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2009-10-06Merged revisions 222176 via svnmerge from kpfleming2-1/+20
https://origsvn.digium.com/svn/asterisk/trunk ................ r222176 | kpfleming | 2009-10-05 20:24:24 -0500 (Mon, 05 Oct 2009) | 27 lines Recorded merge of revisions 222152 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r222152 | kpfleming | 2009-10-05 20:16:36 -0500 (Mon, 05 Oct 2009) | 20 lines Fix ao2_iterator API to hold references to containers being iterated. See Mantis issue for details of what prompted this change. Additional notes: This patch changes the ao2_iterator API in two ways: F_AO2I_DONTLOCK has become an enum instead of a macro, with a name that fits our naming policy; also, it is now necessary to call ao2_iterator_destroy() on any iterator that has been created. Currently this only releases the reference to the container being iterated, but in the future this could also release other resources used by the iterator, if the iterator implementation changes to use additional resources. (closes issue #15987) Reported by: kpfleming Review: https://reviewboard.asterisk.org/r/383/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@222185 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-05Recorded merge of revisions 222110 via svnmerge from kpfleming1-25/+96
https://origsvn.digium.com/svn/asterisk/trunk ........ r222110 | kpfleming | 2009-10-05 14:45:00 -0500 (Mon, 05 Oct 2009) | 25 lines Allow non-compliant T.38 endpoints to be supportable via configuration option. Many T.38 endpoints incorrectly send the maximum IFP frame size they can accept as the T38FaxMaxDatagram value in their SDP, when in fact this value is supposed to be the maximum UDPTL payload size (datagram size) they can accept. If the value they supply is small enough (a commonly supplied value is '72'), T.38 UDPTL transmissions will likely fail completely because the UDPTL packets will not have enough room for a primary IFP frame and the redundancy used for error correction. If this occurs, the Asterisk UDPTL stack will emit log messages warning that data loss may occur, and that the value may need to be overridden. This patch extends the 't38pt_udptl' configuration option in sip.conf to allow the administrator to override the value supplied by the remote endpoint and supply a value that allows T.38 FAX transmissions to be successful with that endpoint. In addition, in any SIP call where the override takes effect, a debug message will be printed to that effect. This patch also removes the T38FaxMaxDatagram configuration option from udptl.conf.sample, since it has not actually had any effect for a number of releases. In addition, this patch cleans up the T.38 documentation in sip.conf.sample (which incorrectly documented that T.38 support was passthrough only). (issue #15586) Reported by: globalnetinc ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@222111 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-02Merged revisions 222030 via svnmerge from dvossel1-2/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r222030 | dvossel | 2009-10-02 12:34:07 -0500 (Fri, 02 Oct 2009) | 9 lines Merged revisions 222026 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r222026 | dvossel | 2009-10-02 12:32:13 -0500 (Fri, 02 Oct 2009) | 3 lines Removes unnecessary unlock, clarifies a memcpy. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@222038 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-02Merged revisions 221844 via svnmerge from rmudgett2-41/+66
https://origsvn.digium.com/svn/asterisk/trunk ................ r221844 | rmudgett | 2009-10-01 20:09:31 -0500 (Thu, 01 Oct 2009) | 33 lines Merged revisions 221769 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r221769 | rmudgett | 2009-10-01 18:18:28 -0500 (Thu, 01 Oct 2009) | 26 lines Occasionally losing use of B channels in chan_misdn. I have not been able to reproduce the problem of losing channels. However, I have seen in the code a reentrancy problem that might give these symptoms. The reentrancy patch does several things: 1) Guards B channel and B channel structure allocation. 2) Makes the B channel structure find routines more precise in locating records. 3) Never leave a B channel allocated if we received cause 44. The last item may cause temporary outgoing call problems, but they should clear when the line becomes idle. (closes issue #15490) Reported by: slutec18 Patches: issue15490_channel_alloc_reentrancy.patch uploaded by rmudgett (license 664) Tested by: rmudgett, slutec18 (closes issue #15458) Reported by: FabienToune Patches: issue15458_channel_alloc_reentrancy.patch uploaded by rmudgett (license 664) Tested by: FabienToune, rmudgett, slutec18 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@221853 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-01Merged revisions 221697 via svnmerge from dvossel1-9/+22
