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path: root/configs/sip.conf.sample
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2011-02-02Replacing doc/* and asterisk.pdf with wiki linksAndrew Latham
Adding links to http(s)://wiki.asterisk.org git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@305843 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-02Replacing doc/* with wiki linksAndrew Latham
Adding links to http(s)://wiki.asterisk.org git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@305799 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-01SIP Configuration DocumentationAndrew Latham
sip show settings reports qualifyfreq in milliseconds. sip.conf configures qualifyfreg in seconds. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@305650 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-31Merged revisions 305247 via svnmerge from Jason Parker
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r305247 | qwell | 2011-01-31 16:25:23 -0600 (Mon, 31 Jan 2011) | 7 lines Add alternative name for config option. The SIP sample configuration had "tlscadir" as the option name, but chan_sip used the more correct "tlscapath". Now both are accepted. Discovered (sort of) by a user on IRC in #asterisk ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@305248 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-17Merged revisions 302005 via svnmerge from Terry Wilson
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r302005 | twilson | 2011-01-17 09:04:59 -0600 (Mon, 17 Jan 2011) | 2 lines Document "encryption" option in sip.conf.sample ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@302006 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-12-17Merged revisions 298773 via svnmerge from Brad Watkins
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r298773 | marquis | 2010-12-17 12:26:31 -0500 (Fri, 17 Dec 2010) | 10 lines Fix parsing of mwi => lines in sip.conf Reworking parsing of mwi => lines to resolve a segfault. Also add a set of unit tests for the function that does the parsing. (closes issue #18350) Reported by: gbour Tested by: Marquis, gbour Review: https://reviewboard.asterisk.org/r/1053/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@298774 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-03Merged revisions 285006 via svnmerge from David Vossel
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r285006 | dvossel | 2010-09-03 17:21:50 -0500 (Fri, 03 Sep 2010) | 9 lines Disables auth_options_request option by default. The auth_options_request option was created to do authentication on OPTIONS request just like INVITES are done. Since it has been noted that some endpoints use OPTIONS requests as a way of qualifying a peer and that a 401 authentication response could result in interoperability issues, this option has been disabled by default. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@285007 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-03Merged revisions 284950 via svnmerge from David Vossel
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r284950 | dvossel | 2010-09-03 12:29:02 -0500 (Fri, 03 Sep 2010) | 14 lines authenticate OPTIONS requests just like we would an INVITE OPTIONS requests should be treated the same as an INVITE This includes authentication. This patch adds the ability for incoming out of dialog OPTION requests to be authenticated before providing a response indicating whether an extension is available or not. The authentication routine works the exact same way as it does for incoming INVITEs. This means that if a peer has 'insecure=invite' in their peer definition, the same will be true for the processing of the OPTIONS request. Review: https://reviewboard.asterisk.org/r/881/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@284951 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-08-24Merged revisions 283493 via svnmerge from David Vossel
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r283493 | dvossel | 2010-08-24 15:34:03 -0500 (Tue, 24 Aug 2010) | 2 lines Changes the default behavior for sip.conf's pedantic option from "no" to "yes". ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@283494 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-08-19Merged revisions 282740 via svnmerge from Terry Wilson
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r282740 | twilson | 2010-08-18 21:18:50 -0500 (Wed, 18 Aug 2010) | 16 lines Merged revisions 282730 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r282730 | twilson | 2010-08-18 21:14:28 -0500 (Wed, 18 Aug 2010) | 9 lines 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/trunk@282751 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-08-13Merged revisions 282302 via svnmerge from David Vossel
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r282302 | dvossel | 2010-08-13 17:23:38 -0500 (Fri, 13 Aug 2010) | 10 lines remove current STUN support from chan_sip.c This patch removes the current broken/useless stun support from chan_sip. (closes issue #17622) Reported by: philipp2 Review: https://reviewboard.asterisk.org/r/855/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@282304 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-08-13Merged revisions 282269 via svnmerge from David Vossel
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r282269 | dvossel | 2010-08-13 15:03:56 -0500 (Fri, 13 Aug 2010) | 4 lines res_stun_monitor for monitoring network changes behind a NAT device Review: https://reviewboard.asterisk.org/r/854 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@282270 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-08-11Merged revisions 281687 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r281687 | simon.perreault | 2010-08-11 09:30:59 -0400 (Wed, 11 Aug 2010) | 9 lines Fix parsing of IPv6 address literals in outboundproxy (closes issue #17757) Reported by: oej Patches: 17757.diff uploaded by sperreault (license 252) sip.conf.diff uploaded by sperreault (license 252) Tested by: oej ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@281688 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-08-10Merged revisions 281650 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r281650 | russell | 2010-08-10 16:47:31 -0500 (Tue, 10 Aug 2010) | 5 lines Change the default value for alwaysauthreject in sip.conf to "yes". (closes issue #17756) Reported by: oej ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@281651 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-08-09Merged revisions 281356 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r281356 | simon.perreault | 2010-08-09 10:31:40 -0400 (Mon, 09 Aug 2010) | 2 lines Added comment about IPv4-mapped IPv6 addresses and the output of netstat. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@281357 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-08-03Merged revisions 280777 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r280777 | simon.perreault | 2010-08-03 15:53:07 -0400 (Tue, 03 Aug 2010) | 8 lines Better documentation related to IPv6. (closes issue #17737) Reported by: oej Patches: doc.diff uploaded by sperreault (license 252) Tested by: mmichelson ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@280779 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-08-03Reverted r280706 and r280707. Will commit in branch 1.8 and merge to trunk ↵
