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2007-06-01Add some more information about the SIP Disclaimer header.russell1-0/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@66856 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-31fix a typo.russell1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@66818 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-31To satisfy some legal concerns, add an option for chan_sip to include arussell1-0/+27
disclaimer along with SIP messages in the header, X-Disclaimer. This is off by default. Also, the text of the disclaimer can be customized in sip.conf. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@66777 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-16Issue #6789 - Marquis - Add option to support regexten removal when host ↵oej1-0/+4
becomes unreachable git-svn-id: http://svn.digium.com/svn/asterisk/trunk@64497 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-30Add support for setting the CoS for VLAN traffic (802.1p) in Linux. Therussell1-1/+7
file doc/qos.tex has been updated to document the new functionality. (issue #9540, patch submitted by IgorG) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@62457 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-28Merge changes from team/russell/eventsrussell1-1/+0
This set of changes introduces a new generic event API for use within Asterisk. I am still working on a way for events to be shared between servers, but this part is ready and can already be used inside of Asterisk. This set of changes introduces the first use of the API, as well. I have restructured the way that MWI (message waiting indication) is handled. It is now event based instead of polling based. For example, if there are a bunch of SIP phones subscribed to mailboxes, then chan_sip will not have to constantly poll the mailboxes for changes. app_voicemail will generate events when changes occur. See UPGRADE.txt and CHANGES for some more information on the effects of these changes from the user perspective. For developer information, see the text in include/asterisk/event.h. As always, additional feedback is welcome on the asterisk-dev mailing list. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@62292 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-12Merged revisions 58779 via svnmerge from file1-1/+3
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r58779 | file | 2007-03-11 20:51:16 -0400 (Sun, 11 Mar 2007) | 2 lines Add matchexterniplocally setting which only substitutes your 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/trunk@58780 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-14Make documentation match the source code. oej1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@54379 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-11Add support for outbound proxy for peers and [general]oej1-2/+6
This replaces the older, broken, implementation where a setting in [general] did not do anything and the [peer] part was broken. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@53932 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-08rename busy-limit to busy-level, since it is not a limitkpfleming1-4/+4
actually parse the busy-limit option from sip.conf, instead of ignoring it git-svn-id: http://svn.digium.com/svn/asterisk/trunk@53577 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-02Patch based on this patch with small changes for trunk...oej1-0/+6
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: http://svn.digium.com/svn/asterisk/trunk@53110 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-01Implementing "busy-limit".oej1-0/+7
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: http://svn.digium.com/svn/asterisk/trunk@53082 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-01Merged revisions 53062 via svnmerge from oej1-0/+1
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: http://svn.digium.com/svn/asterisk/trunk@53063 f38db490-d61c-443f-a65b-d21fe96a405b
2006-12-31Added some docsoej1-0/+3
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@49081 f38db490-d61c-443f-a65b-d21fe96a405b
2006-12-27Be politically correctoej1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@48986 f38db490-d61c-443f-a65b-d21fe96a405b
2006-12-27Add support for buggy Cisco MWI firmware > 8.0.3 (issue 8575 - flewid)oej1-0/+3
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@48983 f38db490-d61c-443f-a65b-d21fe96a405b
2006-12-05Adding docs on t.38oej1-0/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@48269 f38db490-d61c-443f-a65b-d21fe96a405b
2006-12-02- Disable RTP timeouts during T.38 transmissionoej1-8/+18
- 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: http://svn.digium.com/svn/asterisk/trunk@48200 f38db490-d61c-443f-a65b-d21fe96a405b
2006-12-01- Remove T.38 early media, since T.38 requires two way communication ↵oej1-0/+6
(imported from 1.4) - Small fixes to limitonpeer git-svn-id: http://svn.digium.com/svn/asterisk/trunk@48178 f38db490-d61c-443f-a65b-d21fe96a405b
2006-11-30Merged revisions 48143 via svnmerge from file1-2/+4
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: http://svn.digium.com/svn/asterisk/trunk@48144 f38db490-d61c-443f-a65b-d21fe96a405b
2006-11-29Clarify some settings for status reports in subscriptions, queues and manageroej1-0/+9
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@48114 f38db490-d61c-443f-a65b-d21fe96a405b
2006-11-29Explain RTP timeoutsoej1-3/+5
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@48112 f38db490-d61c-443f-a65b-d21fe96a405b
2006-11-20Update docs for videosupportoej1-1/+5
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@47846 f38db490-d61c-443f-a65b-d21fe96a405b
2006-11-16Make it possible to enable/disable onhold tracking, in order to make life easieroej1-1/+4
for realtime users. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@47756 f38db490-d61c-443f-a65b-d21fe96a405b
2006-11-16- CANCEL never uses authenticationoej1-0/+6
- Add docs on canreinvite git-svn-id: http://svn.digium.com/svn/asterisk/trunk@47734 f38db490-d61c-443f-a65b-d21fe96a405b
2006-11-04Adding new config option "limitpeersonly" to only apply call limitsoej1-0/+6
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: http://svn.digium.com/svn/asterisk/trunk@47201 f38db490-d61c-443f-a65b-d21fe96a405b
2006-10-31Fix rport handling.oej1-1/+1
...where did the 1.2 properties come from, really? they're back. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@46629 f38db490-d61c-443f-a65b-d21fe96a405b
2006-10-30Change name of "contact" setting to "callback" which better reflects what itoej1-4/+4
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: http://svn.digium.com/svn/asterisk/trunk@46489 f38db490-d61c-443f-a65b-d21fe96a405b
2006-10-26document the match_auth_username optionrizzo1-0/+4
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@46308 f38db490-d61c-443f-a65b-d21fe96a405b
2006-10-17Update of docsoej1-0/+5
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@45333 f38db490-d61c-443f-a65b-d21fe96a405b
2006-10-16In the course of a data this has been turned into an option to ignore ↵file1-2/+0
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: http://svn.digium.com/svn/asterisk/trunk@45286 f38db490-d61c-443f-a65b-d21fe96a405b
2006-10-16Merged revisions 45280 via svnmerge from file1-1/+2
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: http://svn.digium.com/svn/asterisk/trunk@45281 f38db490-d61c-443f-a65b-d21fe96a405b
2006-10-16Merged revisions 45262 via svnmerge from file1-0/+2
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: http://svn.digium.com/svn/asterisk/trunk@45263 f38db490-d61c-443f-a65b-d21fe96a405b
2006-10-07Recommend using "sip reload" since it's much easier to learn andoej1-1/+1
remember. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@44707 f38db490-d61c-443f-a65b-d21fe96a405b
2006-10-06document a bit the use of templates.rizzo1-0/+44
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: http://svn.digium.com/svn/asterisk/trunk@44579 f38db490-d61c-443f-a65b-d21fe96a405b
2006-10-06document the "contact" option a bit better.rizzo1-0/+8
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@44578 f38db490-d61c-443f-a65b-d21fe96a405b
2006-10-06Two things:rizzo1-0/+9
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: http://svn.digium.com/svn/asterisk/trunk@44566 f38db490-d61c-443f-a65b-d21fe96a405b
2006-10-06update example commands to match current syntaxrizzo1-4/+4
(does not apply to 1.4) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@44537 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
generated by cgit v1.2.3 (git 2.25.1) at 2024-05-18 20:29:47 +0000