https://origsvn.digium.com/svn/asterisk/trunk ........ r221697 | dvossel | 2009-10-01 14:33:33 -0500 (Thu, 01 Oct 2009) | 9 lines outbound tls connections were not defaulting to port 5061 (closes issue #15854) Reported by: dvossel Patches: sip_port_config_trunk.diff uploaded by dvossel (license 671) Tested by: dvossel ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@221745 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-01Merged revisions 221705 via svnmerge from tilghman1-88/+86
https://origsvn.digium.com/svn/asterisk/trunk ........ r221705 | tilghman | 2009-10-01 15:09:46 -0500 (Thu, 01 Oct 2009) | 2 lines Revision 220906 (a merge from 1.4) was not merged correctly, causing a problem with non-dynamic peers. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@221742 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-01Fixes issue with non dynamic hosts not being set for peersdvossel1-12/+12
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@221712 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-01Merged revisions 221554,221589 via svnmerge from mnicholson1-6/+5
https://origsvn.digium.com/svn/asterisk/trunk ................ r221554 | oej | 2009-10-01 02:00:04 -0500 (Thu, 01 Oct 2009) | 3 lines Simplify code for porturi, use TRUE/FALSE constructs when it's just TRUE or FALSE. ................ r221589 | mnicholson | 2009-10-01 10:26:20 -0500 (Thu, 01 Oct 2009) | 9 lines Merged revisions 221588 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r221588 | mnicholson | 2009-10-01 10:24:00 -0500 (Thu, 01 Oct 2009) | 2 lines Use unsigned ints for portinuri flags. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@221662 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-30Merged revisions 221432 via svnmerge from mnicholson1-3/+20
https://origsvn.digium.com/svn/asterisk/trunk ................ r221432 | mnicholson | 2009-09-30 15:40:20 -0500 (Wed, 30 Sep 2009) | 17 lines Merged revisions 221360 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r221360 | mnicholson | 2009-09-30 14:36:06 -0500 (Wed, 30 Sep 2009) | 10 lines Fix SRV lookup and Request-URI generation in chan_sip. This patch adds a new field "portinuri" to the sip dialog struct and the sip peer struct. That field is used during RURI generation to determine if the port should be included in the RURI. It is also used in some places to determine if an SRV lookup should occur. (closes issue #14418) Reported by: klaus3000 Tested by: klaus3000, mnicholson Review: https://reviewboard.asterisk.org/r/369/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@221486 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-30Merged revisions 221266 via svnmerge from twilson1-1/+23
https://origsvn.digium.com/svn/asterisk/trunk ................ r221266 | twilson | 2009-09-30 12:52:30 -0500 (Wed, 30 Sep 2009) | 32 lines Merged revisions 221086 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r221086 | twilson | 2009-09-30 09:49:11 -0500 (Wed, 30 Sep 2009) | 25 lines Change the SSRC by default when our media stream changes Be default, change SSRC when doing an audio stream changes Asterisk doesn't honor marker bit when reinvited to already-bridged RTP streams,resulting in far-end stack discarding packets with "old" timestamps that areactually part of a new stream. This patch sends AST_CONTROL_SRCUPDATE whenever there is a reinvite, unless the 'constantssrc' is set to true in sip.conf. The original issue reported to Digium support detailed the following situation: ITSP <-> Asterisk 1.4.26.2 <-> SIP-based Application Server Call comes in fromITSP, Asterisk dials the app server which sends a re-invite back toAsterisk--not to negotiate to send media directly to the ITSP, but to indicatethat it's changing the stream it's sending to Asterisk. The app servergenerates a new SSRC, sequence numbers, timestamps, and sets the marker bit on the new stream. Asterisk passes through the teimstamp of the new stream, butdoes not reset the SSRC, sequence numbers, or set the marker bit. When the timestamp on the new stream is older than the timestamp on the originalstream, the ITSP (which doesn't know there has been any change) discards the newframes because it thinks they are too old. This patch addresses this by changing the SSRC on a stream update unless constantssrc=true is set in sip.conf. Review: https://reviewboard.asterisk.org/r/374/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@221301 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-29Merged revisions 220906 via svnmerge from tilghman1-347/+436
