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2010-03-31Merged revisions 255504 via svnmerge from lmadsen1-1/+2
https://origsvn.digium.com/svn/asterisk/trunk ........ r255504 | lmadsen | 2010-03-31 12:48:09 -0500 (Wed, 31 Mar 2010) | 5 lines Add documentation clarifying when 't' and 'T' can be used. (closes issue #17021) Reported by: kovzol Tested by: lmadsen, kovzol, davidw, ebroad ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@255506 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-26Finish syncing documentation for tlsbindaddr.lmadsen1-0/+2
(issue #17054) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@255068 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-26Merged revisions 255021 via svnmerge from lmadsen1-2/+2
https://origsvn.digium.com/svn/asterisk/trunk ........ r255021 | lmadsen | 2010-03-26 14:07:38 -0500 (Fri, 26 Mar 2010) | 8 lines Update confusing documentation for tlsbindaddr. Update some confusing documentation for the tlsbindaddr option in sip.conf.sample. Point at a link instead which has better documentation. (closes issue #17054) Reported by: klaus3000 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@255026 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-13Merged revisions 252089 via svnmerge from twilson1-2/+0
https://origsvn.digium.com/svn/asterisk/trunk ........ r252089 | twilson | 2010-03-12 16:04:51 -0600 (Fri, 12 Mar 2010) | 20 lines Only change the RTP ssrc when we see that it has changed This change basically reverts the change reviewed in https://reviewboard.asterisk.org/r/374/ and instead limits the updating of the RTP synchronization source to only those times when we detect that the other side of the conversation has changed the ssrc. The problem is that SRCUPDATE control frames are sent many times where we don't want a new ssrc, including whenever Asterisk has to send DTMF in a normal bridge. This is also not the first time that this mistake has been made. The initial implementation of the ast_rtp_new_source function also changed the ssrc--and then it was removed because of this same issue. Then, we put it back in again to fix a different issue. This patch attempts to only change the ssrc when we see that the other side of the conversation has changed the ssrc. It also renames some functions to make their purpose more clear. Review: https://reviewboard.asterisk.org/r/540/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@252135 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-02Merged revisions 249893 via svnmerge from dvossel1-0/+7
https://origsvn.digium.com/svn/asterisk/trunk ........ r249893 | dvossel | 2010-03-02 13:08:38 -0600 (Tue, 02 Mar 2010) | 11 lines fixes adaptive jitterbuffer configuration When configuring the adaptive jitterbuffer, the target_extra value not only could not be set from the configuration, but was not even being set to its proper default. This value is required in order for the adaptive jitterbuffer to work correctly. To resolve this a config option has been added to expose this value to the conf files, and a default value is provided when no config specific value is present. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@249896 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-12Merged revisions 239520 via svnmerge from lmadsen1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r239520 | lmadsen | 2010-01-12 12:22:45 -0600 (Tue, 12 Jan 2010) | 6 lines Note that direct T.38 is not supported. (closes issue #16411) Reported by: stanusr Patches: __20091210-sip.conf.sample-documentation.txt uploaded by lmadsen (license 10) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@239522 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-07Merged revisions 238313 via svnmerge from tzafrir1-1/+7
https://origsvn.digium.com/svn/asterisk/trunk ........ r238313 | tzafrir | 2010-01-07 11:14:57 +0200 (ה', 07 ינו 2010) | 2 lines Document the usefulness of explicit udp:// in the register string ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@238351 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-16Merged revisions 235298 via svnmerge from jsmith1-0/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r235298 | jsmith | 2009-12-15 23:24:58 -0600 (Tue, 15 Dec 2009) | 11 lines Merged revisions 235181 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r235181 | jsmith | 2009-12-15 15:07:55 -0600 (Tue, 15 Dec 2009) | 4 lines Add a line showing that we can use CIDR notation. patch by jsmith, after discussion with jtodd ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@235333 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-23Merged revisions 230881 via svnmerge from file1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r230881 | file | 2009-11-23 09:45:45 -0600 (Mon, 23 Nov 2009) | 7 lines Change fax detection in chan_sip so it behaves as one would expect. Internally the way T.38 is negotiated has changed and the option no longer reflects a behavior that is valid. It will now look for a CNG tone on received calls and if present send the call to the 'fax' extension. It is then up to the application or channel to request the switch over to T.38. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@230883 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-28Merged revisions 226384 via svnmerge from lmadsen1-0/+7
https://origsvn.digium.com/svn/asterisk/trunk ................ r226384 | lmadsen | 2009-10-28 15:11:07 -0500 (Wed, 28 Oct 2009) | 17 lines Merged revisions 226382 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r226382 | lmadsen | 2009-10-28 15:06:13 -0500 (Wed, 28 Oct 2009) | 9 lines Update documentation in sip.conf.sample. Update the documentation in sip.conf.sample in order to make it more clear that directmedia/canreinvite do not cause Asterisk to ignore reINVITEs. It is only used to stop Asterisk from generating a reINVITE, but does not stop it from accepting them if necessary. (closes issue #15644) Reported by: lmadsen ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@226386 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-21Merged revisions 225033 via svnmerge from dvossel1-0/+9
