asterisk

Commit Graph

Author	SHA1	Message	Date
David Vossel	c0ee60419d	Deadlock between ast_cel_report_event and ast_do_masquerade chan_sip calls pbx_exec on a pvt's owner channel while only the pvt lock is held. Since pbx_exec calls ast_cel_report_event which attempts to lock the channel, invalid locking order occurs. Channels should be locked before pvt's. (closes issue #15512) Reported by: lmsteffan Patches: ast_cel_deadlock_15512.diff uploaded by dvossel (license 671) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222981 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	b764544641	makes externtcpport and externtlsport static variables externtcpport and externtlsport need to be declared as static variables. Thanks to russell for finding and pointing this out. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222947 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	890d500287	Merged revisions 222797 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r222797 \| rmudgett \| 2009-10-08 11:33:06 -0500 (Thu, 08 Oct 2009) \| 12 lines Fix memory leak if chan_misdn config parameter is repeated. Memory leak when the same config option is set more than once in an misdn.conf section. Why must this be considered? Templates! Defining a template with default port options and later adding to or overriding some of them. Patches: memleak-misdn.patch JIRA ABE-1998 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222799 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	9456ab2724	Deadlock in channel masquerade handling Channels are stored in an ao2_container. When accessing an item within an ao2_container the proper locking order is to first lock the container, and then the items within it. In ast_do_masquerade both the clone and original channel must be locked for the entire duration of the function. The problem with this is that it attemptes to unlink and link these channels back into the ao2_container when one of the channel's name changes. This is invalid locking order as the process of unlinking and linking will lock the ao2_container while the channels are locked!!! Now, both the channels in do_masquerade are unlinked from the ao2_container and then locked for the entire function. At the end of the function both channels are unlocked and linked back into the container with their new names as hash values. This new method of requiring all channels and tech pvts to be unlocked before ast_do_masquerade() or ast_change_name() required several changes throughout the code base. (closes issue #15911) Reported by: russell Patches: masq_deadlock_trunk.diff uploaded by dvossel (license 671) Tested by: dvossel, atis (closes issue #15618) Reported by: lmsteffan Patches: deadlock_local_attended_transfers_trunk.diff uploaded by dvossel (license 671) Tested by: lmsteffan, dvossel Review: https://reviewboard.asterisk.org/r/387/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222761 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	49b90d5e61	Merged revisions 222691 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r222691 \| rmudgett \| 2009-10-07 16:51:24 -0500 (Wed, 07 Oct 2009) \| 14 lines chan_misdn.c:process_ast_dsp() memory leak misdn.conf: astdtmf must be set to "yes". With "no", buffer loss does not occur. The translated frame "f2" when passing through ast_dsp_process() is not freed whenever it is not used further in process_ast_dsp(). Then in the end it is never ever freed. Patches: translate.patch JIRA ABE-1993 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222692 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	4ae6bee6da	Change ringt (ring timeout) styles to be consistent across chan_dahdi. (closes issue #15684) Reported by: alecdavis Patches: chan_dahdi.bug15684.diff2.txt uploaded by alecdavis (license 585) Tested by: alecdavis git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222652 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	f819ce5b20	Merged revisions 222542 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r222542 \| dvossel \| 2009-10-07 12:41:21 -0500 (Wed, 07 Oct 2009) \| 8 lines crash on transfer handle_invite_replaces() attempts to uplock a pvt's owner channel without first verifing that it exists. (issue #16027) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222543 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	b5eb0449c0	Merged revisions 222462 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r222462 \| jpeeler \| 2009-10-06 18:51:19 -0500 (Tue, 06 Oct 2009) \| 8 lines Add missing unlock(s) in dahdi_read (two cases in trunk) (closes issue #15683) Reported by: alecdavis ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222463 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	1d40faebac	contact header port ignored transport when using externip This patch adds support for TCP/TLS in the Contact header when using NAT, specifically externip or externhost. The original issue was that Asterisk sent 5060 as the port in the contact header whether TLS was used or not. Additionally, this patch adds 2 config options to sip.conf, specifically externtcpport and externtlsport. This allows a user to specify different external ports for TCP and TLS other than those used internally, this is especially useful in in a PAT/port redirection setup. Thanks to ebroad for reporting the issue and providing the patch! (closes issue #15880) Reported by: ebroad Patches: portmap.patch uploaded by ebroad (license 878) externtXXport_v2.patch uploaded by ebroad (license 878) Tested by: ebroad Review: https://reviewboard.asterisk.org/r/392/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222398 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	f7fa417130	Fix 222298 (crash during destruction of second channel when variable set with setvar). I mistakenly reasoned that setvar would be used on all channels. Since it can be set per channel, give each dahdi channel a copy of the variable. (related to #15899) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222351 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	0c7f4cfb85	Fix crash during destruction of second channel when variable set with setvar. The setvar line in chan_dahdi.conf is shared among all the channels, so make sure to only free the resources only when the last channel is destroyed. (closes issue #15899) Reported by: tzafrir git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222298 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tzafrir Cohen	0c3cd2ee45	Make sure digit events are not reported as "ERROR" dahdievent_to_analogevent used a simple switch statement to convert DAHDI event numbers to "ANALOG_*" event numbers. However "digit" events (DAHDI_EVENT_PULSEDIGIT, DAHDI_EVENT_DTMFDOWN, DAHDI_EVENT_DTMFUP) are accompannied by the digit in the low word of the event number. This fix makes dahdievent_to_analogevent() return the event number as-is for such an event. This is also required to fix #15924 (in addition to r222108). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222237 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	1c9fe00920	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: https://origsvn.digium.com/svn/asterisk/trunk@222176 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	20743ec07d	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: https://origsvn.digium.com/svn/asterisk/trunk@222110 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	10e8ee1746	Add a few missing events to analog_handle_event. The reported bug was actually only for pulsedigit, dtmfup, and dtmfdown handling. Also added recognition for fax events (just some verbose output) and fixed handling for the ec_disabled_event. In order to make comparing the analog version of events to the DAHDI events easier, the ordering has been changed to follow that of the DAHDI events. (closes issue #15924) Reported by: tzafrir git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222108 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	3cce68d329	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: https://origsvn.digium.com/svn/asterisk/trunk@222030 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	80f0a242a7	Whitespace change. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221881 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	1a02b4c659	Whitespace change. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221880 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	3b83d2b414	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: https://origsvn.digium.com/svn/asterisk/trunk@221844 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	c0a884ba29	Revision 220906 (a merge from 1.4) was not merged correctly, causing a problem with non-dynamic peers. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221705 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	9c05faf76d	Prevent deadlock if chan_dahdi attempts to change PRI channel names. The PRI channels can no longer change the channel name if a different B channel is selected during call negotiation. To prevent using the channel name to infer what B channel a call is using and to avoid name collisions, the channel name format is changed. The new channel naming for PRI channels is: DAHDI/ISDN-<span>-<sequence-number> git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221701 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	aaa7284c00	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 Review: https://reviewboard.asterisk.org/r/357/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221697 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	da169b2db4	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: https://origsvn.digium.com/svn/asterisk/trunk@221589 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Olle Johansson	73697dc2c7	Simplify code for porturi, use TRUE/FALSE constructs when it's just TRUE or FALSE. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221554 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	d043f52a2d	Cleaned up merge from r221432 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221484 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	a5eee590f4	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: https://origsvn.digium.com/svn/asterisk/trunk@221432 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	10ce6cd757	Use rtp properties instead of adding a callback Thanks, Josh. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221278 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	865daf4858	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: https://origsvn.digium.com/svn/asterisk/trunk@221266 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	6f5e763fe5	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: https://origsvn.digium.com/svn/asterisk/trunk@220906 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	f3f456f8b6	Miscellaneous minor changes. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220792 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	05f94a05c2	Fix building of registration entry in build_peer when using callbackextension Check for remotesecret option was unintentionally always true, which therefore caused the secret option to never be used. Thanks to dvossel for pointing out the exact fix. (closes issue #15943) Reported by: tpsast git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220718 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	307bf124d2	Locking issues dealing with service_lock. * Removed unneeded and uninitialized service_lock. * Fixed potential locking imbalance in pri_dchannel():PRI_EVENT_RESTART. * Fixed verbose message typo in pri_dchannel():PRI_EVENT_RESTART. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220672 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	146c352144	Reduce indentation in sig_pri_available(). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220543 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Philippe Sultan	b11b94a083	Add JABBER_RECEIVE as a dialplan function, implement SendText in Jingle channels JABBER_RECEIVE (along with JabberSend) makes Asterisk interact with users over XMPP to process calls. SendText can be used instead of JabberSend in the context of XMPP based voice channels (chan_gtalk and chan_jingle). (closes issue #12569) Reported by: eech55 Tested by: phsultan, asannucci, lmadsen, jtodd, maxgo Review: https://reviewboard.asterisk.org/r/88/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220457 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	944b05d51a	Ensure the numeric portion of the P-Asserted-Identity header is properly escaped. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220174 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	c68a2d9d30	Add support for 'setvar=' for MGCP device lines, like other channel drivers provide. (closes issue #14818) Reported by: alea-soluciones Patches: chan_mgcp-setvars-svn-trunk-r219899.patch uploaded by alea (license 514) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@219952 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	9329079bb4	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: https://origsvn.digium.com/svn/asterisk/trunk@219721 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	5996ab0ee2	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: https://origsvn.digium.com/svn/asterisk/trunk@219587 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	95be40493a	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: https://origsvn.digium.com/svn/asterisk/trunk@219520 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	e85e39899f	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: https://origsvn.digium.com/svn/asterisk/trunk@219451 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	06782af238	fixes deadlock when performing directed pickup w Invite/replaces (closes issue #15340) Reported by: lmsteffan Patches: deadlock.patch uploaded by lmsteffan (license 779) Tested by: lmsteffan git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@219371 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	dc6f08e275	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: https://origsvn.digium.com/svn/asterisk/trunk@219324 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	0284951e77	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: https://origsvn.digium.com/svn/asterisk/trunk@219304 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Joshua Colp	8a3f2fff91	Ensure no spaces exist before "refresher=" when doing the comparison. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@219264 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	19aeff195a	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: https://origsvn.digium.com/svn/asterisk/trunk@218933 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Joshua Colp	5c52a7a746	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: https://origsvn.digium.com/svn/asterisk/trunk@218918 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	c373c8807e	upward bound checking for port string to int conversion git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@218687 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	6f6998fef7	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: https://origsvn.digium.com/svn/asterisk/trunk@218586 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	0d5e318cb2	Add some changes related to 218430. * Remove thread_spawned in handle_init_event since it was never used * Always check handle_init_event in case a channel is destroyed git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@218583 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	15c7e6dea2	Use a better method of ensuring null-termination of the buffer while reading the SDP when using TCP. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@218566 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago

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6251 Commits (c0ee60419d622ca87d097b314c5b2741a8b94ce0)