https://origsvn.digium.com/svn/asterisk/trunk ................ r220906 | tilghman | 2009-09-29 14:57:37 -0500 (Tue, 29 Sep 2009) | 16 lines Merged revisions 220873 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r220873 | tilghman | 2009-09-29 12:59:26 -0500 (Tue, 29 Sep 2009) | 9 lines Reduce CPU usage related to building a peer merely for devicestates. This fixes a 100% CPU problem in the SIP driver, found by profiling the driver while the problem was occurring. (closes issue #14309) Reported by: pkempgen Patches: 20090924__issue14309.diff.txt uploaded by tilghman (license 14) Tested by: pkempgen, vrban ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@220976 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-21Merged revisions 219721 via svnmerge from dvossel1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r219721 | dvossel | 2009-09-21 11:59:05 -0500 (Mon, 21 Sep 2009) | 9 lines Merged revisions 219720 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r219720 | dvossel | 2009-09-21 11:55:53 -0500 (Mon, 21 Sep 2009) | 3 lines Reverting merge 219520. This change was not necessary. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@219724 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-19Merged revisions 219587 via svnmerge from russell1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r219587 | russell | 2009-09-18 21:59:52 -0500 (Fri, 18 Sep 2009) | 13 lines Merged revisions 219586 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r219586 | russell | 2009-09-18 21:51:13 -0500 (Fri, 18 Sep 2009) | 6 lines Make sure the iax_pvt exists before dereferencing it. This fixes the latest crash posted on issue 15609. (issue #15609) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@219588 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-18Merged revisions 219520 via svnmerge from dvossel1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r219520 | dvossel | 2009-09-18 18:20:58 -0500 (Fri, 18 Sep 2009) | 15 lines Merged revisions 219519 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r219519 | dvossel | 2009-09-18 18:19:50 -0500 (Fri, 18 Sep 2009) | 9 lines iax2 frame double free The iax frame's retrans sched id was written over right before iax2_frame_free was called. In iax2_frame_free that retrans id is used to delete the sched item. By writing over the retrans field before the sched item could be deleted, it was possible for a retransmit to occur on a freed frame. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@219523 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-18Merged revisions 219451 via svnmerge from dvossel1-5/+6
https://origsvn.digium.com/svn/asterisk/trunk ................ r219451 | dvossel | 2009-09-18 11:20:41 -0500 (Fri, 18 Sep 2009) | 20 lines Merged revisions 219450 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r219450 | dvossel | 2009-09-18 11:19:15 -0500 (Fri, 18 Sep 2009) | 14 lines via-header branches not updated correctly on INVITE INVITE requests must always contain a new unique branch id. When a new branch id is created for an INVITE, the dialog's invite_branch variable must be updated so CANCEL requests use the correct branch id. (closes issue #15262) Reported by: maniax Patches: asterisk-1.6.1.0-sip-branch.patch uploaded by tweety (license 608) invite_new_branch_trunk.diff uploaded by dvossel (license 671) Tested by: maniax, dvossel ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@219454 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-17Merged revisions 219324 via svnmerge from file1-0/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r219324 | mmichelson | 2009-09-17 17:22:01 -0500 (Thu, 17 Sep 2009) | 12 lines Merged revisions 219320 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r219320 | mmichelson | 2009-09-17 17:20:50 -0500 (Thu, 17 Sep 2009) | 6 lines Send a 100 Trying response when we detect a spiral. This was problematic during spiral tests at SIPit... along with some other things as well. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@219365 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-17Merged revisions 219304 via svnmerge from dvossel1-40/+22
