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2008-01-22Documentation updatesoej1-8/+13
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@99483 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-18Merge changes from team/group/sip-tcptlsrussell1-2/+14
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: http://svn.digium.com/svn/asterisk/trunk@99085 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-16Merge the changes from issue #10665 from the team/group/sip_session_timers ↵russell1-7/+35
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: http://svn.digium.com/svn/asterisk/trunk@98978 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-11Add a new global and per-peer option to chan_sip, qualifyfreq, which allows yourussell1-0/+9
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: http://svn.digium.com/svn/asterisk/trunk@98027 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-19Adding the ability to specify the To: header in an outbound INVITEoej1-11/+31
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: http://svn.digium.com/svn/asterisk/trunk@93897 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-16HUGE improvements to QoS/CoS handling by IgorGoej1-5/+5
- 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: http://svn.digium.com/svn/asterisk/trunk@93163 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-16Update documentationoej1-7/+13
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@93160 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-16Make more timers settable in SIP so that we can force timeout earlier on ↵oej1-0/+7
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: http://svn.digium.com/svn/asterisk/trunk@93159 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-05Rename "username" to "defaultuser" to match with "defaultip".oej1-5/+6
"Username" still works, but is deprecated. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@91152 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-27Merged revisions 89624 via svnmerge from oej1-2/+5
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: http://svn.digium.com/svn/asterisk/trunk@89625 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-26Rename "limitonpeer" to "counteronpeer" since the call-limit is deprecated.oej1-1/+1
Both still works in this version. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89613 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-25- Deprecate "call-limit" in chan_sip. No other channel driver enforces ↵oej1-15/+21
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: http://svn.digium.com/svn/asterisk/trunk@89554 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-20Changed occurrences of "busy-level" to "busylevel" in sip.conf.samplemmichelson1-4/+4
in light of commit 89441. Thanks to pj for pointing out the need for this (closes issue #11307, reported by pj) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89453 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-15Add support for application/dtmf SIP INFO dtmf handling. Yep, anotheroej1-1/+2
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: http://svn.digium.com/svn/asterisk/trunk@89278 f38db490-d61c-443f-a65b-d21fe96a405b
2007-09-24merged jcmoore's patch for configurable SDP origin-field username and ↵dhubbard1-0/+5
session field, closes issue# 10795 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@83671 f38db490-d61c-443f-a65b-d21fe96a405b
2007-09-18Merged revisions 82751 via svnmerge from qwell1-1/+1
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: http://svn.digium.com/svn/asterisk/trunk@82752 f38db490-d61c-443f-a65b-d21fe96a405b
2007-09-11Lil' bit more documentation to keep folks happy.file1-2/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@82258 f38db490-d61c-443f-a65b-d21fe96a405b
2007-09-11(closes issue #9433)file1-0/+1
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: http://svn.digium.com/svn/asterisk/trunk@82257 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-27(closes issue #10569)file1-8/+2
Reported by: IgorG Patches: sip_conf-80933-1.patch uploaded by IgorG (license 20) Fix up sip.conf sample configuration. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@80962 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-08Merged revisions 78569 via svnmerge from file1-1/+2
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: http://svn.digium.com/svn/asterisk/trunk@78570 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-21Enhance NAT support as discussed on the -dev list, i.e.:rizzo1-35/+86
+ 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: http://svn.digium.com/svn/asterisk/trunk@76221 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-11Update documentation for proper CLI commands. (issue #9936 reported by eserra)file1-4/+4
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@68662 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-06Remove our little joke that was making fun of email disclaimers which nobodyrussell1-29/+0
else seemed to think was very funny. Oh well ... :) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@67895 f38db490-d61c-443f-a65b-d21fe96a405b
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
generated by cgit v1.2.3 (git 2.25.1) at 2024-06-18 02:14:00 +0000