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2010-07-10Merged revisions 245192 via svnmerge from russell1-14/+0
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: http://svn.digium.com/svn/asterisk/branches/1.6.2@275469 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-06Merged revisions 274284 via svnmerge from twilson1-0/+3
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: http://svn.digium.com/svn/asterisk/branches/1.6.2@274360 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-06Merged revisions 274316 via svnmerge from jpeeler1-6/+8
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: http://svn.digium.com/svn/asterisk/branches/1.6.2@274347 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-08Merged revisions 268988 via svnmerge from lmadsen1-4/+3
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: http://svn.digium.com/svn/asterisk/branches/1.6.2@268990 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-04Merged revisions 254450 via svnmerge from mnicholson1-3/+7
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: http://svn.digium.com/svn/asterisk/branches/1.6.2@260884 f38db490-d61c-443f-a65b-d21fe96a405b
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/1.6.2@255505 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-26Merged revisions 255066 via svnmerge from lmadsen1-0/+2
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: http://svn.digium.com/svn/asterisk/branches/1.6.2@255067 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/1.6.2@255023 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-13Merged revisions 252089 via svnmerge from twilson1-3/+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/1.6.2@252137 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/1.6.2@249895 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/1.6.2@239523 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/1.6.2@238349 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/1.6.2@235334 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/1.6.2@230884 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/1.6.2@226385 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/1.6.2@225035 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-05Merged revisions 222110 via svnmerge from kpfleming1-9/+18
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/1.6.2@222113 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/1.6.2@221477 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-30Merged revisions 221266 via svnmerge from twilson1-0/+3
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/1.6.2@221304 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/1.6.2@216647 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-19Merged revisions 213098 via svnmerge from tilghman1-1/+1
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: http://svn.digium.com/svn/asterisk/branches/1.6.2@213117 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-03Merged revisions 210190 via svnmerge from kpfleming1-16/+16
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: http://svn.digium.com/svn/asterisk/branches/1.6.2@210191 f38db490-d61c-443f-a65b-d21fe96a405b
2009-06-26Merged revisions 203699 via svnmerge from file1-2/+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/1.6.2@203705 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/1.6.2@198794 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/1.6.2@196452 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/1.6.2@186063 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-11Provide correct hint to debug SIP trouble in the default configmvanbaak1-1/+1
(closes issue #14646) Reported by: strk git-svn-id: http://svn.digium.com/svn/asterisk/trunk@181499 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-04Allow for "magic" pickups to work when we wish to ignore the contextmmichelson1-1/+4
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) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@180155 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-29Update documentationoej1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@172270 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-28Add some more notes about device matching.oej1-1/+13
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@171880 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-28Merged revisions 171837 via svnmerge from oej1-0/+19
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/trunk@171838 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-15Clarify some misunderstandings and make it even more clear that you can ↵oej1-4/+8
refer to a peer in the register= line. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@168711 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-13Allow specifying a port number in the user portion of a register => line in ↵mmichelson1-0/+13
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: http://svn.digium.com/svn/asterisk/trunk@168575 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-17This patch adds a new 'ignoresdpversion' option to sip.conf. When this ismnicholson1-0/+10
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: http://svn.digium.com/svn/asterisk/trunk@165180 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-16Qualify trumps poke per lmadsen.file1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@164814 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-16Add configuration options for finer control over how Asterisk handles having ↵file1-0/+2
to poke all peers at seemingly the same time. (closes issue #13217) Reported by: cervajs git-svn-id: http://svn.digium.com/svn/asterisk/trunk@164809 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-04If 'faxdetect=yes' in sip.conf, switch to a 'fax' extension (if it exists) ↵dhubbard1-0/+5
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: http://svn.digium.com/svn/asterisk/trunk@161115 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-23If you enabled 'notifycid' one of the limitations is that the calling channelseanbright1-1/+3
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: http://svn.digium.com/svn/asterisk/trunk@158756 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-04Allow devices that accept dialog-info+xml (like snoms) to get the Caller ID ofseanbright1-0/+10
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: http://svn.digium.com/svn/asterisk/trunk@154187 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-03Updating docsoej1-96/+87
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@153983 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-03Spaces to replace tabs...oej1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@153905 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-03Adding a separation of remote authentication and our authentication.oej1-2/+4
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: http://svn.digium.com/svn/asterisk/trunk@153904 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-01The default in chan_sip for notifyringing is yes, so update the sampleseanbright1-2/+2
conf to reflect that. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@153296 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-09*whistle*file1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@147761 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-09Add support for subscribing to a voice mailbox on a remote SIP server and ↵file1-2/+13
making the new/old message count available to local devices. (issue #AST-77) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@147760 f38db490-d61c-443f-a65b-d21fe96a405b
2008-09-12Merged revisions 142865 via svnmerge from tilghman1-0/+14
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/trunk@142866 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-15Merged revisions 138258 via svnmerge from tilghman1-1/+3
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/trunk@138260 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-14Merged revisions 137731 via svnmerge from russell1-435/+435
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/trunk@137732 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
generated by cgit v1.2.3 (git 2.25.1) at 2024-06-18 14:21:11 +0000