https://origsvn.digium.com/svn/asterisk/trunk ................ r219304 | dvossel | 2009-09-17 16:59:21 -0500 (Thu, 17 Sep 2009) | 27 lines Merged revisions 219303 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r219303 | dvossel | 2009-09-17 16:29:37 -0500 (Thu, 17 Sep 2009) | 21 lines INVITE w/Replaces deadlock fix This patch cleans up the locking logic in chan_sip.c's handle_invite_replaces() function as well as making use of ast_do_masquerade() rather than forcing the masquerade on an ast_read(). The code had several redundant unlocks that would result in 'freed more times than we've locked!' errors. I cleaned these up as well as moving all the unlock logic to the end of the function. This patch should also resolve the issue people were having with the replacecall channel never being unlocked with one legged calls. (closes issue #15151) Reported by: irroot Patches: invite_w_replaces_1.4.diff uploaded by dvossel (license 671) Tested by: irroot, dvossel Review: https://reviewboard.asterisk.org/r/371/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@219305 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-17Merged revisions 219264 via svnmerge from file1-1/+2
https://origsvn.digium.com/svn/asterisk/trunk ........ r219264 | file | 2009-09-17 14:57:39 -0500 (Thu, 17 Sep 2009) | 2 lines Ensure no spaces exist before "refresher=" when doing the comparison. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@219265 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-16Merged revisions 218933 via svnmerge from mmichelson1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r218933 | mmichelson | 2009-09-16 14:25:36 -0500 (Wed, 16 Sep 2009) | 12 lines Reverse order of args to fread. This way, we don't always write a null byte into byte 1 of the buffer (closes issue #15905) Reported by: ebroad Patches: freadfix.patch uploaded by ebroad (license 878) Tested by: ebroad ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@218935 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-16Merged revisions 218918 via svnmerge from file1-10/+13
https://origsvn.digium.com/svn/asterisk/trunk ........ r218918 | file | 2009-09-16 13:31:47 -0500 (Wed, 16 Sep 2009) | 5 lines On TCP and TLS connections do not attempt to stop retransmission of the packet internally. This was preventing responses from being properly processed because the packet was not being found causing handle_response to return prematurely. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@218931 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-15Merged revisions 218687 via svnmerge from dvossel1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r218687 | dvossel | 2009-09-15 14:22:37 -0500 (Tue, 15 Sep 2009) | 2 lines upward bound checking for port string to int conversion ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@218690 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-15Merged revisions 218586 via svnmerge from mnicholson1-1/+5
https://origsvn.digium.com/svn/asterisk/trunk ................ r218586 | mnicholson | 2009-09-15 11:15:02 -0500 (Tue, 15 Sep 2009) | 15 lines Merged revisions 218578 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r218578 | mnicholson | 2009-09-15 11:03:54 -0500 (Tue, 15 Sep 2009) | 8 lines Send request contact header field with response to registrer queries instead of the address of record. (closes issue #14438) Reported by: ravindrad Patches: regquerypatch uploaded by ravindrad (license 684) Tested by: ravindrad ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@218601 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-15Merged revisions 218566 via svnmerge from mmichelson1-2/+3
https://origsvn.digium.com/svn/asterisk/trunk ........ r218566 | mmichelson | 2009-09-15 10:40:14 -0500 (Tue, 15 Sep 2009) | 4 lines Use a better method of ensuring null-termination of the buffer while reading the SDP when using TCP. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@218573 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-15Merged revisions 218499,218504 via svnmerge from mmichelson1-1/+2
https://origsvn.digium.com/svn/asterisk/trunk ........ r218499 | mmichelson | 2009-09-15 09:59:50 -0500 (Tue, 15 Sep 2009) | 3 lines Fix off-by-one error when reading SDP sent over TCP. ........ r218504 | mmichelson | 2009-09-15 10:05:53 -0500 (Tue, 15 Sep 2009) | 3 lines Ensure that SDP read from TCP socket is null-terminated. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@218505 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-14Merged revisions 218430 via svnmerge from jpeeler1-16/+23
