6268 Commits (1ebf7767d0c7fe090b4dd41e110a18988b99985c)

Author	SHA1	Message	Date
Joshua Colp	01ab66275a	Add support for specifying the IP address to use for media streams in sip.conf ... This is the second commit for this and documents the text stream using the configured IP address and fixes a bug in the original patch where the UDPTL stream would also use the different IP address. (closes issue #14729) Reported by: _brent_ Patches: media_address.patch uploaded by brent (license 388) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225089 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Joshua Colp	a31eb5bb35	Revert media_address commit, I'm going to roll a fix to the SDP generation in the next version. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225034 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	984d6500ce	Merged revisions 225032 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r225032 | dvossel | 2009-10-21 09:37:04 -0500 (Wed, 21 Oct 2009) | 20 lines IAX/SIP shrinkcallerid option The shrinking of caller id removes '(', ' ', ')', non-trailing '.', and '-' from the string. This means values such as 555.5555 and test-test result in 555555 and testtest. There are instances, such as Skype integration, where a specific value is passed via caller id that must be preserved unmodified. This patch makes the shrinking of caller id optional in chan_sip and chan_iax in order to support such cases. By default this option is on to preserve previous expected behavior. (closes issue #15940) Reported by: dimas Patches: v2-15940.patch uploaded by dimas (license 88) 15940_shrinkcallerid_trunk.c uploaded by dvossel (license 671) Tested by: dvossel Review: https://reviewboard.asterisk.org/r/408/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225033 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Joshua Colp	28d0ec5421	Add support for specifying the IP address to use for media streams in sip.conf ... (closes issue #14729) Reported by: _brent_ Patches: media_address.patch uploaded by brent (license 388) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225003 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	c5b8e9af7c	Make PRI_SUBCMD_xxx handling subaddress friendly. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224930 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	26638d3a55	Add dynamic range compression support for analog channels. ... (closes issue AST-29) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224637 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Joshua Colp	fd9f9ab01e	Add a callback to sig_pri which is called when sig_pri is going to queue a control frame on a channel. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224491 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	03db5ef0e5	fix typo, sorry ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224335 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	53a95d9c84	Merged revisions 224330 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r224330 | jpeeler | 2009-10-16 20:32:47 -0500 (Fri, 16 Oct 2009) | 13 lines Fix stale caller id data from being reported in AMI NewChannel event The problem here is that chan_dahdi is designed in such a way to set certain values in the dahdi_pvt only once. One of those such values is the configured caller id data in chan_dahdi.conf. For PRI, the configured caller id data could be overwritten during a call. Instead of saving the data and restoring, it was decided that for all non-analog channels it was simply best to not set the configured caller id in the first place and also clear it at the end of the call. (closes issue #15883) Reported by: jsmith ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224331 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	64a32b3ad0	Merged revisions 224260 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r224260 | rmudgett | 2009-10-16 15:25:23 -0500 (Fri, 16 Oct 2009) | 18 lines Never released PRI channels when using Busy() or Congestion() dialplan apps. When the Busy() or Congestion() application is used towards ISDN (an ISDN progress is sent), the responding ISDN Disconnect or Release may contain the ISDN cause user busy or one of the congestion causes. In chan_dahdi.c these causes will only set the needbusy or needcongestion flags and not activate the softhangup procedure. Unfortunately only the latter can interrupt the endless wait loop of Busy()/Congestion(). Result: PRI channels staying in state busy for the rest of asterisk life or until the other end times out and forces the call to clear. (issue #14292) Reported by: tomaso Patches: disc_rel_userbusy.patch uploaded by tomaso (license 564) (This patch is unrelated to the issue.) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224261 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	e3f473f4f3	Allow for adding message body to the SIP NOTIFY message ... Ability has been added to both manager command SIPnotify as well as console command sip notify. Message body is stored in the "Content" variable. An example is present in sip_notify.conf. (closes issue #13926) Reported by: jthurman Patches: sip-notify-svn189463.diff uploaded by gareth (license 208) Tested by: gareth git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224035 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	e197f85b8c	Remove automatic switching from T.38 to voice mode in chan_sip. ... chan_sip has some code to automatically switch from T.38 mode to voice mode when a voice frame is written to the channel while it is in T.38 mode; this was intended to handle the situation when a FAX transmission has ended and the channel is not yet hung up, but is causing problems at the beginning of FAX sessions as well when there are still voice frames 'in flight' at the time the T.38 negotiation completes. This patch removes the automatic switchover, and changes app_fax to explicitly switch off T.38 mode when the FAX transmission process ends. (closes issue #16025) Reported by: jamicque git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@223652 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	9e1598b762	Check the proper page for the SENDRPID flag. ... If a pending reinvite were sent, we might not properly send connected party info since we were checking the wrong flag. This was a rare occurrence, but could still happen nevertheless. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@223617 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	fc27da108d	Merged revisions 223205 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r223205 | dvossel | 2009-10-09 12:52:35 -0500 (Fri, 09 Oct 2009) | 10 lines fixes sip registration using authuser in user.conf (closes issue #14954) Reported by: tornblad Tested by: mmichelson, tornblad, dvossel ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@223206 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	0a50c512da	'auth=' did not parse md5 secret correctly ... (closes issue #15949) Reported by: ebroad Patches: authparsefix.patch uploaded by ebroad (license 878) 15949_trunk.diff uploaded by dvossel (license 671) Tested by: ebroad git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@223132 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	b2a7eecd6e	p->peerauth is always empty in transmit_register() ... When using callbackextension or specifing the peer name in a registration string, the peer's specific auth settings set by the "auth=" strings within the peer definition are not used by the registration. Thanks to ebroad for reporting the issue and providing the patch. (closes issue #15955) Reported by: ebroad Patches: regauthfix.patch uploaded by ebroad (license 878) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@223088 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	799e9962b6	fixed comment line for do_magic_pickup ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@223015 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
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