asterisk

Commit Graph

Author	SHA1	Message	Date
Russell Bryant	6aa9c04ee9	Merged revisions 110499 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r110499 \| russell \| 2008-03-21 10:24:43 -0500 (Fri, 21 Mar 2008) \| 3 lines Note that the TCP and TLS support is currently considered experimental and is subject to change while we work out the remaining issues. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@110501 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	7422f0ee37	Add documentation for setting username/password in SIP dial string. (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	18 years ago
Brett Bryant	55aaa80d15	Adding more tls configuration details to sip.conf sample, with a list of 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	18 years ago
Olle Johansson	c85b71bf72	Documentation updates git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99483 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	b995c78c31	Merge changes from team/group/sip-tcptls This set of changes introduces TCP and TLS support for chan_sip. There are various new options in configs/sip.conf.sample that are used to enable these features. Also, there is a document, doc/siptls.txt that describes some things in more detail. This code was implemented by Brett Bryant and James Golovich. It was reviewed by Joshua Colp and myself. A number of other people participated in the testing of this code, but since it was done outside of the bug tracker, I do not have their names. If you were one of them, thanks a lot for the help! (closes issue #4903, but with completely different code that what exists there.) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99085 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	6aaa992301	Merge the changes from issue #10665 from the team/group/sip_session_timers branch. This set of changes introduces SIP session timers support (RFC 4028). In short, this prevents stuck SIP sessions that were not properly torn down due to network or endpoint failures during an established SIP session. To quote some of the documentation supplied with the patch: "The SIP Session-Timers is an extension of the SIP protocol that allows end-points and proxies to refresh a session periodically. The sessions are kept alive by sending a RE-INVITE or UPDATE request at a negotiated interval. If a session refresh fails then all the entities that support Session- Timers clear their internal session state. In addition, UAs generate a BYE request in order to clear the state in the proxies and the remote UA (this is done for the benefit of SIP entities in the path that do not support Session-Timers)." (closes issue #10665) Reported by: rjain Patches: chan_sip.c.1.diff uploaded by rjain (license 226) chan_sip.c.diff uploaded by rjain (license 226) sip.conf.sample.diff uploaded by rjain (license 226) proc_422_rsp_comment.diff uploaded by rjain (license 226) chan_sip.c.cache.diff uploaded by rjain (license 226) chan_sip.memalloc uploaded by rjain (license 226) chan_sip.memalloc.bugfix uploaded by rjain (license 226) Patches tracked in team/group/sip_session_timers, with some additional fixes by russell and oej. Tested by: jtodd, rjain, loloski git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98978 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	5c2beee6c3	Add a new global and per-peer option to chan_sip, qualifyfreq, which allows you to set the qualify frequency. (closes issue #11597) Reported by: wilder Patches: qualifyfreq5.patch uploaded by wilder (license 362) -- with some mods by me git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98027 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Olle Johansson	1d6b192ce0	Adding the ability to specify the To: header in an outbound INVITE by adding an exclamation mark to the dial string. This patch also exists for 1.4 in the fixtoheader-1.4 branch and has been in production for quite some time. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93897 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Olle Johansson	17afebc1a6	HUGE improvements to QoS/CoS handling by IgorG - Refer to the proper documentation - Implement separate signalling/media QoS/CoS in many channels using RTP - Improve warnings and verbose messages - Deprecate some old settings Minor modifications by me, a big effort from IgorG. Thanks! Reported by: IgorG Patches: qoscleanup-89394-4-trunk.patch uploaded by IgorG (license 20) Tested by: IgorG (closes issue #11145) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93163 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Olle Johansson	00647ff5f7	Update documentation git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93160 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Olle Johansson	d8795b4542	Make more timers settable in SIP so that we can force timeout earlier on non-responsive SIP servers. Thanks, jcmoore, for the patch! Reported by: jcmoore Patches: peer_t1_timerb_trunk_v3.patch.txt uploaded by jcmoore (license 9) (closes issue #9771) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93159 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Olle Johansson	0cc002a48a	Rename "username" to "defaultuser" to match with "defaultip". "Username" still works, but is deprecated. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@91152 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Olle Johansson	b1c0c67e76	Merged revisions 89624 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89624 \| oej \| 2007-11-27 08:34:19 +0100 (Tis, 27 Nov 2007) \| 6 lines Clarify limitonpeers=yes (closes issue #11304) Reported by: pj ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89625 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Olle Johansson	11df6a9119	Rename "limitonpeer" to "counteronpeer" since the call-limit is deprecated. Both still works in this version. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89613 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Olle Johansson	07cb09ad86	- Deprecate "call-limit" in chan_sip. No other channel driver enforces call-limits and we now have the groupcount system to implement call-limits in the dialplan. You can use the "setvar" option in realtime/sip.conf to set limits per device. - Implement "callcounter" as a new option to enable the call counting we need to report device status to queue, manager and SIP subscriptions. The call counter setting is now enabled in the code by setting the device call-limit to 999. When we remove the call limit, we can simply enable this with a boolean setting. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89554 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	f5e5a443cf	Changed occurrences of "busy-level" to "busylevel" in sip.conf.sample in light of commit 89441. Thanks to pj for pointing out the need for this (closes issue #11307, reported by pj) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89453 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Olle Johansson	eab6b00904	Add support for application/dtmf SIP INFO dtmf handling. Yep, another way of handling DTMF in SIP. Totally undocumented, but implemented in enough devices so we have to support it. Code by sergee, small changes by oej. Closes issue #11049 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89278 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Dwayne M. Hubbard	0f53904918	merged jcmoore's patch for configurable SDP origin-field username and session field, closes issue# 10795 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@83671 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Jason Parker	a9c2f441d3	Merged revisions 82751 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 (Closes issue #10753) ........ r82751 \| qwell \| 2007-09-18 10:28:21 -0500 (Tue, 18 Sep 2007) \| 4 lines Correct the allowexternaldomains option in SIP sample config. Issue 10753 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@82752 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	9bd4b3e353	Lil' bit more documentation to keep folks happy. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@82258 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	9642d93117	(closes issue #9433 ) Reported by: junky Patches: register_trying.diff.txt uploaded by jcmoore Disable sending 100 Trying on REGISTER attempts and make it an option. This has been signed off by oej. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@82257 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	7c760f67c3	(closes issue #10569 ) Reported by: IgorG Patches: sip_conf-80933-1.patch uploaded by IgorG (license 20) Fix up sip.conf sample configuration. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@80962 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	afceb3e4aa	Merged revisions 78569 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r78569 \| file \| 2007-08-08 10:51:01 -0300 (Wed, 08 Aug 2007) \| 4 lines (closes issue #10335) Reported by: adamgundy Update sip.conf to include another scenario where directrtpsetup will fail. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@78570 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Luigi Rizzo	2286afa3af	Enhance NAT support as discussed on the -dev list, i.e.: + extensive documentation changes both in sip.conf.sample and in the source; + allow "externip" and "externhost" to include a port number as well; + allow "bindaddr" to have a port number (making bindport unnecessary, even though it is still present for backward compatibility); + introduce the new "stunaddr" parameter to specify an STUN server to be used from the main SIP socket; + extend the "sip show settings" output to show all the above. Internally: + change related data structures from struct in_addr to struct sockaddr_in to store the port numbers as well; + reorganize ast_sip_ouraddrfor() (should also be renamed to sip_ouraddrfor() because it is not a generic API, though it might become so if called with a socket as an additional argument, in which case it can be moved elsewhere). As mentioned in the documentation, media sessions still do not use STUN so the port numbers may still be incorrect when Asterisk is behind a NAT On passing, some of the debugging messages printing media addresses are probably using the wrong values, but this will be checked/fixed in a subsequent commit if needed. Part of the following chunk in the function that handles a "sip reload" is probably needed on previous versions as well, to avoid leaking the memory used for the "localaddr" list: @@ -17244,13 +17274,17 @@ /* Reset IP addresses / memset(&bindaddr, 0, sizeof(bindaddr)); + memset(&stunaddr, 0, sizeof(stunaddr)); + memset(&internip, 0, sizeof(internip)); + / Free memory for local network address mask */ + ---> ast_free_ha(localaddr); <----- memset(&localaddr, 0, sizeof(localaddr)); memset(&externip, 0, sizeof(externip)); memset(&default_prefs, 0 , sizeof(default_prefs)); git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@76221 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Joshua Colp	cb55dbe8eb	Update documentation for proper CLI commands. (issue #9936 reported by eserra) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@68662 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	6aec360466	Remove our little joke that was making fun of email disclaimers which nobody else seemed to think was very funny. Oh well ... :) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@67895 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	0b6c6b2e89	Add some more information about the SIP Disclaimer header. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@66856 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	3ce231fe95	fix a typo. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@66818 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	3d2b58751f	To satisfy some legal concerns, add an option for chan_sip to include a 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: https://origsvn.digium.com/svn/asterisk/trunk@66777 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
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

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189 Commits (e72038d2c955d33480573d546db02e14f873a93b)