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2009-10-28Update documentation in sip.conf.sample.lmadsen1-0/+7
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/1.4@226382 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-21IAX/SIP shrinkcallerid optiondvossel1-0/+10
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/1.4@225032 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-30Fix SRV lookup and Request-URI generation in chan_sip.mnicholson1-0/+3
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/1.4@221360 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-30Change the SSRC by default when our media stream changestwilson1-1/+4
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/1.4@221086 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-04Make apps send PROGRESS control frame for early media and fix too early ↵oej1-0/+6
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/1.4@216430 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-02Merged revisions 186056 via svnmerge from tilghman1-2/+4
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/1.4@186059 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-28Add a better explanation of the difference between the device namespace and ↵oej1-0/+19
the dialplan for newbies. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@171837 f38db490-d61c-443f-a65b-d21fe96a405b
2008-09-12Create rules for disallowing contacts at certain addresses, which maytilghman1-0/+14
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/branches/1.4@142865 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-15More fixes for realtime peers.tilghman1-1/+3
(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/branches/1.4@138258 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-14Comments in this config file were aligned only if your tab size was set to 8.russell1-334/+334
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/branches/1.4@137731 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-01Clear up documentation on "domain=" setting in sip.confoej1-4/+3
Reported by: davidw (closes issue #12413) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@126844 f38db490-d61c-443f-a65b-d21fe96a405b
2008-06-19Correct description of notifyringing option.tilghman1-1/+2
(Closes issue #12890) Reported by gminet git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@123883 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-28Add an option to use the source IP address of RTP as the destination IP ↵file1-1/+6
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: http://svn.digium.com/svn/asterisk/branches/1.4@118646 f38db490-d61c-443f-a65b-d21fe96a405b
2008-04-25Reference documentation files that actually exist.tilghman1-2/+2
(closes issue #12516) Reported by: linuxmaniac Patches: diff_rev114611.patch uploaded by linuxmaniac (license 472) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@114649 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-27Clarify limitonpeers=yesoej1-0/+3
(closes issue #11304) Reported by: pj git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89624 f38db490-d61c-443f-a65b-d21fe96a405b
2007-09-18Correct the allowexternaldomains option in SIP sample config.qwell1-1/+1
Issue 10753 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@82751 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-08(closes issue #10335)file1-1/+2
Reported by: adamgundy Update sip.conf to include another scenario where directrtpsetup will fail. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@78569 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-12Add matchexterniplocally setting which only substitutes your ↵file1-0/+4
externip/externhost setting if it matches the localnet setting. I know of at least two people who need opposite settings, so I made it an option! (issue #8821 reported by kokoskarokoska) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@58779 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-02Disable the direct p2p RTP call setup in SIP. You can enable it in sip.conf, ↵oej1-0/+6
but it is now considered experimental until we solve the AST_CONTROL_ANSWER with payload and videocaps stuff. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@53109 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-01Add explanation of port= in combination with defaultip= (thanks jsmith)oej1-0/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@53062 f38db490-d61c-443f-a65b-d21fe96a405b
2006-12-27Be a bit more politically correctoej1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@48985 f38db490-d61c-443f-a65b-d21fe96a405b
2006-12-27Issue #8575 - Buggy cisco MWI support.oej1-0/+3
Normally we try not to change our software for bugs in other devices. But in this case, the Cisco phones are so widespread so we try to implement a fix while waiting for a bugfix from Cisco. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@48982 f38db490-d61c-443f-a65b-d21fe96a405b
2006-12-05Add missing s from another repository. (thanks jcmoore!)oej1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@48268 f38db490-d61c-443f-a65b-d21fe96a405b
2006-12-05Updating sip.conf.sample with information about T38 not workingoej1-0/+2
when chan_local or chan_agent is involved in the call. I don't know how big a fix that would be to solve, but this is the current state of affairs. (Chan_sip currently checks if the other side of the bridge has a SIP tech. We could/should implement another check, possibly for udptl_write or some flag in the ast_channel structure). git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@48264 f38db490-d61c-443f-a65b-d21fe96a405b
2006-12-02- Disable RTP hold timers while T.38 fax transmission happensoej1-8/+18
- Encapsulate RTP timers in the rtp structure so we have one for video and one for audio The video one is not used in 1.4, really. Will be used for RTP keepalives when we can send something that video phones support in the RTP stream. I now this is a big architectual change at this stage for 1.4, but decided it was needed to avoid future bug reports. - Document the RTP NAT keepalive option in sip.conf.sample Issue 7679 in the bug tracker. Please test. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@48199 f38db490-d61c-443f-a65b-d21fe96a405b
2006-12-01- Backport of the "limitonpeers" patch from trunk, to fix a lot of issues ↵oej1-0/+6
with queues and SIP device states - Remove support for T.38 early media, since it's impossible. (Two patches in one - extra friday evening offer due to being off line from svn today... :-) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@48177 f38db490-d61c-443f-a65b-d21fe96a405b
2006-11-30Merged revisions 48142 via svnmerge from file1-2/+4
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: http://svn.digium.com/svn/asterisk/branches/1.4@48143 f38db490-d61c-443f-a65b-d21fe96a405b
2006-11-29Explain the use device status system implemented in SIP for subscriptions,oej1-0/+9
queues and manager a bit better. Like in 1.2, you will get more detailed information if you set a call limit for a device. When the call limit is reached, the status system will report a device as busy. For queues, setting a call limit per SIP device is propably a requirement. In most cases, it will work much better if you only use type=peer and not type=friend. We might decide to backport the new setting from trunk to apply all call limits to the peer part of a friend only. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@48113 f38db490-d61c-443f-a65b-d21fe96a405b
2006-11-29Clarify RTP timers. Sorry, grandma.oej1-3/+5
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@48105 f38db490-d61c-443f-a65b-d21fe96a405b
2006-11-20Explain properly how videosupport works. oej1-1/+5
Committ from Asterisk Video Task Force meeting in Paris! git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@47845 f38db490-d61c-443f-a65b-d21fe96a405b
2006-11-16Make the HOLD notification optional, in order to avoid a lot of extra ↵oej1-1/+4
database lookups for all those realtime users out there. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@47755 f38db490-d61c-443f-a65b-d21fe96a405b
2006-11-16- CANCEL is never authenticated (according to the RFC)oej1-0/+6
- Update docs on canreinvite. "nonat" is the recommended setting for most users with phones behind a NAT. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@47733 f38db490-d61c-443f-a65b-d21fe96a405b
2006-10-31Support ;rport when we're supposed to support ;rport. Issue #7473.oej1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@46628 f38db490-d61c-443f-a65b-d21fe96a405b
2006-10-17Adding information about Marks direct-RTP hack to the docs...oej1-0/+5
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@45329 f38db490-d61c-443f-a65b-d21fe96a405b
2006-10-17Now, remove all traces of the option that we did not need :-)oej1-2/+0
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@45314 f38db490-d61c-443f-a65b-d21fe96a405b
2006-10-16Merged revisions 45265 via svnmerge from file1-1/+2
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: http://svn.digium.com/svn/asterisk/branches/1.4@45280 f38db490-d61c-443f-a65b-d21fe96a405b
2006-10-16Merged revisions 45260 via svnmerge from file1-0/+2
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: http://svn.digium.com/svn/asterisk/branches/1.4@45262 f38db490-d61c-443f-a65b-d21fe96a405b
2006-09-20Add documentation on rtp packetization.qwell1-1/+1
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: http://svn.digium.com/svn/asterisk/trunk@43344 f38db490-d61c-443f-a65b-d21fe96a405b
2006-09-11Merged revisions 42716 via svnmerge from tilghman1-6/+6
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: http://svn.digium.com/svn/asterisk/trunk@42717 f38db490-d61c-443f-a65b-d21fe96a405b
2006-08-31Merge in VLDTMF support with Zaptel/Core done by the ever great Darumkilla ↵file1-1/+8
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: http://svn.digium.com/svn/asterisk/trunk@41507 f38db490-d61c-443f-a65b-d21fe96a405b
2006-07-19merge Russell's 'hold_handling' branch, finally implementing music-on-hold ↵kpfleming1-2/+16
handling the way it was decided at AstriDevCon Europe 2006 (and the way people really want it to be) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@37988 f38db490-d61c-443f-a65b-d21fe96a405b
2006-07-13actually make the non-standard G726-32 behavior available for SIP clientskpfleming1-0/+7
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@37564 f38db490-d61c-443f-a65b-d21fe96a405b
2006-07-10Remove configuration option "restrictcid" that is nowhere tooej1-2/+0
be seen in the code. Did it exist, was it planned to exist or was it documentationware only? Ask Dr Asterisk. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@37324 f38db490-d61c-443f-a65b-d21fe96a405b
2006-07-02- Make use of system name in realtime SIP peers optionaloej1-0/+3
- Fix small issue with SIP history git-svn-id: http://svn.digium.com/svn/asterisk/trunk@36602 f38db490-d61c-443f-a65b-d21fe96a405b
2006-06-30Removing configuration options that does not do anything yet. No need tooej1-3/+4
add "promises" to the sip.conf.sample... git-svn-id: http://svn.digium.com/svn/asterisk/trunk@36355 f38db490-d61c-443f-a65b-d21fe96a405b
2006-06-29Merged revisions 36253-36254 via svnmerge from kpfleming1-5/+7
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: http://svn.digium.com/svn/asterisk/trunk@36262 f38db490-d61c-443f-a65b-d21fe96a405b
2006-06-29reformatting sip.conf.sample a bit, adding dumphistory that was not documentedoej1-38/+42
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@36251 f38db490-d61c-443f-a65b-d21fe96a405b
2006-06-26Speling error. Avoid swenglish :-) (thanks, jtodd!)oej1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@36109 f38db490-d61c-443f-a65b-d21fe96a405b
2006-06-26Add example of permit/deny to sip.conf.sampleoej1-0/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@36054 f38db490-d61c-443f-a65b-d21fe96a405b
2006-06-13Allow AST_FRAME_MODEM frames to be dumped, and document T.38 passthrough supportfile1-0/+7
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@33912 f38db490-d61c-443f-a65b-d21fe96a405b
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