https://origsvn.digium.com/svn/asterisk/trunk ................ r225033 | dvossel | 2009-10-21 09:39:10 -0500 (Wed, 21 Oct 2009) | 27 lines 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: http://svn.digium.com/svn/asterisk/branches/1.6.1@225062 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-05Merged revisions 222110 via svnmerge from kpfleming1-7/+22
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/1.6.1@222112 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-30Merged revisions 221432 via svnmerge from mnicholson1-0/+3
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/1.6.1@221478 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-30Merged revisions 221266 via svnmerge from twilson1-0/+2
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/1.6.1@221302 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-07Make code and documentation agree with each otheroej1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@216657 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-07Merged revisions 216438 via svnmerge from oej1-0/+7
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/1.6.1@216646 f38db490-d61c-443f-a65b-d21fe96a405b
2009-06-26Merged revisions 203699 via svnmerge from file1-4/+6
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: http://svn.digium.com/svn/asterisk/branches/1.6.1@203703 f38db490-d61c-443f-a65b-d21fe96a405b
2009-06-15Merged revisions 165180,200689 via svnmerge from kpfleming1-12/+22
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: http://svn.digium.com/svn/asterisk/branches/1.6.1@200707 f38db490-d61c-443f-a65b-d21fe96a405b
2009-06-02Merged revisions 198791 via svnmerge from file1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r198791 | file | 2009-06-02 10:48:06 -0300 (Tue, 02 Jun 2009) | 5 lines Correct documentation for the register line, specifically where the domain should be specified. (closes issue #14367) Reported by: Nick_Lewis ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@198793 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-22Merged revisions 196416 via svnmerge from dvossel1-4/+7
https://origsvn.digium.com/svn/asterisk/trunk ........ r196416 | dvossel | 2009-05-22 16:09:45 -0500 (Fri, 22 May 2009) | 19 lines SIP set outbound transport type from Registration In sip.conf the transport option allows for the configuration of what transport types (udp, tcp, and tls) a peer will accept, but only the first type listed was used for outbound connections. This patch changes this. Now the default transport type is only used until the peer registers. When registration takes place the transport type is parsed out of the Contact header. If the Contact header's transport type is equal to one that the peer supports, the peer's default transport type for outbound connections is set to match the Contact header's type. If the Contact header's transport type is not present, then the peer's default transport type is set to match the one the peer registered with. When a peer unregisters or the registration expires, the default transport type for that peer is reset. (closes issue #12282) Reported by: rjain Patches: reg_patch_1.diff uploaded by dvossel (license 671) Tested by: dvossel (closes issue #14727) Reported by: pj Patches: reg_patch_3.diff uploaded by dvossel (license 671) Tested by: pj, dvossel Review: https://reviewboard.asterisk.org/r/249/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@196453 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-02Merged revisions 186060 via svnmerge from tilghman1-2/+4
https://origsvn.digium.com/svn/asterisk/trunk ................ r186060 | tilghman | 2009-04-02 12:10:28 -0500 (Thu, 02 Apr 2009) | 16 lines Merged revisions 186059 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r186059 | tilghman | 2009-04-02 12:09:13 -0500 (Thu, 02 Apr 2009) | 9 lines Merged revisions 186056 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r186056 | tilghman | 2009-04-02 12:02:18 -0500 (Thu, 02 Apr 2009) | 2 lines Fix for AST-2009-003 ........ ................ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@186062 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-11Merged revisions 181499 via svnmerge from mvanbaak1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk Provide correct hint to debug SIP trouble in the default config (closes issue #14646) Reported by: strk git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@181521 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-29Merged revisions 171880 via svnmerge from oej1-1/+21
https://origsvn.digium.com/svn/asterisk/trunk ........ r171880 | oej | 2009-01-28 14:26:31 +0100 (Ons, 28 Jan 2009) | 2 lines Add some more notes about device matching. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@172218 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-28Merged revisions 171838 via svnmerge from oej1-0/+19
https://origsvn.digium.com/svn/asterisk/trunk ................ r171838 | oej | 2009-01-28 14:11:44 +0100 (Ons, 28 Jan 2009) | 10 lines Merged revisions 171837 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r171837 | oej | 2009-01-28 14:07:27 +0100 (Ons, 28 Jan 2009) | 2 lines Add a better explanation of the difference between the device namespace and the dialplan for newbies. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@171857 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-15Merged revisions 168711 via svnmerge from oej1-4/+8