https://origsvn.digium.com/svn/asterisk/trunk ................ r218430 | jpeeler | 2009-09-14 17:38:25 -0500 (Mon, 14 Sep 2009) | 18 lines Merged revisions 218401 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r218401 | jpeeler | 2009-09-14 16:47:11 -0500 (Mon, 14 Sep 2009) | 11 lines Fix handling of DAHDI_EVENT_REMOVED event to prevent crash in do_monitor. After talking to rmudgett about some of his recent iflist locking changes, it was determined that the only place that would destroy a channel without being explicitly to do so was in handle_init_event. The loop to walk the interface list has been modified to wait to destroy the channel until the dahdi_pvt of the channel to be destroyed is no longer needed. (closes issue #15378) Reported by: samy ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@218431 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-13gcc 4.4: Remove a nop memset size 0 that annoys gcctzafrir1-1/+0
This memset doesn't write beyond the end of the buffer. (tmpbuf has size of 4). Merged revisions 218184 via svnmerge from http://svn.digium.com/svn/asterisk/trunk git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@218216 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-10Merged revisions 217916 via svnmerge from tilghman2-4/+9
https://origsvn.digium.com/svn/asterisk/trunk ........ r217916 | tilghman | 2009-09-10 18:12:16 -0500 (Thu, 10 Sep 2009) | 2 lines Make calltoken support work with realtime users and peers. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@217920 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-10sip peer matching by address only with TCP/TLSdvossel1-41/+41
This patch removes the contact header matching logic and adds logic to match all tcp/tls connections by ip only. Thanks to oej for finding the issue and suggesting solutions. Review: https://reviewboard.asterisk.org/r/355/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@217913 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-10Merged revisions 217807 via svnmerge from dvossel1-4/+25
https://origsvn.digium.com/svn/asterisk/trunk ................ r217807 | dvossel | 2009-09-10 16:07:47 -0500 (Thu, 10 Sep 2009) | 28 lines Merged revisions 217806 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r217806 | dvossel | 2009-09-10 16:06:07 -0500 (Thu, 10 Sep 2009) | 22 lines IAX2 encryption regression The IAX2 Call Token security patch inadvertently broke the use of encryption due to the reorganization of code in the socket_process() function. When encryption is used, an incoming full frame must first be decrypted before the information elements can be parsed. The security release mistakenly moved IE parsing before decryption in order to process the new Call Token IE. To resolve this, decryption of full frames is once again done before looking into the frame. This involves searching for an existing callno, checking the pvt to see if encryption is turned on, and decrypting the packet before the internal fields of the full frame are accessed. (closes issue #15834) Reported by: karesmakro Patches: iax2_encryption_fix_1.4.diff uploaded by dvossel (license 671) Tested by: dvossel, karesmakro Review: https://reviewboard.asterisk.org/r/355/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@217858 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-10Merged revisions 217593 via svnmerge from oej1-1/+2
https://origsvn.digium.com/svn/asterisk/trunk ........ r217593 | oej | 2009-09-10 14:06:55 +0200 (Tor, 10 Sep 2009) | 8 lines Include ActionID in all events that are responsed to AMI Action SIPShowRegistry (closes issue #15868) Reported by: nic_bellamy Patches: manager_SIPshowregistry_actionid.patch uploaded by nic bellamy (license 299) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@217596 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-09Merged revisions 217368 via svnmerge from oej1-0/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r217368 | oej | 2009-09-09 12:39:43 +0200 (Ons, 09 Sep 2009) | 2 lines Not having any TLS session to write to is a serious XMIT_ERROR. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@217405 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-08Merged revisions 216993 via svnmerge from dvossel1-4/+4
https://origsvn.digium.com/svn/asterisk/trunk ........ r216993 | dvossel | 2009-09-08 09:26:30 -0500 (Tue, 08 Sep 2009) | 14 lines caller id number empty parse_uri was not being given the correct scheme's, as a result, uri parsing did not parse the username correctly. One of the side effects of this is an empty caller id. (closes issue #15839) Reported by: ebroad Patches: blank_cidv2.patch uploaded by ebroad (license 878) parse_uri_fix.diff uploaded by dvossel (license 671) Tested by: ebroad, dvossel ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@216996 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-07Merged revisions 216842 via svnmerge from oej1-2/+3
