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path: root/configs/sip.conf.sample
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2007-02-14Make documentation match the source code. Olle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@54379 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-11Add support for outbound proxy for peers and [general]Olle Johansson
This replaces the older, broken, implementation where a setting in [general] did not do anything and the [peer] part was broken. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@53932 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-08rename busy-limit to busy-level, since it is not a limitKevin P. Fleming
actually parse the busy-limit option from sip.conf, instead of ignoring it git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@53577 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-02Patch based on this patch with small changes for trunk...Olle Johansson
Merged revisions 53109 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r53109 | oej | 2007-02-02 01:24:03 +0100 (Fri, 02 Feb 2007) | 4 lines Disable the direct p2p RTP call setup in SIP. You can enable it in sip.conf, but it is now considered experimental until we solve the AST_CONTROL_ANSWER with payload and videocaps stuff. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@53110 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-01Implementing "busy-limit".Olle Johansson
If you set call limit and busy limit, chan_sip will indicate BUSY for a device that has reached the busy limit and allow calls up to the call limit, allowing for call transfers (that generate a new call). If you only set call limit, chan_sip will not indicate BUSY until that limit is filled. This affects SIP subscriptions, call queues and manager applications. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@53082 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-01Merged revisions 53062 via svnmerge from Olle Johansson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r53062 | oej | 2007-02-01 17:35:12 +0100 (Thu, 01 Feb 2007) | 2 lines Add explanation of port= in combination with defaultip= (thanks jsmith) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@53063 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-12-31Added some docsOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@49081 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-12-27Be politically correctOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48986 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-12-27Add support for buggy Cisco MWI firmware > 8.0.3 (issue 8575 - flewid)Olle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48983 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-12-05Adding docs on t.38Olle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48269 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-12-02- Disable RTP timeouts during T.38 transmissionOlle Johansson
- Encapsulate RTP timers to the RTP structure, so we have one set for video and one for audio - Document RTP keepalive configuration option - Cleanup and document the monitor support function to hangup on RTP timeouts - Add RTP keepalive to SIP show settings Imported from 1.4 with modifications for trunk. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48200 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-12-01- Remove T.38 early media, since T.38 requires two way communication ↵Olle Johansson
(imported from 1.4) - Small fixes to limitonpeer git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48178 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-30Merged revisions 48143 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r48143 | file | 2006-11-30 12:57:35 -0500 (Thu, 30 Nov 2006) | 10 lines Merged revisions 48142 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48142 | file | 2006-11-30 12:55:23 -0500 (Thu, 30 Nov 2006) | 2 lines Document 'port' for SIP peers, came up because of the current mailing list thread. (issue #8450 reported by blitzrage) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48144 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-29Clarify some settings for status reports in subscriptions, queues and managerOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48114 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-29Explain RTP timeoutsOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48112 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-20Update docs for videosupportOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47846 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-16Make it possible to enable/disable onhold tracking, in order to make life easierOlle Johansson
for realtime users. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47756 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-16- CANCEL never uses authenticationOlle Johansson
- Add docs on canreinvite git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47734 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-04Adding new config option "limitpeersonly" to only apply call limitsOlle Johansson
to the peer side of a type=friend. This is for trying to support BJ in his quest to solve some issues with the queue system and type=friend objects. BJ: Please test! git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47201 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-10-31Fix rport handling.Olle Johansson
...where did the 1.2 properties come from, really? they're back. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46629 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-10-30Change name of "contact" setting to "callback" which better reflects what itOlle Johansson
is to the person that configures asterisk. That we use it internally in the contact header is a totally different story. Still not convinced this is a good option. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46489 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-10-26document the match_auth_username optionLuigi Rizzo
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46308 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-10-17Update of docsOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@45333 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-10-16In the course of a data this has been turned into an option to ignore ↵Joshua Colp
replies, then ignore responses and finally I'm just getting rid of the option altogether and making it the default no matter what. C'est la vie! git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@45286 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-10-16Merged revisions 45280 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r45280 | file | 2006-10-16 16:06:18 -0400 (Mon, 16 Oct 2006) | 10 lines Merged revisions 45265 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r45265 | file | 2006-10-16 15:59:54 -0400 (Mon, 16 Oct 2006) | 2 lines Use responses rather then replies even though they mean the same thing. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@45281 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-10-16Merged revisions 45262 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r45262 | file | 2006-10-16 15:37:34 -0400 (Mon, 16 Oct 2006) | 10 lines Merged revisions 45260 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r45260 | file | 2006-10-16 15:28:18 -0400 (Mon, 16 Oct 2006) | 2 lines Add 'ignoreoodreplies' option which will not create a pvt structure on a SIP response but instead basically drop it. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@45263 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-10-07Recommend using "sip reload" since it's much easier to learn andOlle Johansson