https://origsvn.digium.com/svn/asterisk/trunk ........ r168711 | oej | 2009-01-15 18:55:53 +0100 (Tor, 15 Jan 2009) | 4 lines Clarify some misunderstandings and make it even more clear that you can refer to a peer in the register= line. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@168715 f38db490-d61c-443f-a65b-d21fe96a405b
2008-09-12Merged revisions 142866 via svnmerge from tilghman1-0/+14
https://origsvn.digium.com/svn/asterisk/trunk ................ r142866 | tilghman | 2008-09-12 15:49:46 -0500 (Fri, 12 Sep 2008) | 18 lines Merged revisions 142865 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r142865 | tilghman | 2008-09-12 15:37:18 -0500 (Fri, 12 Sep 2008) | 11 lines Create rules for disallowing contacts at certain addresses, which may improve the security of various installations. As this does not change any default behavior, it is not classified as a direct security fix for anything within Asterisk, but may help PBX admins better secure their SIP servers. (closes issue #11776) Reported by: ibc Patches: 20080829__bug11776.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76, blitzrage ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@142868 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-15Merged revisions 138260 via svnmerge from tilghman1-1/+3
https://origsvn.digium.com/svn/asterisk/trunk ................ r138260 | tilghman | 2008-08-15 17:54:57 -0500 (Fri, 15 Aug 2008) | 16 lines Merged revisions 138258 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r138258 | tilghman | 2008-08-15 17:33:42 -0500 (Fri, 15 Aug 2008) | 8 lines More fixes for realtime peers. (closes issue #12921) Reported by: Nuitari Patches: 20080804__bug12921.diff.txt uploaded by Corydon76 (license 14) 20080815__bug12921.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@138261 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-14Merged revisions 137732 via svnmerge from russell1-435/+435
https://origsvn.digium.com/svn/asterisk/trunk ................ r137732 | russell | 2008-08-14 09:15:50 -0500 (Thu, 14 Aug 2008) | 12 lines Merged revisions 137731 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r137731 | russell | 2008-08-14 09:05:23 -0500 (Thu, 14 Aug 2008) | 4 lines Comments in this config file were aligned only if your tab size was set to 8. So, convert tabs to spaces so that things should be aligned regardless of what tab size you use in your editor. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@137813 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-01SIP should use the transport type set in the Moved Temporarily for the nexttilghman1-1/+1
invite. (closes issue #11843) Reported by: pestermann Patches: 20080723__issue11843_302_ignores_transport_16branch.diff uploaded by bbryant (license 36) 20080723__issue11843_302_ignores_transport_trunk.diff uploaded by bbryant (license 36) Tested by: pabelanger git-svn-id: http://svn.digium.com/svn/asterisk/trunk@135126 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-30Move implementation of an attended-transfer-complete sound from one channeltilghman1-2/+4
driver into a common place for multiple channel drivers. (closes issue #13152) Reported by: caio1982 Patches: atxfer_complete_sound3.diff uploaded by caio1982 (license 22) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@134401 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-15Additional option for videosupport (always) that disables the optimization totilghman1-5/+11
fail to setup video RTP if the two endpoints will not support it. This assists with call files and certain transfers to ensure that if two video phones are ever connected, they will always share a video feed. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@130951 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-06- Adding alias "udpbindaddr" for the UDP port to comply with "tcpbindaddr" ↵oej1-5/+2
and "tlsbindaddr". Note: I don't think we can start properly without UDP port open, that needs to be tested. - Removing "bindport" from configuration example, not needed to mention this any more I suggest we deprecate "bindaddr" and "bindport" in trunk (for 1.6.1) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@128525 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-06- Fixing issues with "sip show settings"oej1-1/+2
- Adding IP address for TCP and/or TLS too if auto-domain is enabled and binding to a different IP address - Fixing documentation in sip.conf.sample git-svn-id: http://svn.digium.com/svn/asterisk/trunk@128524 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-05Make TCP disabled by default (it's considered experimental)oej1-3/+4
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@128237 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-05Reformatting the config sampleoej1-16/+15
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@128236 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-01Add a configuration option so the global outboundproxy can use tcptls ↵bbryant1-0/+3
without it being defined by each sip user. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@127154 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-01Merged revisions 126844 via svnmerge from oej1-4/+3
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r126844 | oej | 2008-07-01 14:53:01 +0200 (Tis, 01 Jul 2008) | 5 lines Clear up documentation on "domain=" setting in sip.conf Reported by: davidw (closes issue #12413) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@126845 f38db490-d61c-443f-a65b-d21fe96a405b
2008-06-27Change the way that the transport option works for sip users. transport will ↵bbryant1-0/+7