properly. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@280745 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-08-03Better documentation related to IPv6.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@280706 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-19Fix port setting of external address in SIP.Mark Michelson
There are two changes here: 1. Since the externip setting can now have a port attached to it, calling it "externip" is misleading. The option is now documented and parsed as "externaddr." This also extends to the "matchexterniplocally" setting. It is now documented and parsed as "matchexternaddrlocally." The old names for the options may still be used, but they are no longer used in the sip.conf.sample file. 2. If no port is set for the externaddr, and UDP is the transport to be used, then we will set the port of the externaddr to that of the udpbindaddr. This was how things worked prior to the IPv6 merge, so this is a regression fix. (closes issue #17665) Reported by: mmichelson Patches: 17665.diff#2 uploaded by pprindeville (license 347) Tested by: pprindeville git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@277873 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-19Make ACLs IPv6-capable.Mark Michelson
ACLs can now be configured to match IPv6 networks. This is only relevant for ACLs in chan_sip for now since other channel drivers do not support IPv6 addressing. However, once those channel drivers are outfitted to support IPv6 addressing, the ACLs will already be ready for IPv6 support. https://reviewboard.asterisk.org/r/791 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@277814 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-16Clarify syntax changesOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@277028 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-16Add ability to configure the Max-Forwards header in the dialplan, as well as inOlle Johansson
sip.conf configuration for the channel and for devices. The Max-Forwards header is used to prevent loops in a SIP network. Each intermediary, like SIP proxys and SBCs, decrement this counter and detects when it reaches zero, at which point the SIP request is nicely killed in a SIP-friendly way. Review: https://reviewboard.asterisk.org/r/778/ Thanks to dvossel for the review and good advice. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276951 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-08Add IPv6 to Asterisk.Mark Michelson
This adds a generic API for accommodating IPv6 and IPv4 addresses within Asterisk. While many files have been updated to make use of the API, chan_sip and the RTP code are the files which actually support IPv6 addresses at the time of this commit. The way has been paved for easier upgrading for other files in the near future, though. Big thanks go to Simon Perrault, Marc Blanchet, and Jean-Philippe Dionne for their hard work on this. (closes issue #17565) Reported by: russell Patches: asteriskv6-test-report.pdf uploaded by russell (license 2) Review: https://reviewboard.asterisk.org/r/743 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274783 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-06Merged revisions 274283 via svnmerge from Jeff Peeler
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/trunk@274316 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-08Update note in sip.conf.sample.Leif Madsen
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/trunk@268988 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-02Generic Advice of Charge.Richard Mudgett
Asterisk Generic AOC Representation - Generic AOC encode/decode routines. (Generic AOC must be encoded to be passed on the wire in the AST_CONTROL_AOC frame) - AST_CONTROL_AOC frame type to represent generic encoded AOC data - Manager events for AOC-S, AOC-D, and AOC-E messages Asterisk App Support - app_dial AOC-S pass-through support on call setup - app_queue AOC-S pass-through support on call setup AOC Unit Tests - AOC Unit Tests for encode/decode routines - AOC Unit Test for manager event representation. SIP AOC Support - Pass-through of generic AOC-D and AOC-E messages to snom phones via the snom AOC specification. - Creation of chan_sip page3 flags for the addition of the new 'snom_aoc_enabled' sip.conf option. IAX AOC Support - Natively supports AOC pass-through through the use of the new AST_CONTROL_AOC frame type DAHDI AOC Support - ETSI PRI full AOC Pass-through support - 'aoc_enable' chan_dahdi.conf option for independently enabling pass-through of AOC-S, AOC-D, AOC-E. - 'aoce_delayhangup' option for retrieving AOC-E on disconnect. - DAHDI A() dial string option for requesting AOC services. example usage: ;requests AOC-S, AOC-D, and AOC-E on call setup exten=>1111,1,Dial(DAHDI/g1/1112/A(s,d,e)) Review: https://reviewboard.asterisk.org/r/552/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267096 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-20Add support for direct media ACLsTerry Wilson
directmediapermit/directmediadeny support to restrict which peers can do directmedia based on ip address. In some networks not all phones are fully routed, i.e. not all phones can ping each other. This patch adds a way to restrict directmedia for certain peers between certain networks. (closes issue #16645) Reported by: raarts Patches: directmediapermit.patch uploaded by raarts (license 937) Tested by: raarts Review: https://reviewboard.asterisk.org/r/467/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264626 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-04-13Update sample dialstrings in sip.conf.sample file.Mark Michelson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@257032 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-04-01Removed documentation of the non existent 'both' option to 'faxdetect' in ↵Matthew Nicholson
