aboutsummaryrefslogtreecommitdiffstats
path: root/configs/sip.conf.sample
AgeCommit message (Collapse)AuthorFilesLines
2009-06-26Merged revisions 203699 via svnmerge from file1-4/+6
https://origsvn.digium.com/svn/asterisk/trunk ........ r203699 | file | 2009-06-26 16:27:24 -0300 (Fri, 26 Jun 2009) | 2 lines Improve T.38 negotiation by exchanging session parameters between application and channel. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@203703 f38db490-d61c-443f-a65b-d21fe96a405b
2009-06-15Merged revisions 165180,200689 via svnmerge from kpfleming1-12/+22
https://origsvn.digium.com/svn/asterisk/trunk ........ r165180 | mnicholson | 2008-12-17 12:49:12 -0600 (Wed, 17 Dec 2008) | 14 lines This patch adds a new 'ignoresdpversion' option to sip.conf. When this is enabled (either globally or for a specific peer), chan_sip will treat any SDP data it receives as new data and update the media stream accordingly. By default, Asterisk will only modify the media stream if the SDP session version received is different from the current SDP session version. This option is required to interoperate with devices that have non-standard SDP session version implementations (observed by toc on the bug tracker with Microsoft OCS which always uses 0 as the session version). http://reviewboard.digium.com/r/94/ (closes issue #13958) Reported by: toc Tested by: toc ........ r200689 | kpfleming | 2009-06-15 15:42:38 -0500 (Mon, 15 Jun 2009) | 12 lines Accept T.38 re-INVITE responses with invalid SDP versions. This commit changes the 'incoming SDP version' check logic a bit more; when 'ignoresdpversion' is *not* set for a peer, if we initiate a re-INVITE to switch to T.38, we'll always accept the peer's SDP response, even if they don't properly increment the SDP version number as they should. If this situation occurs, a warning message will be generated suggesting that the peer's configuration be changed to include the 'ignoresdpversion' configuration option (although ideally they'd fix their SIP implementation to be RFC compliant). AST-221 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@200707 f38db490-d61c-443f-a65b-d21fe96a405b
2009-06-02Merged revisions 198791 via svnmerge from file1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r198791 | file | 2009-06-02 10:48:06 -0300 (Tue, 02 Jun 2009) | 5 lines Correct documentation for the register line, specifically where the domain should be specified. (closes issue #14367) Reported by: Nick_Lewis ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@198793 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-22Merged revisions 196416 via svnmerge from dvossel1-4/+7
https://origsvn.digium.com/svn/asterisk/trunk ........ r196416 | dvossel | 2009-05-22 16:09:45 -0500 (Fri, 22 May 2009) | 19 lines SIP set outbound transport type from Registration In sip.conf the transport option allows for the configuration of what transport types (udp, tcp, and tls) a peer will accept, but only the first type listed was used for outbound connections. This patch changes this. Now the default transport type is only used until the peer registers. When registration takes place the transport type is parsed out of the Contact header. If the Contact header's transport type is equal to one that the peer supports, the peer's default transport type for outbound connections is set to match the Contact header's type. If the Contact header's transport type is not present, then the peer's default transport type is set to match the one the peer registered with. When a peer unregisters or the registration expires, the default transport type for that peer is reset. (closes issue #12282) Reported by: rjain Patches: reg_patch_1.diff uploaded by dvossel (license 671) Tested by: dvossel (closes issue #14727) Reported by: pj Patches: reg_patch_3.diff uploaded by dvossel (license 671) Tested by: pj, dvossel Review: https://reviewboard.asterisk.org/r/249/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@196453 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-02Merged revisions 186060 via svnmerge from tilghman1-2/+4
https://origsvn.digium.com/svn/asterisk/trunk ................ r186060 | tilghman | 2009-04-02 12:10:28 -0500 (Thu, 02 Apr 2009) | 16 lines Merged revisions 186059 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r186059 | tilghman | 2009-04-02 12:09:13 -0500 (Thu, 02 Apr 2009) | 9 lines Merged revisions 186056 via svnmerge from 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/1.6.1@186062 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-11Merged revisions 181499 via svnmerge from mvanbaak1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk Provide correct hint to debug SIP trouble in the default config (closes issue #14646) Reported by: strk git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@181521 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-29Merged revisions 171880 via svnmerge from oej1-1/+21
https://origsvn.digium.com/svn/asterisk/trunk ........ r171880 | oej | 2009-01-28 14:26:31 +0100 (Ons, 28 Jan 2009) | 2 lines Add some more notes about device matching. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@172218 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-28Merged revisions 171838 via svnmerge from oej1-0/+19
