257 Commits (cef5e9d4340f94de8801aaecfb004a080df8a68f)

Author	SHA1	Message	Date
Terry Wilson	7c140ca0b5	Use remotesecret to authenticate with a remote party ... The remotesecret option was only being used for outbound registration and not for placing calls. This patch uses remotesecret on outbound calls if it is set, otherwise secret is still used. Review: https://reviewboard.asterisk.org/r/1107/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@308678 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Andrew Latham	32447fcfb4	SIP Configuration Documentation ... sip show settings reports qualifyfreq in milliseconds. sip.conf configures qualifyfreg in seconds. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@305648 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	0faad5f43b	Merged revisions 282729 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r282729 | twilson | 2010-08-18 21:12:55 -0500 (Wed, 18 Aug 2010) | 2 lines Add some documentation about codec negotiation to sip.conf ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@282730 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	3e9c3706c0	Merged revisions 245192 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r245192 | mmichelson | 2010-02-06 08:43:03 -0600 (Sat, 06 Feb 2010) | 21 lines Remove useless sip options related to hash table size. First off, these options weren't actually doing anything. By the time the options were parsed, the peer and dialog containers had already been allocated with their default values. Second, hash table size is something that doesn't really make sense to change in a config file. If a user is that interested in changing the hashtable size, he can modify the source itself. I have removed the parsing of the hash_peer, hash_user, and hash_dialog options. I have removed the hash_user_size variable altogether since it is not used at all. I also changed hash_peer_size and hash_dialog_size to be constant, and have changed the symbols to be in all caps as constants typically are. I have also removed the entire section in sip.conf.sample regarding configurable hashtable sizes. ........ (merge to 1.6.2 inspired by issue #17553) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@275469 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	29beb317eb	Merged revisions 274284 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r274284 | twilson | 2010-07-06 17:15:27 -0500 (Tue, 06 Jul 2010) | 18 lines Merged revisions 274280 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r274280 | twilson | 2010-07-06 17:08:20 -0500 (Tue, 06 Jul 2010) | 9 lines Add option to not do a call forward on 482 Loop Detected Asterisk has always set up a forwarded call when receiving a 482 Loop Detected. This prevents handling the call failure by just continuing on in the dialplan. Since this would be a change in behavior, the new option to disable this behavior is forwardloopdetected which defaults to 'yes'. Review: https://reviewboard.asterisk.org/r/764/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@274360 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	0257ac0bf3	Merged revisions 274316 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r274316 | jpeeler | 2010-07-06 17:23:35 -0500 (Tue, 06 Jul 2010) | 14 lines Merged revisions 274283 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r274283 | jpeeler | 2010-07-06 17:15:21 -0500 (Tue, 06 Jul 2010) | 7 lines Correct sip.conf.sample comments for prematuremedia option. (closes issue #17513) Reported by: festr Patches: patch uploaded by festr (license 443) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@274347 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	5a15c4f8ea	Merged revisions 268988 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r268988 | lmadsen | 2010-06-08 10:23:20 -0500 (Tue, 08 Jun 2010) | 8 lines Update note in sip.conf.sample. Update note in sip.conf.sample about externip and externhost with STUN. (closes issue #16323) Reported by: klaus3000 Patches: sip.conf.sample-patch.txt uploaded by klaus3000 (license 65) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@268990 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	55993d7b13	Merged revisions 254450 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r254450 | kpfleming | 2010-03-25 10:27:31 -0500 (Thu, 25 Mar 2010) | 49 lines Improve handling of T.38 re-INVITEs that arrive before a T.38-capable application is executing on a channel. This patch addresses an issue found during working with end-users using res_fax. If an incoming call is answered in the dialplan, or jumps to the 'fax' extension due to reception of a CNG tone (with faxdetect enabled), and then the remote endpoint sends a T.38 re-INVITE, it is possible for the channel's T.38 state to be 'T38_STATE_NEGOTIATING' when the application starts up. Unfortunately, even if the application wants to use T.38, it can't respond to the peer's negotiation request, because the AST_CONTROL_T38_PARAMETERS control frame that chan_sip sent originally has been lost, and the application needs the content of that frame to be able to formulate a reply. This patch adds a new 'request' type to AST_CONTROL_T38_PARAMETERS, AST_T38_REQUEST_PARMS. If the application sends this request, chan_sip will re-send the original control frame (with AST_T38_REQUEST_NEGOTIATE as the request type), and the application can respond as normal. If this occurs within the five second timeout in chan_sip, the automatic cancellation of the peer reinvite will be stopped, and the application will 'own' the negotiation process from that point onwards. This also improves the code path in chan_sip to allow sip_indicate(), when called for AST_CONTROL_T38_PARAMETERS, to be able to return a non-zero response, which should have been in place before since the control frame *can* fail to be processed properly. It also modifies ast_indicate() to return whatever result the channel driver