kenmirrors/asterisk
1
0
Fork 0
mirror of https://github.com/asterisk/asterisk.git synced 2025-09-21 20:56:39 +00:00
Code Issues Packages Projects Releases Wiki Activity
11,383 Commits 150 Branches 1,597 Tags
38a51a53dbdb45d4e6eeeef8fc6bf1b99f3361a0
Commit Graph

1685 Commits

Author SHA1 Message Date
Tilghman Lesher
a7ade6f213 If peer is not found, the error message is misleading (should be peer not found, not ACL failure) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@78139 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-08-05 03:29:01 +00:00
Mark Michelson
05ba4d90d4 Changed the behavior of sip's realtime_peer function to match the corresponding way of matching for non-realtime peers. 
Now matches are made on both the IP address and port number, or if the insecure setting is set to "port" then just match on the
IP address.

In order to accomplish this, I also added a new API call, ast_category_root, which returns the first variable of an ast_category struct



git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@78103 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-08-03 20:25:22 +00:00
Mark Michelson
43b39d02ae This patch makes Asterisk send 100 Trying provisional responses upon receipt of re-invites. This makes it so that if there are two or more Asterisk 
servers between endpoints, the Asterisk servers will not keep retransmitting the re-invites.

(closes issue #10274, reported by cstadlmann, patched by me with approval from file)



git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@77824 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-07-31 15:21:22 +00:00
Joshua Colp
20d0b01607 (closes issue #10323) 
Reported by: julianjm
Patches:
      chan_sip_device_state_hold_fix.v1.diff.txt uploaded by julianjm (license 99)
Clear ONHOLD flag when decrementing the onHold peer count. If we did not do this the count may keep decreasing.


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@77536 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-07-27 16:27:16 +00:00
Mark Michelson
7a09244181 "re-invite" was misspelled 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@77490 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-07-27 14:30:43 +00:00
Joshua Colp
91eec8f228 Merged revisions 76560 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

........
r76560 | file | 2007-07-23 11:32:07 -0300 (Mon, 23 Jul 2007) | 6 lines

(closes issue #10236)
Reported by: homesick
Patches:
      rpid_1.4_75840.patch uploaded by homesick (license 91)
Accept Remote Party ID on guest calls.

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@76561 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-07-23 14:34:21 +00:00
Russell Bryant
b75f30bdd8 Merged revisions 76226 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

........
r76226 | russell | 2007-07-20 21:01:46 -0500 (Fri, 20 Jul 2007) | 4 lines

Backport a fix for a memory leak that was fixed in trunk in reivision 76221
by rizzo.  The memory used for the localaddr list was not freed during a
configuration reload.

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@76227 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-07-21 02:02:54 +00:00
Joshua Colp
24e7873766 Merged revisions 76080 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

........
r76080 | file | 2007-07-20 14:16:48 -0300 (Fri, 20 Jul 2007) | 6 lines

(closes issue #10247)
Reported by: fkasumovic
Patches:
      chan_sip.patch uploaded by fkasumovic (license #101)
Drop any peer realm authentication entries when reloading so multiple entries do not get added to the peer.

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@76087 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-07-20 17:20:09 +00:00
Joshua Colp
6d143d401f Backport GCC 4.2 fixes. Without these Asterisk won't build under devmode using GCC 4.2. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@75712 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-07-18 20:00:23 +00:00
Joshua Colp
55a12a986e Few more places that needs to check for onhold state. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@75623 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-07-18 15:44:02 +00:00
Joshua Colp
f08e137283 (closes issue #10165) 
Reported by: elandivar

It is possible for hold status to exist without call limits set, so we need to ensure update_call_counter is executed regardless.


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@75621 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-07-18 15:41:06 +00:00
Steve Murphy
2aab6c341f This patch resolves 10143; thanks to irroot for the patch; looked acceptable. Let the community decide if it messes things up 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@74955 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-07-12 20:42:08 +00:00
Joshua Colp
446f14f0dc Only spit out an inringing warning message when it is applicable. Since call limits are already toast in realtime let's not scare the user if they are using it. (issue #10166 reported by bcnit) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@74262 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-07-10 14:07:13 +00:00
Olle Johansson
7bbda30564 While tracking down a bug, I need some more history. Dumphistory is very useful, indeed. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@73849 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-07-08 09:47:31 +00:00
Russell Bryant
5b544349d9 Merged revisions 73768 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

........
r73768 | russell | 2007-07-06 18:01:22 -0500 (Fri, 06 Jul 2007) | 4 lines

If a sip_pvt struct has already registered an extension state callback,
remove the old one before adding a new one.  If this isn't done, Asterisk
will crash.  (issue #10120)

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@73769 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-07-06 23:02:58 +00:00
Russell Bryant
1e588db75a Merged revisions 73678 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

........
r73678 | russell | 2007-07-06 10:55:41 -0500 (Fri, 06 Jul 2007) | 7 lines

(closes issue #10125)
Reported by: makoto
Patches submitted by: makoto

This fixes a crash in chan_sip that happens when the bindaddr setting is not
valid on Asterisk startup, gets fixed, and then a reload gets issued.

