inf_loop.patch uploaded by mmichelson (license 60) Tested by: aragon ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@202339 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	e8a03ddfdd	Added deadlock protection to try_suggested_sip_codec in chan_sip.c. Review: https://reviewboard.asterisk.org/r/287/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@202008 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	c2e5311110	Merged revisions 201570 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r201570 \| dvossel \| 2009-06-18 10:16:05 -0500 (Thu, 18 Jun 2009) \| 11 lines parsing extension correctly from sip register lines If a transport type was specified, but no extension, parsing of the extension would return whatever was after the transport rather than defaulting to 's'. (closes issue #15111) Reported by: ffs Patches: chan_sip.c_register-parser.patch uploaded by ffs (license 730) Tested by: ffs, dvossel ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@201601 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	0a92ebc9bd	Merged revisions 201462 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r201462 \| mmichelson \| 2009-06-17 15:10:01 -0500 (Wed, 17 Jun 2009) \| 12 lines Fix problem with no audio due to ignoring the SDP. A recent change to our SDP version comparison made audio not function on some calls. This was because of a test wherein we were trying to see if an unsigned value was less than 0. This is a dumb comparison and arguably the compiler should have warned about it. Alas, though, it slipped past. Now it's fixed by changing the variable to be a signed type. Found by several developers. Tested by mnicholson and dbrooks. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@201464 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Brooks	c33eb64920	Merged revisions 201381 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r201381 \| dbrooks \| 2009-06-17 14:15:07 -0500 (Wed, 17 Jun 2009) \| 16 lines Merged revisions 201380 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r201380 \| dbrooks \| 2009-06-17 13:45:50 -0500 (Wed, 17 Jun 2009) \| 9 lines Checks for NULL sip_pvt pointer in chan_sip.c->acf_channel_read() Zombie channels could be passed, and chan_sip.c wasn't checking for it. Could crash Asterisk. Now checking for NULL pointer. (closes issue #15330) Reported by: okrief Tested by: dbrooks ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@201444 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	a3d2d156ee	Merged revisions 201344 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r201344 \| dvossel \| 2009-06-17 10:20:26 -0500 (Wed, 17 Jun 2009) \| 16 lines SIP registry ref count error During a sip reload, the list of sip_registry objects are supposed to be traversed, unlinked, and destroyed, but destruction never takes place due to a ref counting error. This causes a memory leak when registry items are removed from sip.conf and reloaded. While the registries are removed from the global list, they are not removed from the scheduler. Because of this, SIP register attempts continue to be sent out for the item even though it may no longer be in the .conf. (closes issue #15295) Reported by: amorsen Review: https://reviewboard.asterisk.org/r/282/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@201365 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	f5fca5c8e1	Merged revisions 201223 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r201223 \| dvossel \| 2009-06-16 17:29:30 -0500 (Tue, 16 Jun 2009) \| 2 lines fix issue with build_contact introduced by the "SIP trasnport type issues" commit ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@201225 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	8e5e00bd07	Merged revisions 200946 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r200946 \| dvossel \| 2009-06-16 11:03:30 -0500 (Tue, 16 Jun 2009) \| 32 lines SIP transport type issues What this patch addresses: 1. ast_sip_ouraddrfor() by default binds to the UDP address/port reguardless if the sip->pvt is of type UDP or not. Now when no remapping is required, ast_sip_ouraddrfor() checks the sip_pvt's transport type, attempting to set the address and port to the correct TCP/TLS bindings if necessary. 2. It is not necessary to send the port number in the Contact header unless the port is non-standard for the transport type. This patch fixes this and removes the todo note. 3. In sip_alloc(), the default dialog built always uses transport type UDP. Now sip_alloc() looks at the sip_request (if present) and determines what transport type to use by default. 4. When changing the transport type of a sip_socket, the file descriptor must be set to -1 and in some cases the tcptls_session's ref count must be decremented and set to NULL. I've encountered several issues associated with this process and have created a function, set_socket_transport(), to handle the setting of the socket type. (closes issue #13865) Reported by: st Patches: dont_add_port_if_tls.patch uploaded by Kristijan (license 753) 13865.patch uploaded by mmichelson (license 60) tls_port_v5.patch uploaded by vrban (license 756) transport_issues.diff uploaded by dvossel (license 671) Tested by: mmichelson, Kristijan, vrban, jmacz, dvossel Review: https://reviewboard.asterisk.org/r/278/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@200987 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	7375533824	Merged revisions 165180,200689 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r165180 \| mnicholson \| 2008-12-17 12:49:12 -0600 (Wed, 17 Dec 2008) \| 14 lines This patch adds a new 'ignoresdpversion' option to sip.conf. When this is enabled (either globally or for a specific peer), chan_sip will treat any SDP data it receives as new data and update the media stream accordingly. By default, Asterisk will only modify the media stream if the SDP session version received is different from the current SDP session version. This option is required to interoperate with devices that have non-standard SDP session version implementations (observed by toc on the bug tracker with Microsoft OCS which always uses 0 as the session version). http://reviewboard.digium.com/r/94/ (closes issue #13958) Reported by: toc Tested by: toc ........ r200689 \| kpfleming \| 2009-06-15 15:42:38 -0500 (Mon, 15 Jun 2009) \| 12 lines Accept T.38 re-INVITE responses with invalid SDP versions. This commit changes the 'incoming SDP version' check logic a bit more; when 'ignoresdpversion' is not set for a peer, if we initiate a re-INVITE to switch to T.38, we'll always accept the peer's SDP response, even if they don't properly increment the SDP version number as they should. If this situation occurs, a warning message will be generated suggesting that the peer's configuration be changed to include the 'ignoresdpversion' configuration option (although ideally they'd fix their SIP implementation to be RFC compliant). AST-221 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@200707 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	369810c36c	Merged revisions 200514 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r200514 \| mmichelson \| 2009-06-15 10:22:11 -0500 (Mon, 15 Jun 2009) \| 11 lines Merged revisions 200513 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r200513 \| mmichelson \| 2009-06-15 10:21:46 -0500 (Mon, 15 Jun 2009) \| 5 lines Add INFO to our allowed methods so that endpoints know they may send it to us. AST-223 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@200516 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	cb76dba60a	Merged revisions 200146 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r200146 \| mmichelson \| 2009-06-11 16:17:14 -0500 (Thu, 11 Jun 2009) \| 5 lines Fix a crash due to a potentially NULL p->options. Thanks to mnicholson for pointing it out. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@200152 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago

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