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path: root/configs/sip.conf.sample
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2009-04-29SIP option to specify outbound TLS/SSL client protocol.David Vossel
chan_sip allows for outbound TLS connections, but does not allow the user to specify what protocol to use (default was SSLv2, and still is if this new option is not specified). This patch lets the user pick the SSL/TLS client method for outbound connections in sip. (closes issue #14770) Reported by: TheOldSaint (closes issue #14768) Reported by: TheOldSaint Review: http://reviewboard.digium.com/r/240/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191177 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-27Remove nonexistent option from sip.conf.sample.Mark Michelson
The option to choose which connected line header to use is not 'rpid_header' but 'sendrpid' git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190577 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-24TLS/SSL private key optionDavid Vossel
Adds option to specify a private key .pem file when configuring TLS or SSL in AMI, HTTP, and SIP. Before this, the certificate file was used for both the public and private key. It is possible for this file to hold both, but most configurations allow for a separate private key file to be specified. Clarified in .conf files how these options are to be used. The current conf files do not explain how the private key is handled at all, so without knowledge of Asterisk's TLS implementation, it would be hard to know for sure what was going on or how to set it up. Review: http://reviewboard.digium.com/r/234/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190545 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-09Add a new option, mwi_from, to sip.conf.Mark Michelson
This allows for you to change the From header for outgoing MWI NOTIFY requests. Prior to this, the best you could do was to set a callerid in the general section of sip.conf. The problem was that this was used for all outbound requests, not just MWI NOTIFY requests. AST-201 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187560 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-03This commit introduces COLP/CONP and Redirecting party information into ↵Mark Michelson
Asterisk. The channel drivers which have been most heavily tested with these enhancements are chan_sip and chan_misdn. Further work is being done to add Q.SIG support and will be introduced in a later commit. chan_skinny has code added to it here, but according to user pj, the support on chan_skinny is not working as of now. This will be fixed in a later commit. A special thanks goes out to bugtracker user gareth for getting the ball rolling and providing the initial support for this work. Without his initial work on this, this would not have been nearly as painless as it was. This functionality has been tested by Digium's product quality department, as well as a customer site running thousands of calls every day. In addition, many many many many bugtracker users have tested this, too. (closes issue #8824) Reported by: gareth Review: http://reviewboard.digium.com/r/201 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186525 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-02Merge in the RTP engine API.Joshua Colp
This API provides a generic way for multiple RTP stacks to be integrated into Asterisk. Right now there is only one present, res_rtp_asterisk, which is the existing Asterisk RTP stack. Functionality wise this commit performs the same as previously. API documentation can be viewed in the rtp_engine.h header file. Review: http://reviewboard.digium.com/r/209/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186078 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-02Merged revisions 186059 via svnmerge from Tilghman Lesher
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/trunk@186060 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-24SIP preferred codec only featureDavid Vossel
Added an option to respond to a SIP invite with only the single most preferred joint codec. This limits the options of what codecs the other side can use. (closes issue #12485) Reported by: bamby Review: http://reviewboard.digium.com/r/206/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@183995 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-11Provide correct hint to debug SIP trouble in the default configMichiel van Baak
(closes issue #14646) Reported by: strk git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@181499 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-04Allow for "magic" pickups to work when we wish to ignore the contextMark Michelson
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
2009-01-29Update documentationOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172270 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-28Add some more notes about device matching.Olle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@171880 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-28Merged revisions 171837 via svnmerge from Olle Johansson
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
2009-01-15Clarify some misunderstandings and make it even more clear that you can ↵Olle Johansson
refer to a peer in the register= line. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168711 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-13Allow specifying a port number in the user portion of a register => line in ↵Mark Michelson
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
2008-12-17This patch adds a new 'ignoresdpversion' option to sip.conf. When this isMatthew Nicholson
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
2008-12-16Qualify trumps poke per lmadsen.Joshua Colp
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164814 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-16Add configuration options for finer control over how Asterisk handles having ↵Joshua Colp
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
2008-12-04If 'faxdetect=yes' in sip.conf, switch to a 'fax' extension (if it exists) ↵Dwayne M. Hubbard
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
2008-11-23If you enabled 'notifycid' one of the limitations is that the calling channelSean Bright
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
2008-11-04Allow devices that accept dialog-info+xml (like snoms) to get the Caller ID ofSean Bright
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
2008-11-03Updating docsOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@153983 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-03Spaces to replace tabs...Olle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@153905 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-03Adding a separation of remote authentication and our authentication.Olle Johansson
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
2008-11-01The default in chan_sip for notifyringing is yes, so update the sampleSean Bright
conf to reflect that. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@153296 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-09*whistle*Joshua Colp
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@147761 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-09Add support for subscribing to a voice mailbox on a remote SIP server and ↵Joshua Colp
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
2008-09-12Merged revisions 142865 via svnmerge from Tilghman Lesher
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
2008-08-15Merged revisions 138258 via svnmerge from Tilghman Lesher
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
2008-08-14Merged revisions 137731 via svnmerge from Russell Bryant