sip.conf git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@255751 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-31Add documentation clarifying when 't' and 'T' can be used.Leif Madsen
(closes issue #17021) Reported by: kovzol Tested by: lmadsen, kovzol, davidw, ebroad git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@255504 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-26Replace some documentation from 1.6.x back into trunk.Leif Madsen
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/trunk@255066 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-26Update confusing documentation for tlsbindaddr.Leif Madsen
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/trunk@255021 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-25Improve handling of T.38 re-INVITEs that arrive before a T.38-capableKevin P. Fleming
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/trunk@254450 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-12Only change the RTP ssrc when we see that it has changedTerry Wilson
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/trunk@252089 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-02fixes adaptive jitterbuffer configurationDavid Vossel
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/trunk@249893 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-02-06Remove useless sip options related to hash table size.Mark Michelson
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. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@245192 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-01-15Clarify RTP NAT handling a bit.Kevin P. Fleming
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@240328 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-01-12Note that direct T.38 is not supported.Leif Madsen
(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/trunk@239520 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-01-07Document the usefulness of explicit udp:// in the register stringTzafrir Cohen
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@238313 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-16Merged revisions 235181 via svnmerge from Jared Smith
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/trunk@235298 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-23Change fax detection in chan_sip so it behaves as one would expect.Joshua Colp
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/trunk@230881 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-12Update sip.conf.sample.Leif Madsen
Just updating a spelling error and some capitalization in a documentation update that Olle added. May the Swenglish be with you. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@229639 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-12ClarificationOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@229607 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-12Clarify some security issues early in the sample configurationOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@229606 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-02This patch adds support for a draft proposal for adding Q.850 reason headers ↵Matthew Nicholson
to sip messages. (closes issue #13385) Reported by: adomjan Patches: sip.conf.sample-trunk20090929-reason_q850.patch uploaded by adomjan (license 487) CHANGES-trunk20090929-reason_q850.patch uploaded by adomjan (license 487) chan_sip.c-trunk20090929-reason_q850_atoi_fix.patch uploaded by adomjan (license 487) sip-q850-hangupcause1.diff uploaded by mnicholson (license 96) Tested by: adomjan git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226687 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-28Merged revisions 226382 via svnmerge from Leif Madsen
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/trunk@226384 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-27Add support for receiving unsolicited MWI NOTIFY messages.Joshua Colp
This change adds a configuration option to SIP peers, unsolicited_mailbox, which configures a virtual mailbox to use for received new/old MWI information. This virtual mailbox can then be used by any device supporting MWI. (closes issue #13028) Reported by: AsteriskRocks Patches: bug_13028_chan_sip_external_mwi_20090707.patch uploaded by cmaj (license 830) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226060 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-21Add support for specifying the IP address to use for media streams in sip.confJoshua Colp
This is the second commit for this and documents the text stream using the configured IP address and fixes a bug in the original patch where the UDPTL stream would also use the different IP address. (closes issue #14729) Reported by: _brent_ Patches: media_address.patch uploaded by brent (license 388) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225089 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-21Revert media_address commit, I'm going to roll a fix to the SDP generation ↵Joshua Colp
in the next version. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225034 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-21Merged revisions 225032 via svnmerge from David Vossel
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r225032 | dvossel | 2009-10-21 09:37:04 -0500 (Wed, 21 Oct 2009) | 20 lines IAX/SIP shrinkcallerid option The shrinking of caller id removes '(', ' ', ')', non-trailing '.', and '-' from the string. This means values such as 555.5555 and test-test result in 555555 and testtest. There are instances, such as Skype integration, where a specific value is passed via caller id that must be preserved unmodified. This patch makes the shrinking of caller id optional in chan_sip and chan_iax in order to support such cases. By default this option is on to preserve previous expected behavior. (closes issue #15940) Reported by: dimas Patches: v2-15940.patch uploaded by dimas (license 88) 15940_shrinkcallerid_trunk.c uploaded by dvossel (license 671) Tested by: dvossel Review: https://reviewboard.asterisk.org/r/408/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225033 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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