returned for this control frame, rather than converting all non-zero results into '-1'. Finally, the new request type intentionally returns a positive value, so that an application that sends AST_T38_REQUEST_PARMS can know for certain whether the channel driver accepted it and will be replying with a control frame of its own, or whether it was ignored (if the sip_indicate()/ast_indicate() path had properly supported failure responses before, this would not be necessary). This patch also modifies res_fax to take advantage of the new request. In addition, this patch makes sip_t38_abort() actually lock the private structure before doing its work... bad programmer, no donut. This patch also enhances chan_sip's 'faxdetect' support to allow triggering on T.38 re-INVITEs received as well as CNG tone detection. Review: https://reviewboard.asterisk.org/r/556/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@260884 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	81245345dd	Merged revisions 255504 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@255505 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	f881ff7510	Merged revisions 255066 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r255066 | lmadsen | 2010-03-26 14:27:56 -0500 (Fri, 26 Mar 2010) | 6 lines Replace some documentation from 1.6.x back into trunk. This documentation associated wth tlsbindaddr is still useful so lets synchronize it between trunk and 1.6.x branches. (issue #17054) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@255067 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	fd23720c01	Merged revisions 255021 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@255023 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	6bfb1fca95	Merged revisions 252089 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@252137 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	07a05534e0	Merged revisions 249893 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@249895 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	a50f9f456f	Merged revisions 239520 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@239523 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tzafrir Cohen	c213189326	Merged revisions 238313 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@238349 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jared Smith	f27a27f0c8	Merged revisions 235298 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@235334 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Joshua Colp	fbbd80432b	Merged revisions 230881 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@230884 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	8a19549935	Merged revisions 226384 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@226385 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	25a24086a6	Merged revisions 225033 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@225035 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	ba54bb0506	Merged revisions 222110 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@222113 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Matthew Nicholson	5d6d3d7676	Merged revisions 221432 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@221477 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Terry Wilson	9a61ba3e28	Merged revisions 221266 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@221304 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Olle Johansson	9e57fd58e2	Merged revisions 216438 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@216647 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	8afb651407	Merged revisions 213098 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r213098 | tilghman | 2009-08-19 16:05:17 -0500 (Wed, 19 Aug 2009) | 9 lines Better parsing for the "register" line Allows characters that are otherwise used as delimiters to be used within certain fields (like the secret). (closes issue #15008, closes issue #15672) Reported by: tilghman Patches: 20090818__issue15008.diff.txt uploaded by tilghman (license 14) Tested by: lmadsen, tilghman ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@213117 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	bbf4a08fe7	Merged revisions 210190 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r210190 | kpfleming | 2009-08-03 15:48:48 -0500 (Mon, 03 Aug 2009) | 11 lines Rename 'canreinvite' option to 'directmedia', with backwards compatibility. It is clear from multiple mailing list, forum, wiki and other sorts of posts that users don't really understand the effects that the 'canreinvite' config option actually has, and that in some cases they think that setting it to 'no' will actually cause various other features (T.38, MOH, etc.) to not work properly, when in fact this is not the case. This patch changes the proper name of the option to what it should have been from the beginning ('directmedia'), but preserves backwards compatibility for existing configurations. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@210191 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	45d9ad47ca	Merged revisions 203699 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r203699 | file | 2009-06-26 16:27:24 -0300 (Fri, 26 Jun 2009) | 2 lines Improve T.38 negotiation by exchanging session parameters between application and channel. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@203705 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	2292d0e19b	Merged revisions 198791 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@198794 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	49cda05d53	Merged revisions 196416 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@196452 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	f381fb5421	Merged revisions 186060 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@186063 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Michiel van Baak	f1ae8e9f3b	Provide correct hint to debug SIP trouble in the default config ... (closes issue #14646) Reported by: strk git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@181499 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	3a14487abf	Allow for "magic" pickups to work when we wish to ignore the context ... When the subscription context for a call pickup subscription differs from the context of the call pickup target, there's not an easy way to divine what context should be used for the pickup. The way to work around this is to use PICKUPMARK as the context for the pickup. This has been documented in the sip.conf.sample file (ABE-1708) closes issue #14567 submitted by: alecdavis git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180155 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Olle Johansson	0685c4b281	Update documentation ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172270 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Olle Johansson	aca43d126a	Add some more notes about device matching. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@171880 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Olle Johansson	2c4f19eb2c	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: https://origsvn.digium.com/svn/asterisk/trunk@171838 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Olle Johansson	d4736e9897	Clarify some misunderstandings and make it even more clear that you can refer to a peer ... in the register= line. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168711 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	453b4cb8fb	Allow specifying a port number in the user portion of a register => line in sip.conf ... With this commit, a register => line in sip.conf may contain a port number in the "user" section of the line. Please see CHANGES and sip.conf.sample for more details regarding this. (closes issue #14198) Reported by: Nick_Lewis Patches: chan_sip.c-domainport2.patch uploaded by Nick (license 657) Tested by: Nick_Lewis git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168575 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Matthew Nicholson	91192e30c5	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165180 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	fd62012a31	Qualify trumps poke per lmadsen. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164814 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	92a4edc593	Add configuration options for finer control over how Asterisk handles having to poke all peers at seemingly the same time. ... (closes issue #13217) Reported by: cervajs git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164809 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Dwayne M. Hubbard	f9b6507796	If 'faxdetect=yes' in sip.conf, switch to a 'fax' extension (if it exists) after T38 is negotiated. ... Terry Wilson created the original patch for this functionality, which I slightly modified and added the faxdetect=yes|no configuration option. This patch is only for T38 fax detection and does not do anything for G711 over SIP fax detection. By default, this option is disabled. Reviewboard: http://reviewboard.digium.com/r/69/ This functionality is for issue AST-140. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@161115 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	7bd3ce358b	If you enabled 'notifycid' one of the limitations is that the calling channel ... is only found if it dialed the extension that was subscribed to. You can now specify 'ignore-context' for the 'notifycid' option in sip.conf which will, as it's value implies, ignore the current context of the caller when doing the lookup. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@158756 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	6ba4e7853e	Allow devices that accept dialog-info+xml (like snoms) to get the Caller ID of ... the calling party when subscribed to the state of an extension that is ringing. This has some limitations which are documented in sip.conf.sample. (closes issue #13827) Reported by: seanbright Patches: issue13827.patch uploaded by seanbright (license 71) Reviewed by: russellb git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@154187 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Olle Johansson	007807bf41	Updating docs ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@153983 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Olle Johansson	d3517de987	Spaces to replace tabs... ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@153905 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Olle Johansson	204845843e	Adding a separation of remote authentication and our authentication. ... remotesecret => our password for a remote service secret => our authentication when someone calls us Secret => still has both functions if remotesecret is not used. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@153904 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	0327f37d34	The default in chan_sip for notifyringing is yes, so update the sample ... conf to reflect that. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@153296 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	f6c78aa0fe	*whistle* ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@147761 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	cebd2c1df2	Add support for subscribing to a voice mailbox on a remote SIP server and making the new/old message count available to local devices. (issue #AST-77) ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@147760 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	aada13230f	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: https://origsvn.digium.com/svn/asterisk/trunk@142866 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	8b6dd2ad43	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: https://origsvn.digium.com/svn/asterisk/trunk@138260 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago