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1674 Commits

Author SHA1 Message Date
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

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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)

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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

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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.

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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

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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

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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

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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)

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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

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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)

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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

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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)

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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

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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)

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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

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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)

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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

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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)

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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

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

Issue #9802 - Change inuse counter on CANCEL

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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

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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)

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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

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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

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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

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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.

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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

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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.

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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
Olle Johansson
5acc63d688 Issue #9681 - Handle www-auth on BYE 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@64602 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-16 10:38:18 +00:00
Olle Johansson
374b52f717 Final part of issue #9483 - fixing transfer() of sip calls in the dial plan (twilson) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@64578 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-16 10:05:47 +00:00
Olle Johansson
70ad556544 Issue #9439 - properly handle username parameters in SIP uri. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@64565 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-16 09:57:22 +00:00
Olle Johansson
bf1a15b9bf Merged revisions 64535 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r64535 | oej | 2007-05-16 11:08:22 +0200 (Wed, 16 May 2007) | 2 lines

Support SIP uri's starting with SIP: and sip: (reported by Tony Mountfield on the mailing list. Thanks!)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@64543 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-16 09:12:34 +00:00
Olle Johansson
56af259505 Merged following patch with a lot of changes for 1.4 
------

Merged revisions 64514 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r64514 | oej | 2007-05-16 10:25:56 +0200 (Wed, 16 May 2007) | 6 lines

Issue #9726 - rlister - Better logging for ACL denials

While at it, also added better logging and handling of peers that are not supposed to register.

My patch, stole the issue report from Russell. My apologies, Russell :-)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@64516 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-16 08:46:18 +00:00
Olle Johansson
1f2afa0ff1 Change -2 to XMIT_ERROR to clarify a bit more 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@64324 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-14 19:26:50 +00:00
Olle Johansson
e041d57175 Handle network errors, like host or network unreachable, in a better way. This means that 
calls to hosts or qualify (OPTION) messages will fail quicker if the TCP/IP stack tells us
that there is an issue.

Since this is an unconnected UDP socket, we will not get error messages directly
in most cases, but maybe on the second and third try.

This is already implemented in trunk.


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@64280 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-14 18:52:09 +00:00
Joshua Colp
9a73c07fb1 This concludes my final adventure with bitmasks and the onhold flag. Would anyone care for some peanuts? 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@64114 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-12 22:27:04 +00:00
Joshua Colp
699aa6ad74 Tweak hold flags some more. They can be of three states when active: active, inactive, one direction. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@64086 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-12 21:10:45 +00:00
Joshua Colp
026000bae8 Ensure the onhold flag is set no matter what when being put on hold. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@64044 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-12 16:32:15 +00:00
Olle Johansson
ca1ae5e81a Merged revisions 63748 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r63748 | oej | 2007-05-10 22:38:54 +0200 (Thu, 10 May 2007) | 4 lines

Do not allocate SIP pvt's for PEERs we can not reach. 

This was seen as a lot of dialogs being created then immediately destroyed at reload/restart of the SIP channel.

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 2007-05-10 20:46:41 +00:00