Tilghman Lesher
9d05ff8ed5
Issue 9869 - replace malloc and memset with ast_calloc, and other coding guidelines changes
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2007-06-06 21:20:11 +00:00
Russell Bryant
033a3df22a
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r67862 | russell | 2007-06-06 16:14:46 -0500 (Wed, 06 Jun 2007) | 4 lines
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)
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2007-06-06 21:16:18 +00:00
Joshua Colp
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r67068 | file | 2007-06-04 15:31:09 -0400 (Mon, 04 Jun 2007) | 2 lines
Better handle SIP devices that say they have SDP content... but really don't. (issue #9398 reported by mthomasslo)
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2007-06-04 19:32:08 +00:00
Russell Bryant
93f2be2675
Fix a couple of places where "tos" was used instead of "cos".
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(issue #9540 , patch by IgorG)
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2007-06-04 15:14:52 +00:00
Russell Bryant
3d2b58751f
To satisfy some legal concerns, add an option for chan_sip to include a
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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.
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2007-05-31 19:41:03 +00:00
Joshua Colp
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r66768 | file | 2007-05-31 12:14:48 -0400 (Thu, 31 May 2007) | 10 lines
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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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2007-05-31 16:18:14 +00:00
Olle Johansson
0b2db74e5a
Issue #9842 - Doxygen updates by snuffy. Thanks!
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(Committed from Media Plaza in Utrecht, Netherlands - Open Source VoIP conference)
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2007-05-31 10:26:55 +00:00
Olle Johansson
4c1068c136
oops. Thanks Terry.
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2007-05-29 19:53:40 +00:00
Olle Johansson
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r66503 | oej | 2007-05-29 21:32:57 +0200 (Tue, 29 May 2007) | 2 lines
Properly handle 408 request timeout - according to the RFC, the dialog dies if a request in a dialog gets this response.
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2007-05-29 19:35:43 +00:00
Olle Johansson
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r66474 | oej | 2007-05-29 21:02:04 +0200 (Tue, 29 May 2007) | 2 lines
Don't issue hangup on hangup on hangup on hangup (for jcmoore)
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2007-05-29 19:17:49 +00:00
Olle Johansson
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r66414 | oej | 2007-05-29 18:07:44 +0200 (Tue, 29 May 2007) | 2 lines
Don't reset hangupcause if we already have one
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2007-05-29 16:19:53 +00:00
Olle Johansson
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r66404 | oej | 2007-05-29 18:02:50 +0200 (Tue, 29 May 2007) | 2 lines
Tracking down hanging channels, killing them one by one. Issue #9235 and related
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2007-05-29 16:17:17 +00:00
Olle Johansson
36f15091bb
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r66363 | oej | 2007-05-29 11:41:40 +0200 (Tue, 29 May 2007) | 10 lines
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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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2007-05-29 10:02:31 +00:00
Joshua Colp
4d03b4f268
Don't try to unregister a peer using the sip unregister CLI command if they are not registered. (issue #9811 reported by eliel)
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2007-05-28 23:28:52 +00:00
Joshua Colp
39e9b3112c
Due to the way stringfields work the value of the url pointer will always be non-NULL so we have to use ast_strlen_zero to make sure it is not empty. (issue #9821 reported by pj)
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2007-05-28 23:24:04 +00:00
Russell Bryant
4b3a3fb14c
Add a new API call for creating detached threads. Then, go replace all of the
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places in the code where the same block of code for creating detached threads
was replicated. (patch from bbryant)
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2007-05-24 18:30:19 +00:00
Joshua Colp
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r65839 | file | 2007-05-24 10:42:12 -0400 (Thu, 24 May 2007) | 10 lines
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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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2007-05-24 14:43:49 +00:00
Olle Johansson
9b95428cce
Issue #8409 and accidentally a fix to chan_sip that wasn't supposed to be there
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but is still ok... Sorry. Lack of Tea, really.
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2007-05-24 14:41:43 +00:00
Kevin P. Fleming
0ec502099f
Yes Virginia, there is a reason why we have stringfields in the sip_pvt structure...
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2007-05-24 11:38:20 +00:00
Russell Bryant
d1ba4f90de
- Remove debug variable that was only used in one place
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- convert string handling to the ast_str API
- Convert strdup() to ast_strdup() and check the result
- Minor formatting changes
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2007-05-24 03:28:39 +00:00
Mark Spencer
04e45cfda3
Add SendURL support
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2007-05-24 02:23:08 +00:00
Kevin P. Fleming
5dc23536ec
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r65683 | kpfleming | 2007-05-23 16:51:56 -0400 (Wed, 23 May 2007) | 10 lines
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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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2007-05-23 20:53:10 +00:00
Olle Johansson
a3f9350ec2
Related to issue #9235 btw.
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r65123 | oej | 2007-05-18 20:16:09 +0200 (Fri, 18 May 2007) | 10 lines
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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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2007-05-18 18:18:59 +00:00
Olle Johansson
1bc7fdeb6b
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r65076 | oej | 2007-05-18 17:18:13 +0200 (Fri, 18 May 2007) | 13 lines
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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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Olle Johansson
491827a28a
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r64974 | oej | 2007-05-18 12:37:44 +0200 (Fri, 18 May 2007) | 2 lines
Issue 9487 - stop media flows at hangup of call
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2007-05-18 10:41:31 +00:00
Olle Johansson
ac343d43c8
Makeup, darling.
