160 Commits (2ce08e259cd4b5e5525427e65ac07148356e068d)

Author	SHA1	Message	Date
Olle Johansson	90bad9d2f5	Issue #6789 - Marquis - Add option to support regexten removal when host becomes unreachable ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@64497 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	b419fc1134	Add support for setting the CoS for VLAN traffic (802.1p) in Linux. The ... file doc/qos.tex has been updated to document the new functionality. (issue #9540, patch submitted by IgorG) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@62457 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	b6b1bf3213	Merge changes from team/russell/events ... 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: https://origsvn.digium.com/svn/asterisk/trunk@62292 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Joshua Colp	ea226e9d77	Merged revisions 58779 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/trunk@58780 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Olle Johansson	88928f67ed	Make documentation match the source code. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@54379 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Olle Johansson	32495f91f0	Add support for outbound proxy for peers and [general] ... 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	19 years ago
Kevin P. Fleming	44c6630e4d	rename busy-limit to busy-level, since it is not a limit ... 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	19 years ago
Olle Johansson	cfe66e6b26	Patch based on this patch with small changes for trunk... ... 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	19 years ago
Olle Johansson	0b84b386b9	Implementing "busy-limit". ... 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	19 years ago
Olle Johansson	064e6cff1a	Merged revisions 53062 via svnmerge from ... 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	19 years ago
Olle Johansson	0375227e5c	Added some docs ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@49081 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Olle Johansson	29ed493b40	Be politically correct ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48986 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Olle Johansson	da7a35a1cc	Add support for buggy Cisco MWI firmware > 8.0.3 (issue 8575 - flewid) ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48983 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Olle Johansson	d1b621c6a5	Adding docs on t.38 ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48269 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Olle Johansson	c23bc46089	- Disable RTP timeouts during T.38 transmission ... - 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	19 years ago
Olle Johansson	4ce5b7c080	- Remove T.38 early media, since T.38 requires two way communication (imported from 1.4) ... - Small fixes to limitonpeer git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48178 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Joshua Colp	c946e3b3fb	Merged revisions 48143 via svnmerge from ... 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	19 years ago
Olle Johansson	7e46275b51	Clarify some settings for status reports in subscriptions, queues and manager ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48114 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Olle Johansson	e5145bebe4	Explain RTP timeouts ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48112 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Olle Johansson	4e47ce525b	Update docs for videosupport ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47846 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Olle Johansson	a6f5adefa1	Make it possible to enable/disable onhold tracking, in order to make life easier ... for realtime users. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47756 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Olle Johansson	a427a2a89a	- CANCEL never uses authentication ... - Add docs on canreinvite git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47734 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Olle Johansson	d900b47ccf	Adding new config option "limitpeersonly" to only apply call limits ... 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	19 years ago
Olle Johansson	b136baaff4	Fix rport handling. ... ...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	19 years ago
Olle Johansson	f98f457727	Change name of "contact" setting to "callback" which better reflects what it ... 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	19 years ago
Luigi Rizzo	e85d8e98d1	document the match_auth_username option ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46308 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Olle Johansson	a8a26ad389	Update of docs ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@45333 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Joshua Colp	c62784c10d	In the course of a data this has been turned into an option to ignore 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	19 years ago
Joshua Colp	da330feb60	Merged revisions 45280 via svnmerge from ... 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	19 years ago
Joshua Colp	b58cc9e1bd	Merged revisions 45262 via svnmerge from ... 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	19 years ago
Olle Johansson	77c69dc4ef	Recommend using "sip reload" since it's much easier to learn and ... remember. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44707 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Luigi Rizzo	b19b4b9764	document a bit the use of templates. ... 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	19 years ago
Luigi Rizzo	f94849ca2a	document the "contact" option a bit better. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44578 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Luigi Rizzo	ccca5843fd	Two things: ... 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	19 years ago
Luigi Rizzo	2a7ac3f735	update example commands to match current syntax ... (does not apply to 1.4) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44537 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Jason Parker	8bd82ebc0d	Add documentation on rtp packetization. ... 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	19 years ago
Tilghman Lesher	091e1aed8d	Merged revisions 42716 via svnmerge from ... 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	19 years ago
Joshua Colp	c6977b9983	Merge in VLDTMF support with Zaptel/Core done by the ever great Darumkilla 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	19 years ago
Kevin P. Fleming	6d0742fc16	merge Russell's 'hold_handling' branch, finally implementing music-on-hold 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	20 years ago
Kevin P. Fleming	4376af0080	actually make the non-standard G726-32 behavior available for SIP clients ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37564 65c4cc65-6c06-0410-ace0-fbb531ad65f3	20 years ago
Olle Johansson	0e0059c0f3	Remove configuration option "restrictcid" that is nowhere to ... 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	20 years ago
Olle Johansson	b971f65978	- Make use of system name in realtime SIP peers optional ... - Fix small issue with SIP history git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@36602 65c4cc65-6c06-0410-ace0-fbb531ad65f3	20 years ago
Olle Johansson	f3594bd1a0	Removing configuration options that does not do anything yet. No need to ... add "promises" to the sip.conf.sample... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@36355 65c4cc65-6c06-0410-ace0-fbb531ad65f3	20 years ago
Kevin P. Fleming	dec3d7d4c0	Merged revisions 36253-36254 via svnmerge from ... 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	20 years ago
Olle Johansson	4177596e8d	reformatting sip.conf.sample a bit, adding dumphistory that was not documented ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@36251 65c4cc65-6c06-0410-ace0-fbb531ad65f3	20 years ago
Olle Johansson	cc43f0bdc7	Speling error. Avoid swenglish :-) (thanks, jtodd!) ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@36109 65c4cc65-6c06-0410-ace0-fbb531ad65f3	20 years ago
Olle Johansson	e2b0c5b558	Add example of permit/deny to sip.conf.sample ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@36054 65c4cc65-6c06-0410-ace0-fbb531ad65f3	20 years ago
Joshua Colp	5456f425c6	Allow AST_FRAME_MODEM frames to be dumped, and document T.38 passthrough support ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@33912 65c4cc65-6c06-0410-ace0-fbb531ad65f3	20 years ago
Russell Bryant	4c76028de9	- add the ability to configure forced jitterbuffers on h323, jingle, ... 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	20 years ago
Kevin P. Fleming	6bce269454	Merged revisions 31321 via svnmerge from ... 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	20 years ago