https://origsvn.digium.com/svn/asterisk/trunk ................ r171838 | oej | 2009-01-28 14:11:44 +0100 (Ons, 28 Jan 2009) | 10 lines Merged revisions 171837 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r171837 | oej | 2009-01-28 14:07:27 +0100 (Ons, 28 Jan 2009) | 2 lines Add a better explanation of the difference between the device namespace and the dialplan for newbies. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@171857 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-15Merged revisions 168711 via svnmerge from oej1-4/+8
https://origsvn.digium.com/svn/asterisk/trunk ........ r168711 | oej | 2009-01-15 18:55:53 +0100 (Tor, 15 Jan 2009) | 4 lines Clarify some misunderstandings and make it even more clear that you can refer to a peer in the register= line. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@168715 f38db490-d61c-443f-a65b-d21fe96a405b
2008-09-12Merged revisions 142866 via svnmerge from tilghman1-0/+14
https://origsvn.digium.com/svn/asterisk/trunk ................ r142866 | tilghman | 2008-09-12 15:49:46 -0500 (Fri, 12 Sep 2008) | 18 lines Merged revisions 142865 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r142865 | tilghman | 2008-09-12 15:37:18 -0500 (Fri, 12 Sep 2008) | 11 lines Create rules for disallowing contacts at certain addresses, which may 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/1.6.1@142868 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-15Merged revisions 138260 via svnmerge from tilghman1-1/+3
https://origsvn.digium.com/svn/asterisk/trunk ................ r138260 | tilghman | 2008-08-15 17:54:57 -0500 (Fri, 15 Aug 2008) | 16 lines Merged revisions 138258 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r138258 | tilghman | 2008-08-15 17:33:42 -0500 (Fri, 15 Aug 2008) | 8 lines More fixes for realtime peers. (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/1.6.1@138261 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-14Merged revisions 137732 via svnmerge from russell1-435/+435
https://origsvn.digium.com/svn/asterisk/trunk ................ r137732 | russell | 2008-08-14 09:15:50 -0500 (Thu, 14 Aug 2008) | 12 lines Merged revisions 137731 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r137731 | russell | 2008-08-14 09:05:23 -0500 (Thu, 14 Aug 2008) | 4 lines Comments in this config file were aligned only if your tab size was set to 8. 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/1.6.1@137813 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-01SIP should use the transport type set in the Moved Temporarily for the nexttilghman1-1/+1
invite. (closes issue #11843) Reported by: pestermann Patches: 20080723__issue11843_302_ignores_transport_16branch.diff uploaded by bbryant (license 36) 20080723__issue11843_302_ignores_transport_trunk.diff uploaded by bbryant (license 36) Tested by: pabelanger git-svn-id: http://svn.digium.com/svn/asterisk/trunk@135126 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-30Move implementation of an attended-transfer-complete sound from one channeltilghman1-2/+4
driver into a common place for multiple channel drivers. (closes issue #13152) Reported by: caio1982 Patches: atxfer_complete_sound3.diff uploaded by caio1982 (license 22) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@134401 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-15Additional option for videosupport (always) that disables the optimization totilghman1-5/+11
fail to setup video RTP if the two endpoints will not support it. This assists with call files and certain transfers to ensure that if two video phones are ever connected, they will always share a video feed. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@130951 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-06- Adding alias "udpbindaddr" for the UDP port to comply with "tcpbindaddr" ↵oej1-5/+2
and "tlsbindaddr". Note: I don't think we can start properly without UDP port open, that needs to be tested. - Removing "bindport" from configuration example, not needed to mention this any more I suggest we deprecate "bindaddr" and "bindport" in trunk (for 1.6.1) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@128525 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-06- Fixing issues with "sip show settings"oej1-1/+2
- Adding IP address for TCP and/or TLS too if auto-domain is enabled and binding to a different IP address - Fixing documentation in sip.conf.sample git-svn-id: http://svn.digium.com/svn/asterisk/trunk@128524 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-05Make TCP disabled by default (it's considered experimental)oej1-3/+4
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@128237 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-05Reformatting the config sampleoej1-16/+15
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@128236 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-01Add a configuration option so the global outboundproxy can use tcptls ↵bbryant1-0/+3
without it being defined by each sip user. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@127154 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-01Merged revisions 126844 via svnmerge from oej1-4/+3