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2007-05-18 10:28:56 +00:00
Olle Johansson
f3ec447a23
Another fix for the support for recordings controlled by INFO-packets
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We still lack a setting to enable/disable this per peer
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2007-05-18 08:49:34 +00:00
Joshua Colp
a769766c53
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r64754 | file | 2007-05-17 12:10:12 -0400 (Thu, 17 May 2007) | 2 lines
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)
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2007-05-17 16:11:26 +00:00
Olle Johansson
d83dcae6b1
Below patches with some re-structuring for trunk
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r64602 | oej | 2007-05-16 12:38:18 +0200 (Wed, 16 May 2007) | 2 lines
Issue #9681 - Handle www-auth on BYE
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2007-05-16 10:58:22 +00:00
Olle Johansson
c472b899ef
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r64578 | oej | 2007-05-16 12:05:47 +0200 (Wed, 16 May 2007) | 2 lines
Final part of issue #9483 - fixing transfer() of sip calls in the dial plan (twilson)
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2007-05-16 10:09:42 +00:00
Olle Johansson
c352f7b0d5
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r64543 | oej | 2007-05-16 11:12:34 +0200 (Wed, 16 May 2007) | 10 lines
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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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2007-05-16 10:02:06 +00:00
Olle Johansson
09aec2f622
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r64516 | oej | 2007-05-16 10:46:18 +0200 (Wed, 16 May 2007) | 17 lines
Merged following patch with a lot of changes for 1.4
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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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2007-05-16 08:51:39 +00:00
Olle Johansson
7532b0bc4b
Issue #9304 - Update help text to match functionality. Patch by kshumard with changes by oej
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2007-05-16 07:58:43 +00:00
Olle Johansson
90bad9d2f5
Issue #6789 - Marquis - Add option to support regexten removal when host becomes unreachable
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2007-05-16 07:35:56 +00:00
Olle Johansson
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r64324 | oej | 2007-05-14 21:26:50 +0200 (Mon, 14 May 2007) | 2 lines
Change -2 to XMIT_ERROR to clarify a bit more
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2007-05-14 19:35:58 +00:00
Joshua Colp
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If no port is specified in the outboundproxy setting then use the standard SIP port. (issue #9665 reported by tootai)
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2007-05-14 18:21:30 +00:00
Olle Johansson
fc057b1890
Improve handling network errors on transmission to hosts that don't reply or are unreachable
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With this code, the call will fail as soon as we get a network error. This may happen on
first xmit or a later one, so the retransmit code handles this too.
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2007-05-13 19:20:36 +00:00
Joshua Colp
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r64114 | file | 2007-05-12 18:27:04 -0400 (Sat, 12 May 2007) | 2 lines
This concludes my final adventure with bitmasks and the onhold flag. Would anyone care for some peanuts?
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2007-05-12 22:28:09 +00:00
Joshua Colp
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r64086 | file | 2007-05-12 17:10:45 -0400 (Sat, 12 May 2007) | 2 lines
Tweak hold flags some more. They can be of three states when active: active, inactive, one direction.
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2007-05-12 21:12:18 +00:00
Joshua Colp
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r64044 | file | 2007-05-12 12:32:15 -0400 (Sat, 12 May 2007) | 2 lines
Ensure the onhold flag is set no matter what when being put on hold.
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2007-05-12 16:33:34 +00:00
Olle Johansson
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r63749 | oej | 2007-05-10 22:46:41 +0200 (Thu, 10 May 2007) | 12 lines
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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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Joshua Colp
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r63611 | file | 2007-05-09 12:54:56 -0400 (Wed, 09 May 2007) | 10 lines
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r63610 | file | 2007-05-09 12:51:03 -0400 (Wed, 09 May 2007) | 2 lines
Properly handle hints that point to multiple devices in chan_sip. Why chan_sip is even doing this I have no idea but I would rather not go into a rant. (issue #9536 reported by rlister)
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Olle Johansson
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r63532 | oej | 2007-05-09 15:04:14 +0200 (Wed, 09 May 2007) | 2 lines
Don't retransmit 200 OK's on ignore status. (Reported on asterisk-users)
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2007-05-09 13:07:44 +00:00
Russell Bryant
314c874d7d
I noted this on the dev list but got no response, so I just did it myself.
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Lock the call features when being used in chan_sip.
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2007-05-08 16:41:35 +00:00
Olle Johansson
d326d84ae0
- Adding some missing spaces
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- Correcting error messages
- Disabling code that doesn't do anything
- Making sure we always respond to this request, happily
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2007-05-05 08:05:38 +00:00
Steve Murphy
02337303ef
a small upgrade to the coding standard, and an update to the code that triggered the upgrade.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@63048 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-04 17:49:20 +00:00
Steve Murphy
3ee0077f04
Added a small bit of code to support the SNOM 360's Record button. Made the find_feature func in res_features.c public, so I could use it to find the automon dial sequence as configured by the user. When the INFO packet has a Record: header with on/off, the sequence is sent as consecutive DTMF frames on the phone's channel, triggering the automon functionality. The user has to configure the automon in features.conf, and set up his dialplan accordingly.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@63046 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-04 16:37:23 +00:00
Olle Johansson
1b15d8852d
Add the new ChannelUpdate event to inform manager clients about the PVT ID and some other channel driver data that
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is needed to follow the call through the PBX.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@63032 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-04 13:56:25 +00:00
Joshua Colp
81cade7a4c
Merged revisions 62989 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r62989 | file | 2007-05-03 13:44:00 -0300 (Thu, 03 May 2007) | 10 lines
Merged revisions 62987 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r62987 | file | 2007-05-03 13:42:19 -0300 (Thu, 03 May 2007) | 2 lines
When a peer is seeded or built tell the devicestate core to update it's status. This is easier then having chan_sip load before pbx_config. (issue #9658 reported by dlynes)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@62990 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-03 16:45:39 +00:00
Olle Johansson
e1ec3f917c
Add a small message that we're doing something. On my systems, there's a long
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dead period with a non-responsive CLI after I issue "load chan_sip.so"
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@62656 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-02 12:12:02 +00:00