https://origsvn.digium.com/svn/asterisk/trunk ........ r216842 | oej | 2009-09-07 18:35:12 +0200 (MÃ¥n, 07 Sep 2009) | 2 lines Make sure we reset global_exclude_static at channel reload ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@216843 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-07Merged revisions 216695 via svnmerge from oej1-1/+2
https://origsvn.digium.com/svn/asterisk/trunk ........ r216695 | oej | 2009-09-07 15:06:19 +0200 (MÃ¥n, 07 Sep 2009) | 8 lines If there is no session timer in the INVITE, set it to default value (not unset minimum = -1) Patch by oej closes issue #15621 Reported by: fnordian Tested by: atis ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@216696 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-07Add doc and turn off premature media filter by defaultoej1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@216654 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-07Merged revisions 216438 via svnmerge from oej1-3/+11
https://origsvn.digium.com/svn/asterisk/trunk ................ r216438 | oej | 2009-09-04 16:02:34 +0200 (Fre, 04 Sep 2009) | 35 lines Merged revisions 216430 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r216430 | oej | 2009-09-04 15:45:48 +0200 (Fre, 04 Sep 2009) | 27 lines Make apps send PROGRESS control frame for early media and fix too early media issue in SIP The issue at hand is that some legacy (dying) PBX systems send empty media frames on PRI links *before* any call progress. The SIP channel receives these frames and by default signals 183 Session progress and starts sending media. This will cause phones to play silence and ignore the later 180 ringing message. A bad user experience. The fix is twofold: - We discovered that asterisk apps that support early media ("noanswer") did not send any PROGRESS frame to indicate early media. Fixed. - We introduce a setting in chan_sip so that users can disable any relay of media frames before the outbound channel actually indicates any sort of call progress. In 1.4, 1.6.0 and 1.6.1, this will be disabled for backward compatibility. In later versions of Asterisk, this will be enabled. We don't assume that it will change your Asterisk phone experience - only for the better. We encourage third-party application developers to make sure that if they have applications that wants to send early media, add a PROGRESS control frame transmission to make sure that all channel drivers actually will start sending early media. This has not been the default in Asterisk previous to this patch, so if you got inspiration from our code, you need to update accordingly. Sorry for the trouble and thanks for your support. This code has been running for a few months in a large scale installation (over 250 servers with PRI and/or BRI links to old PBX systems). That's no proof that this is an excellent patch, but, well, it's tested :-) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@216645 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-03Merged revisions 215955 via svnmerge from dvossel4-145/+1232
https://origsvn.digium.com/svn/asterisk/trunk ........ r215955 | dvossel | 2009-09-03 11:31:54 -0500 (Thu, 03 Sep 2009) | 6 lines Merge code associated with AST-2009-006 (closes issue #12912) Reported by: rathaus Tested by: tilghman, russell, dvossel, dbrooks ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@216003 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-03Merged revisions 215758 via svnmerge from twilson1-8/+73
https://origsvn.digium.com/svn/asterisk/trunk ................ r215758 | twilson | 2009-09-02 18:31:04 -0500 (Wed, 02 Sep 2009) | 25 lines Merged revisions 215682 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r215682 | twilson | 2009-09-02 16:41:22 -0500 (Wed, 02 Sep 2009) | 18 lines Re-send non-100 provisional responses to prevent cancellation From section 13.3.1.1 of RFC 3261: If the UAS desires an extended period of time to answer the INVITE, it will need to ask for an "extension" in order to prevent proxies from canceling the transaction. A proxy has the option of canceling a transaction when there is a gap of 3 minutes between responses in a transaction. To prevent cancellation, the UAS MUST send a non-100 provisional response at every minute, to handle the possibility of lost provisional responses. (closes issue #11157) Reported by: rjain Tested by: twilson Review: https://reviewboard.asterisk.org/r/315/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@215759 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-02Merged revisions 215681 via svnmerge from dvossel1-17/+28