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: https://origsvn.digium.com/svn/asterisk/trunk@137732 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-08-01SIP should use the transport type set in the Moved Temporarily for the nextTilghman Lesher
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: https://origsvn.digium.com/svn/asterisk/trunk@135126 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-30Move implementation of an attended-transfer-complete sound from one channelTilghman Lesher
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: https://origsvn.digium.com/svn/asterisk/trunk@134401 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-15Additional option for videosupport (always) that disables the optimization toTilghman Lesher
fail to setup video RTP if the two endpoints will not support it. This assists with call files and certain transfers to ensure that if two video phones are ever connected, they will always share a video feed. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@130951 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-06- Adding alias "udpbindaddr" for the UDP port to comply with "tcpbindaddr" ↵Olle Johansson
and "tlsbindaddr". Note: I don't think we can start properly without UDP port open, that needs to be tested. - Removing "bindport" from configuration example, not needed to mention this any more I suggest we deprecate "bindaddr" and "bindport" in trunk (for 1.6.1) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@128525 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-06- Fixing issues with "sip show settings"Olle Johansson
- Adding IP address for TCP and/or TLS too if auto-domain is enabled and binding to a different IP address - Fixing documentation in sip.conf.sample git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@128524 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-05Make TCP disabled by default (it's considered experimental)Olle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@128237 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-05Reformatting the config sampleOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@128236 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-01Add a configuration option so the global outboundproxy can use tcptls ↵Brett Bryant
without it being defined by each sip user. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@127154 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-01Merged revisions 126844 via svnmerge from Olle Johansson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r126844 | oej | 2008-07-01 14:53:01 +0200 (Tis, 01 Jul 2008) | 5 lines Clear up documentation on "domain=" setting in sip.conf Reported by: davidw (closes issue #12413) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@126845 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-06-27Change the way that the transport option works for sip users. transport will ↵Brett Bryant
now take multiple arguments, the first one listed will be the one used for new dialogs, and the rest listed will be acceptable ways for that peer to contact us. This fixes a minor bug where, because SIP TCP/UDP run on the same port, could cause a TCP peer to be saved in the ast_db. There will also be warnings when a transport is changed for an unexpected reason. (issue #12799) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@125891 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-06-19Merged revisions 123883 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r123883 | tilghman | 2008-06-19 11:20:41 -0500 (Thu, 19 Jun 2008) | 4 lines Correct description of notifyringing option. (Closes issue #12890) Reported by gminet ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@123887 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-05-28Merged revisions 118646 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r118646 | file | 2008-05-28 11:23:34 -0300 (Wed, 28 May 2008) | 4 lines Add an option to use the source IP address of RTP as the destination IP address of UDPTL when a specific option is enabled. If the remote side is properly configured (ports forwarded) then UDPTL will flow. (closes issue #10417) Reported by: cstadlmann ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@118647 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-30Add support for specifying the registration expiry on a per registration ↵Joshua Colp
basis in the register line. This comes from a Switchvox patch. (issue AST-24) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114912 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-21(closes issue #6113)Jeff Peeler
Reported by: oej Tested by: jpeeler This patch implements multiple parking lots for parked calls. The default parkinglot is used by default, however setting the channel variable PARKINGLOT in the dialplan will allow use of any other configured parkinglot. See configs/features.conf.sample for more details on setting up another non-default parkinglot. Also, one can (currently) set the default parkinglot to use in the driver configuration file via the parkinglot option. Patch initially written by oej, brought up to date and finalized by mvanbaak, and then stabilized and converted to astobj2 by me. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114487 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-16This is the scariest commit I've done in a long time. This is the ↵Steve Murphy
astobj2-ification of chan_sip. I've tested a number of scenarios like crazy. It used to have 4x the call setup/teardown performance of trunk, but now it's roughly at parity. I will attempt to find the bottlenecks and get it back to the 4x mark. The changes made were somewhat invasive, but the value to the community of these upgrades outweighs waiting further for more testing. Every change being made to chan_sip was lousing this code up when we tried to merge. Peers, Users, Dialogs, are all now astobj2 objects, indexed via hashtables. Refcounting is used to track objects and free them at the bitter end of their lives. Please file issues on bugs.digium.com, and PLEASE, please, please be patient. One natural advantage to all the hash-table work is that loading large sip.conf files full of thousands of peers now goes much faster. One more please: PLEASE help thrash this code and test it. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114190 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-25Add a special dialplan variable to chan_sip which will cause an audio file ↵Joshua Colp
to be played upon completion of an attended transfer. (closes issue #9239) Reported by: sunder git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110631 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-21Note that the TCP and TLS support is currently considered experimental andRussell Bryant
is subject to change while we work out the remaining issues. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110499 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-18Add manager peerstatus events when peer can't authenticate.Olle Johansson
(closes issue #11959) Reported by: mostyn Patches: peerstatus3.patch uploaded by mostyn (license 398) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109316 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-02-29Add documentation for setting username/password in SIP dial string.Joshua Colp
(closes issue #11587) Reported by: sobomax Patches: dialstring_doc.diff uploaded by sobomax (license 359) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105378 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-02-25Adding more tls configuration details to sip.conf sample, with a list of ↵Brett Bryant
valid ciphers provided in both files. .. First commit since July, woot git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@104088 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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