now take multiple arguments, the first one listed will be the one used for new dialogs, and the rest listed will be acceptable ways for that peer to contact us. This fixes a minor bug where, because SIP TCP/UDP run on the same port, could cause a TCP peer to be saved in the ast_db. There will also be warnings when a transport is changed for an unexpected reason. (issue #12799) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@125891 f38db490-d61c-443f-a65b-d21fe96a405b
2008-06-19Merged revisions 123883 via svnmerge from tilghman1-1/+2
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r123883 | tilghman | 2008-06-19 11:20:41 -0500 (Thu, 19 Jun 2008) | 4 lines Correct description of notifyringing option. (Closes issue #12890) Reported by gminet ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@123887 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-28Merged revisions 118646 via svnmerge from file1-8/+12
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r118646 | file | 2008-05-28 11:23:34 -0300 (Wed, 28 May 2008) | 4 lines Add an option to use the source IP address of RTP as the destination IP address of UDPTL when a specific option is enabled. If the remote side is properly configured (ports forwarded) then UDPTL will flow. (closes issue #10417) Reported by: cstadlmann ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@118647 f38db490-d61c-443f-a65b-d21fe96a405b
2008-04-30Add support for specifying the registration expiry on a per registration ↵file1-1/+1
basis in the register line. This comes from a Switchvox patch. (issue AST-24) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@114912 f38db490-d61c-443f-a65b-d21fe96a405b
2008-04-21(closes issue #6113)jpeeler1-0/+3
Reported by: oej Tested by: jpeeler This patch implements multiple parking lots for parked calls. The default parkinglot is used by default, however setting the channel variable PARKINGLOT in the dialplan will allow use of any other configured parkinglot. See configs/features.conf.sample for more details on setting up another non-default parkinglot. Also, one can (currently) set the default parkinglot to use in the driver configuration file via the parkinglot option. Patch initially written by oej, brought up to date and finalized by mvanbaak, and then stabilized and converted to astobj2 by me. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@114487 f38db490-d61c-443f-a65b-d21fe96a405b
2008-04-16This is the scariest commit I've done in a long time. This is the ↵murf1-2/+16
astobj2-ification of chan_sip. I've tested a number of scenarios like crazy. It used to have 4x the call setup/teardown performance of trunk, but now it's roughly at parity. I will attempt to find the bottlenecks and get it back to the 4x mark. The changes made were somewhat invasive, but the value to the community of these upgrades outweighs waiting further for more testing. Every change being made to chan_sip was lousing this code up when we tried to merge. Peers, Users, Dialogs, are all now astobj2 objects, indexed via hashtables. Refcounting is used to track objects and free them at the bitter end of their lives. Please file issues on bugs.digium.com, and PLEASE, please, please be patient. One natural advantage to all the hash-table work is that loading large sip.conf files full of thousands of peers now goes much faster. One more please: PLEASE help thrash this code and test it. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@114190 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-25Add a special dialplan variable to chan_sip which will cause an audio file ↵file1-0/+2
to be played upon completion of an attended transfer. (closes issue #9239) Reported by: sunder git-svn-id: http://svn.digium.com/svn/asterisk/trunk@110631 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-21Note that the TCP and TLS support is currently considered experimental andrussell1-0/+6
is subject to change while we work out the remaining issues. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@110499 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-18Add manager peerstatus events when peer can't authenticate.oej1-0/+2
(closes issue #11959) Reported by: mostyn Patches: peerstatus3.patch uploaded by mostyn (license 398) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@109316 f38db490-d61c-443f-a65b-d21fe96a405b
2008-02-29Add documentation for setting username/password in SIP dial string.file1-1/+10
(closes issue #11587) Reported by: sobomax Patches: dialstring_doc.diff uploaded by sobomax (license 359) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@105378 f38db490-d61c-443f-a65b-d21fe96a405b
2008-02-25Adding more tls configuration details to sip.conf sample, with a list of ↵bbryant1-0/+21
valid ciphers provided in both files. .. First commit since July, woot git-svn-id: http://svn.digium.com/svn/asterisk/trunk@104088 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-22Documentation updatesoej1-8/+13
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@99483 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-18Merge changes from team/group/sip-tcptlsrussell1-2/+14
This set of changes introduces TCP and TLS support for chan_sip. There are various new options in configs/sip.conf.sample that are used to enable these features. Also, there is a document, doc/siptls.txt that describes some things in more detail. This code was implemented by Brett Bryant and James Golovich. It was reviewed by Joshua Colp and myself. A number of other people participated in the testing of this code, but since it was done outside of the bug tracker, I do not have their names. If you were one of them, thanks a lot for the help! (closes issue #4903, but with completely different code that what exists there.) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@99085 f38db490-d61c-443f-a65b-d21fe96a405b
generated by cgit v1.2.3 (git 2.25.1) at 2024-06-16 22:49:33 +0000