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${ noResults }
221 Commits (f09b0b3a83c2c33a44f903575924e277801e215b)
| Author | SHA1 | Message | Date |
|---|---|---|---|
Matthew Nicholson
|
91192e30c5 |
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165180 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Joshua Colp
|
fd62012a31 |
Qualify trumps poke per lmadsen.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164814 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Joshua Colp
|
92a4edc593 |
Add configuration options for finer control over how Asterisk handles having to poke all peers at seemingly the same time.
(closes issue #13217) Reported by: cervajs git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164809 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Dwayne M. Hubbard
|
f9b6507796 |
If 'faxdetect=yes' in sip.conf, switch to a 'fax' extension (if it exists) after T38 is negotiated.
Terry Wilson created the original patch for this functionality, which I slightly modified and added the faxdetect=yes|no configuration option. This patch is only for T38 fax detection and does not do anything for G711 over SIP fax detection. By default, this option is disabled. Reviewboard: http://reviewboard.digium.com/r/69/ This functionality is for issue AST-140. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@161115 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Sean Bright
|
7bd3ce358b |
If you enabled 'notifycid' one of the limitations is that the calling channel
is only found if it dialed the extension that was subscribed to. You can now specify 'ignore-context' for the 'notifycid' option in sip.conf which will, as it's value implies, ignore the current context of the caller when doing the lookup. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@158756 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Sean Bright
|
6ba4e7853e |
Allow devices that accept dialog-info+xml (like snoms) to get the Caller ID of
the calling party when subscribed to the state of an extension that is ringing. This has some limitations which are documented in sip.conf.sample. (closes issue #13827) Reported by: seanbright Patches: issue13827.patch uploaded by seanbright (license 71) Reviewed by: russellb git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@154187 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Olle Johansson
|
007807bf41 |
Updating docs
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@153983 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Olle Johansson
|
d3517de987 |
Spaces to replace tabs...
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@153905 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Olle Johansson
|
204845843e |
Adding a separation of remote authentication and our authentication.
remotesecret => our password for a remote service secret => our authentication when someone calls us Secret => still has both functions if remotesecret is not used. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@153904 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Sean Bright
|
0327f37d34 |
The default in chan_sip for notifyringing is yes, so update the sample
conf to reflect that. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@153296 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Joshua Colp
|
f6c78aa0fe |
*whistle*
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@147761 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Joshua Colp
|
cebd2c1df2 |
Add support for subscribing to a voice mailbox on a remote SIP server and making the new/old message count available to local devices. (issue #AST-77)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@147760 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Tilghman Lesher
|
aada13230f |
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: https://origsvn.digium.com/svn/asterisk/trunk@142866 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Tilghman Lesher
|
8b6dd2ad43 |
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: https://origsvn.digium.com/svn/asterisk/trunk@138260 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Russell Bryant
|
35a37e6724 |
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: https://origsvn.digium.com/svn/asterisk/trunk@137732 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Tilghman Lesher
|
6cb6583475 |
SIP should use the transport type set in the Moved Temporarily for the next
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: https://origsvn.digium.com/svn/asterisk/trunk@135126 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Tilghman Lesher
|
853f6a8b3e |
Move implementation of an attended-transfer-complete sound from one channel
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: https://origsvn.digium.com/svn/asterisk/trunk@134401 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Tilghman Lesher
|
5a1d90e1fb |
Additional option for videosupport (always) that disables the optimization to
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: https://origsvn.digium.com/svn/asterisk/trunk@130951 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Olle Johansson
|
e18e813814 |
- Adding alias "udpbindaddr" for the UDP port to comply with "tcpbindaddr" 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: https://origsvn.digium.com/svn/asterisk/trunk@128525 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Olle Johansson
|
638234f146 |
- Fixing issues with "sip show settings"
- 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: https://origsvn.digium.com/svn/asterisk/trunk@128524 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Olle Johansson
|
0fd94cb93d |
Make TCP disabled by default (it's considered experimental)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@128237 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Olle Johansson
|
90098f3cc9 |
Reformatting the config sample
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@128236 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
Brett Bryant
|
1b07e87538 |
Add a configuration option so the global outboundproxy can use tcptls without it being defined by each sip user.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@127154 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Olle Johansson
|
1626397996 |
Merged revisions 126844 via svnmerge from
|
18 years ago |
|
Brett Bryant
|
12d5cebea2 |
Change the way that the transport option works for sip users. transport will 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: https://origsvn.digium.com/svn/asterisk/trunk@125891 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Tilghman Lesher
|
48a9e5cada |
Merged revisions 123883 via svnmerge from
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: https://origsvn.digium.com/svn/asterisk/trunk@123887 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Joshua Colp
|
e4d1b39bd8 |
Merged revisions 118646 via svnmerge from
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: https://origsvn.digium.com/svn/asterisk/trunk@118647 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Joshua Colp
|
f4237076bf |
Add support for specifying the registration expiry on a per registration basis in the register line. This comes from a Switchvox patch. (issue AST-24)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114912 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
Jeff Peeler
|
41fd7a6a21 |
(closes issue #6113)
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: https://origsvn.digium.com/svn/asterisk/trunk@114487 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
Steve Murphy
|
5fb4b1bbe5 |
This is the scariest commit I've done in a long time. This is the 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: https://origsvn.digium.com/svn/asterisk/trunk@114190 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Joshua Colp
|
738e4ec94e |
Add a special dialplan variable to chan_sip which will cause an audio file to be played upon completion of an attended transfer.
(closes issue #9239) Reported by: sunder git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110631 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Russell Bryant
|
a567b41083 |
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/trunk@110499 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Olle Johansson
|
0de4eba640 |
Add manager peerstatus events when peer can't authenticate.
(closes issue #11959) Reported by: mostyn Patches: peerstatus3.patch uploaded by mostyn (license 398) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109316 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 |
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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
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18 years ago |
|
Olle Johansson
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00647ff5f7 |
Update documentation
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93160 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Olle Johansson
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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
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18 years ago |
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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
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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 |
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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
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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 |