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@73679 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-07-06 15:57:25 +00:00
Russell Bryant
be09062a6a Fix a crash in chan_sip. Don't try to stop the monitor thread if it was never 
started.  (closes issue #10124, reported by gzero, fixed by me)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@73598 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-07-05 23:59:22 +00:00
Kevin P. Fleming
3b8be36363 Merged revisions 73547 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

........
r73547 | kpfleming | 2007-07-05 17:11:51 -0500 (Thu, 05 Jul 2007) | 2 lines

we shouldn't allow G.723.1 endpoints to use VAD, just like we don't support it for G.729

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@73548 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-07-05 22:20:44 +00:00
Joshua Colp
9b753a0649 Merged revisions 73466 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

........
r73466 | file | 2007-07-05 16:15:18 -0300 (Thu, 05 Jul 2007) | 2 lines

Copy language information to the dialog structure when calling a peer for situations where a PBX may be started on the dialed channel. (issue #10121 reported by clegall_proformatique)

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@73467 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-07-05 19:18:02 +00:00
Joshua Colp
a6895cb26d Merged revisions 71414 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

........
r71414 | file | 2007-06-24 21:02:49 -0400 (Sun, 24 Jun 2007) | 2 lines

Ignore other URIs after the first in a 300 Multiple Choice response. (issue #10041 reported by homesick)

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@71430 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-25 01:10:06 +00:00
Joshua Colp
f1c32710a8 Merged revisions 70551 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

........
r70551 | file | 2007-06-20 18:20:16 -0400 (Wed, 20 Jun 2007) | 2 lines

Don't overwrite the configured username setting upon a REGISTER. (issue #8565 reported by jsmith)

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@70552 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-20 22:22:20 +00:00
Russell Bryant
a2084d8ab8 Fix a crash that could occur when handing device state changes. 
When the state of a device changes, the device state thread tells the extension
state handling code that it changed.  Then, the extension state code calls the
callback in chan_sip so that it can update subscriptions to that extension.
A pointer to a sip_pvt structure is passed to this function as the call which
needs a NOTIFY sent.  However, there was no locking done to ensure that the pvt
struct didn't disappear during this process.
(issue #9946, reported by tdonahue, patch by me, patch updated to trunk to use
 the sip_pvt lock wrappers by eliel)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@69944 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-19 15:22:36 +00:00
Tilghman Lesher
759dc00599 Issue 10005 - Segfault with missing arguments, plus fix a missing define for SIP INFO channels 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@69796 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-18 19:48:17 +00:00
Joshua Colp
a50bc6e3a8 Don't count RTP timeout when involved in a T38 fax session. (issue #9222 reported by ivoc) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@69794 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-18 19:00:50 +00:00
Joshua Colp
f95038d97e Merged revisions 69765 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

........
r69765 | file | 2007-06-18 14:13:03 -0400 (Mon, 18 Jun 2007) | 2 lines

Set the peer name on the dialog to the one configured in sip.conf and NOT the username to be used for authentication attempts. (issue #9967 reported by achauvin)

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@69775 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-18 18:18:12 +00:00
Joshua Colp
009f6f9112 Don't defer the BYE till later on a transfer when the transfer itself goes kaboom and has no hope of working. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@69668 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-18 16:04:55 +00:00
Joshua Colp
3e29d89200 Few minor transfer tweaks. We can't unlock something we never locked, and better handle a specific scenario with doing an attended transfer between two non-bridged calls. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@69661 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-18 15:46:32 +00:00
Joshua Colp
2c9ffadffb Fix issue where it would be possible for the negotiated codecs to get set back to nothing. (issue #9992 reported by yehavi) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@69625 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-18 13:55:00 +00:00
Russell Bryant
93f3abb3e8 Move the logic for destroying a call when no response is received to a BYE 
outside of the block that checks for FLAG_FATAL to be set.  This flag is only
set when the packet is transmitted with the reliability set to XMIT_CRITICAL
when the original packet is transmitted.  A BYE is always sent with it set
to XMIT_RELIABLE, meaning this code could never be encountered.  This resulted
in seeing some SIP channels that would never go away with the last packet
sent being a BYE.
(part of issue #9235, patch from jcmoore)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@69183 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-13 19:57:38 +00:00
Russell Bryant
f56c3be8ad Clarify a bit of logic. This doesn't change behavior in any way, but it is 
helpful when following the logic to debug problems like 9235.


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@69071 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-13 16:56:16 +00:00
Joshua Colp
b58a48d672 Merged revisions 67938 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

........
r67938 | file | 2007-06-06 20:09:13 -0400 (Wed, 06 Jun 2007) | 2 lines

Only notify the devicestate system of a peer state change when the peer is built from the config file. (issue #9900 reported by arkadia)

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@67941 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-07 00:10:48 +00:00
Russell Bryant
6595debbc5 Fix a crash when doing call pickups with SIP phones. The code unlocked the 
channel when it should not have.
(issue #9652, reported by corruptor, fixed by me)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@67862 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-06 21:14:46 +00:00
Joshua Colp
01456184f9 Better handle SIP devices that say they have SDP content... but really don't. (issue #9398 reported by mthomasslo) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@67068 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-04 19:31:09 +00:00
Joshua Colp
c7112015ba Merged revisions 66764 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

........
r66764 | file | 2007-05-31 12:12:39 -0400 (Thu, 31 May 2007) | 2 lines

It is now possible for this path of execution to have the frame pointer be NULL, therefore we need to check for it before trying to access it. (issue #9836 reported by barthpbx)

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@66768 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-31 16:14:48 +00:00
Joshua Colp
612f61a9b2 Silly me for having out of date source! Oh well... I'm still leaving my comment. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@66639 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-30 17:28:12 +00:00
Joshua Colp
3d8d697e34 When calling some peer/host that may not exist/reply back... don't keep the dialog in memory for all of eternity. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@66637 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-30 17:21:06 +00:00
Olle Johansson
332eabcc07 Properly handle 408 request timeout - according to the RFC, the dialog dies if a request in a dialog gets this response. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@66503 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-29 19:32:57 +00:00
Olle Johansson
8d06f379fe Don't issue hangup on hangup on hangup on hangup (for jcmoore) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@66474 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-29 19:02:04 +00:00
Olle Johansson
9f15005143 Don't reset hangupcause if we already have one 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@66414 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-29 16:07:44 +00:00
Olle Johansson
ff9e2751c6 Tracking down hanging channels, killing them one by one. Issue #9235 and related 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@66404 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-29 16:02:50 +00:00
Olle Johansson
bab6473879 Merged revisions 66349 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

........
r66349 | oej | 2007-05-29 09:53:14 +0200 (Tue, 29 May 2007) | 2 lines

Issue #9802 - Change inuse counter on CANCEL

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@66363 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-29 09:41:40 +00:00
Joshua Colp
0df2a42f96 Merged revisions 65837 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

........
r65837 | file | 2007-05-24 10:40:38 -0400 (Thu, 24 May 2007) | 2 lines

Allow RFC2833 to be negotiated when an INVITE comes in without SDP and is not matched to a user or peer. (issue #9546 reported by mcrawford)

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65839 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-24 14:42:12 +00:00
Olle Johansson
6cfe6a550e Issue 8409 - phsultan - Fix "login" as component to jabber server. 
...and, by accident, fix a bug in chan_sip for stopping a loop on retransmits
   of BYE requests.


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65836 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-24 14:38:30 +00:00
Kevin P. Fleming
ca6b421be4 Merged revisions 65682 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

........
r65682 | kpfleming | 2007-05-23 16:46:22 -0400 (Wed, 23 May 2007) | 2 lines

ensure that variables are set on a newly created channel before we start a PBX on it

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65683 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-23 20:51:56 +00:00
Olle Johansson
4483fa12e8 Merged revisions 65122 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

........
r65122 | oej | 2007-05-18 20:10:46 +0200 (Fri, 18 May 2007) | 2 lines

Not getting an ACK to a 200 OK in the initial invite is critical to the call.

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65123 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-18 18:16:09 +00:00
Olle Johansson
7fe3608300 Merged revisions 65075 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

........
r65075 | oej | 2007-05-18 17:12:09 +0200 (Fri, 18 May 2007) | 5 lines

Issue 9235 - part of the problem, maybe not all. Please retry with this patch (and no
other patch) if you have problems with hanging SIP channels. Thank you.

A special Thank You to WeBRainstorm that gave me access to his system.

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65076 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-18 15:18:13 +00:00
Olle Johansson
50f79ba4b2 - Adding support for putting calls OFF hold with a re-invite with blank SDP. This was a bug found while doing tests at SIPit in Antwerp. 
- In order to not duplicate code, I restructured some of the code for putting calls on/off hold.

Thanks DEA for reminding me. This fix has been asleep in the videocaps branch until now.


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65041 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-18 12:58:39 +00:00
Olle Johansson
73d0ba053b Issue 9487 - stop media flows at hangup of call 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@64974 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-18 10:37:44 +00:00
Joshua Colp
7a8ca54257 Even more direct RTP setup fixes! Don't allow a codec that isn't supported to creep into the SDP of either side. (issue #9446 reported by marcelbarbulescu) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@64754 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-17 16:10:12 +00:00
Olle Johansson
4ae20ba8e4 Fix auth on BYE. (Different patch than for 1.2) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@64605 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-16 10:59:28 +00:00