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r126844 | oej | 2008-07-01 14:53:01 +0200 (Tis, 01 Jul 2008) | 5 lines Clear up documentation on "domain=" setting in sip.conf Reported by: davidw (closes issue #12413) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@126845 f38db490-d61c-443f-a65b-d21fe96a405b
2008-06-27Change the way that the transport option works for sip users. transport will ↵bbryant1-0/+7
now take multiple arguments, the first one listed will be the one used for new dialogs, and the rest listed will be acceptable ways for that peer to contact us. This fixes a minor bug where, because SIP TCP/UDP run on the same port, could cause a TCP peer to be saved in the ast_db. There will also be warnings when a transport is changed for an unexpected reason. (issue #12799) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@125891 f38db490-d61c-443f-a65b-d21fe96a405b
2008-06-19Merged revisions 123883 via svnmerge from tilghman1-1/+2
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r123883 | tilghman | 2008-06-19 11:20:41 -0500 (Thu, 19 Jun 2008) | 4 lines Correct description of notifyringing option. (Closes issue #12890) Reported by gminet ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@123887 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-28Merged revisions 118646 via svnmerge from file1-8/+12
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r118646 | file | 2008-05-28 11:23:34 -0300 (Wed, 28 May 2008) | 4 lines Add an option to use the source IP address of RTP as the destination IP 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/trunk@118647 f38db490-d61c-443f-a65b-d21fe96a405b
2008-04-30Add support for specifying the registration expiry on a per registration ↵file1-1/+1
basis in the register line. This comes from a Switchvox patch. (issue AST-24) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@114912 f38db490-d61c-443f-a65b-d21fe96a405b
2008-04-21(closes issue #6113)jpeeler1-0/+3
Reported by: oej Tested by: jpeeler This patch implements multiple parking lots for parked calls. The default parkinglot is used by default, however setting the channel variable PARKINGLOT in the dialplan will allow use of any other configured parkinglot. See configs/features.conf.sample for more details on setting up another non-default parkinglot. Also, one can (currently) set the default parkinglot to use in the driver configuration file via the parkinglot option. Patch initially written by oej, brought up to date and finalized by mvanbaak, and then stabilized and converted to astobj2 by me. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@114487 f38db490-d61c-443f-a65b-d21fe96a405b
2008-04-16This is the scariest commit I've done in a long time. This is the ↵murf1-2/+16
astobj2-ification of chan_sip. I've tested a number of scenarios like crazy. It used to have 4x the call setup/teardown performance of trunk, but now it's roughly at parity. I will attempt to find the bottlenecks and get it back to the 4x mark. The changes made were somewhat invasive, but the value to the community of these upgrades outweighs waiting further for more testing. Every change being made to chan_sip was lousing this code up when we tried to merge. Peers, Users, Dialogs, are all now astobj2 objects, indexed via hashtables. Refcounting is used to track objects and free them at the bitter end of their lives. Please file issues on bugs.digium.com, and PLEASE, please, please be patient. One natural advantage to all the hash-table work is that loading large sip.conf files full of thousands of peers now goes much faster. One more please: PLEASE help thrash this code and test it. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@114190 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-25Add a special dialplan variable to chan_sip which will cause an audio file ↵file1-0/+2
to be played upon completion of an attended transfer. (closes issue #9239) Reported by: sunder git-svn-id: http://svn.digium.com/svn/asterisk/trunk@110631 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-21Note that the TCP and TLS support is currently considered experimental andrussell1-0/+6
is subject to change while we work out the remaining issues. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@110499 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-18Add manager peerstatus events when peer can't authenticate.oej1-0/+2
(closes issue #11959) Reported by: mostyn Patches: peerstatus3.patch uploaded by mostyn (license 398) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@109316 f38db490-d61c-443f-a65b-d21fe96a405b
2008-02-29Add documentation for setting username/password in SIP dial string.file1-1/+10
(closes issue #11587) Reported by: sobomax Patches: dialstring_doc.diff uploaded by sobomax (license 359) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@105378 f38db490-d61c-443f-a65b-d21fe96a405b
2008-02-25Adding more tls configuration details to sip.conf sample, with a list of ↵bbryant1-0/+21
valid ciphers provided in both files. .. First commit since July, woot git-svn-id: http://svn.digium.com/svn/asterisk/trunk@104088 f38db490-d61c-443f-a65b-d21fe96a405b
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
generated by cgit v1.2.3 (git 2.25.1) at 2024-06-16 11:19:04 +0000