https://origsvn.digium.com/svn/asterisk/trunk ........ r215681 | dvossel | 2009-09-02 16:39:31 -0500 (Wed, 02 Sep 2009) | 10 lines port string to int conversion using sscanf There are several instances where a port is parsed from a uri or some other source and converted to an int value using atoi(), if for some reason the port string is empty, then a standard port is used. This logic is used over and over, so I created a function to handle it in a safer way using sscanf(). ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@215687 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-02Merged revisions 215522 via svnmerge from dvossel1-43/+40
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: http://svn.digium.com/svn/asterisk/branches/1.6.0@215525 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-31Merged revisions 214945 via svnmerge from tilghman1-0/+3
https://origsvn.digium.com/svn/asterisk/trunk ................ r214945 | tilghman | 2009-08-31 11:18:33 -0500 (Mon, 31 Aug 2009) | 14 lines Merged revisions 214940 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r214940 | tilghman | 2009-08-31 11:16:52 -0500 (Mon, 31 Aug 2009) | 7 lines Also unlock the "other" channel, when returning, due to glare. (closes issue #15787) Reported by: tim_ringenbach Patches: chan_local.diff uploaded by tim ringenbach (license 540) Tested by: tim_ringenbach ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@214957 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-26Merged revisions 214199 via svnmerge from tilghman1-1/+1
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: http://svn.digium.com/svn/asterisk/branches/1.6.0@214200 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-21Merged revisions 213716 via svnmerge from dvossel1-14/+7
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: http://svn.digium.com/svn/asterisk/branches/1.6.0@213727 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-19Merged revisions 213093 via svnmerge from tilghman1-1/+1
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: http://svn.digium.com/svn/asterisk/branches/1.6.0@213094 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-18Merged revisions 212758 via svnmerge from rmudgett1-29/+30
https://origsvn.digium.com/svn/asterisk/trunk ................ r212758 | rmudgett | 2009-08-18 11:29:47 -0500 (Tue, 18 Aug 2009) | 9 lines Merged revisions 212727 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r212727 | rmudgett | 2009-08-18 11:00:56 -0500 (Tue, 18 Aug 2009) | 1 line Removed some deadwood and added some doxygen comments. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@212765 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-17Merged revisions 212506 via svnmerge from jpeeler1-0/+5
https://origsvn.digium.com/svn/asterisk/trunk ................ r212506 | jpeeler | 2009-08-17 11:50:45 -0500 (Mon, 17 Aug 2009) | 19 lines Merged revisions 212498 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r212498 | jpeeler | 2009-08-17 11:34:56 -0500 (Mon, 17 Aug 2009) | 12 lines Fix segfault when reloading chan_misdn. If more ports were specified than configured in misdn.conf a reload would crash asterisk. The problem was the unconfigured port was using data from the previously configured port. When the data for an unconfigured port was freed a crash would result from the double free. (closes issue #12113) Reported by: agupta Patches: bug12113.patch uploaded by jpeeler (license 325) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@212507 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-17Merged revisions 212431 via svnmerge from rmudgett1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r212431 | rmudgett | 2009-08-17 10:42:51 -0500 (Mon, 17 Aug 2009) | 16 lines Merged revisions 212430 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r212430 | rmudgett | 2009-08-17 10:36:28 -0500 (Mon, 17 Aug 2009) | 1 line Fix uninitialized variable. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@212432 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-13Merged revisions 212113 via svnmerge from kpfleming1-1/+1
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: http://svn.digium.com/svn/asterisk/branches/1.6.0@212114 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-13Merged revisions 212067 via svnmerge from file1-1/+1
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: http://svn.digium.com/svn/asterisk/branches/1.6.0@212068 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-12Merged revisions 211876 via svnmerge from mnicholson1-3/+6
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: http://svn.digium.com/svn/asterisk/branches/1.6.0@211952 f38db490-d61c-443f-a65b-d21fe96a405b
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