remember. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44707 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-10-06document a bit the use of templates.Luigi Rizzo
They are highly convenient for writing configuration files, especially if you have many similar entries, or want to switch quickly between different configurations without having to comment/uncomment large sections of the files. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44579 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-10-06document the "contact" option a bit better.Luigi Rizzo
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44578 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-10-06Two things:Luigi Rizzo
1. slightly rearrange/simplify the parsing of the argument in sip_register. This brings in a patch that has been in Mantis (5834) for ages, and is the larger part of the commit; 2. implement the "contact" option for peers, similar to the one in users.conf: If you put a "contact" option with a non-empty argument (e.g. contact=123) in a peer section, asterisk will register with the provider as if you had a register= username:secret@host/contact line in the general section. The latter is a very small is a new feature so i am not putting it in the 1.4 branch, although the "contact" option in user.conf is already in the 1.4 branch and so it wouldn't be too strange to merge it. Note that the implementation of "contact" is much simpler than the one in 5834, and limited to a few lines in build_peer(). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44566 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-10-06update example commands to match current syntaxLuigi Rizzo
(does not apply to 1.4) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44537 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-09-20Add documentation on rtp packetization.Jason Parker
Technically, ooh323 doesn't support it yet, but there is a patch that should be committed very soon. Issue #7989, patch by DEA, slightly modified. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43344 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-09-11Merged revisions 42716 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r42716 | tilghman | 2006-09-11 11:39:06 -0500 (Mon, 11 Sep 2006) | 2 lines Spelling/grammar fixes (Issue 7929) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@42717 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-08-31Merge in VLDTMF support with Zaptel/Core done by the ever great Darumkilla ↵Joshua Colp
Russell Bryant and the RTP portion done by myself, Muffinlicious Joshua Colp. This has gone through so many discussions/revisions it's not funny but we finally have it! git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@41507 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-07-19merge Russell's 'hold_handling' branch, finally implementing music-on-hold ↵Kevin P. Fleming
handling the way it was decided at AstriDevCon Europe 2006 (and the way people really want it to be) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37988 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-07-13actually make the non-standard G726-32 behavior available for SIP clientsKevin P. Fleming
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37564 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-07-10Remove configuration option "restrictcid" that is nowhere toOlle Johansson
be seen in the code. Did it exist, was it planned to exist or was it documentationware only? Ask Dr Asterisk. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37324 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-07-02- Make use of system name in realtime SIP peers optionalOlle Johansson
- Fix small issue with SIP history git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@36602 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-06-30Removing configuration options that does not do anything yet. No need toOlle Johansson
add "promises" to the sip.conf.sample... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@36355 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-06-29Merged revisions 36253-36254 via svnmerge from Kevin P. Fleming
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r36253 | kpfleming | 2006-06-29 02:19:27 -0500 (Thu, 29 Jun 2006) | 2 lines add documentation for peer-specific 'outboundproxy' setting ........ r36254 | kpfleming | 2006-06-29 02:19:54 -0500 (Thu, 29 Jun 2006) | 2 lines clarify documentation for 'persistentmembers' setting ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@36262 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-06-29reformatting sip.conf.sample a bit, adding dumphistory that was not documentedOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@36251 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-06-26Speling error. Avoid swenglish :-) (thanks, jtodd!)Olle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@36109 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-06-26Add example of permit/deny to sip.conf.sampleOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@36054 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-06-13Allow AST_FRAME_MODEM frames to be dumped, and document T.38 passthrough supportJoshua Colp
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@33912 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-06-01- add the ability to configure forced jitterbuffers on h323, jingle,Russell Bryant
and mgcp channels - remove the jitterbuffer configuration from the pvt structures in the sip, zap, and skinny channel drivers, as copying the same global configuration into each pvt structure has no benefit. - update and fix some typos in jitterbuffer related documentation (issue #7257, north, with additional updates and modifications) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@31413 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-06-01Merged revisions 31321 via svnmerge from Kevin P. Fleming
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r31321 | kpfleming | 2006-06-01 07:41:47 -0500 (Thu, 01 Jun 2006) | 2 lines remove a sample entry that never should have been added (code to support it was not merged) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@31322 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-31Add support for using a jitterbuffer for RTP on bridged calls. This includesRussell Bryant
a new implementation of a fixed size jitterbuffer, as well as support for the existing adaptive jitterbuffer implementation. (issue #3854, Slav Klenov) Thank you very much to Slav Klenov of Securax and all of the people involved in the testing of this feature for all of your hard work! git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@31052 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-24add a new option for 'obscuring' SIP user/peer names from fishersKevin P. Fleming
use an enum for authentication results and clean up code fix a bug where SUBSCRIBE for an unknown user/peer would not generate a response git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@29903 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-18add another media path reinvite 'flavor', where we will only redirect our ↵Kevin P. Fleming
media to devices that we know are not behind a NAT (based on the evidence collected when we receive media from them) also, documented the 'canreinvite=update' option in the sample config file git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@28215 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-18Allow contexts in regexten so that extensions can be added to multiple ↵Joshua Colp
contexts when peer registers (issue #6869 reported by and created by Marquis) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@28168 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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