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r108289 | mmichelson | 2008-03-12 16:57:41 -0500 (Wed, 12 Mar 2008) | 22 lines
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r108288 | mmichelson | 2008-03-12 16:53:46 -0500 (Wed, 12 Mar 2008) | 14 lines
Change AST_SCHED_DEL use to ast_sched_del for autocongestion in chan_sip.
The scheduler callback will always return 0. This means that this id
is never rescheduled, so it makes no sense to loop trying to delete
the id from the scheduler queue. If we fail to remove the item from the
queue once, it will fail every single time.
(Yes I realize that in this case, the macro would exit early because the
id is set to -1 in the callback, but it still makes no sense to use
that macro in favor of calling ast_sched_del once and being done with it)
This is the first of potentially several such fixes.
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r108191 | kpfleming | 2008-03-12 15:27:01 -0500 (Wed, 12 Mar 2008) | 14 lines
Merged revisions 108086 via svnmerge from
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r108086 | kpfleming | 2008-03-12 14:16:07 -0500 (Wed, 12 Mar 2008) | 6 lines
if we receive an INVITE with a Content-Length that is not a valid number, or is zero, then don't process the rest of the message body looking for an SDP
closes issue #11475
Reported by: andrebarbosa
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r106946 | kpfleming | 2008-03-08 10:03:48 -0600 (Sat, 08 Mar 2008) | 10 lines
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r106945 | kpfleming | 2008-03-08 09:59:42 -0600 (Sat, 08 Mar 2008) | 2 lines
don't generate D-Channel "up" and "down" messages unless the channel state is actually changing; also, generate the "up" message when an implicit "up" occurs due to reception of a normal event when we thought the channel was "down"
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r105734 | russell | 2008-03-04 14:36:16 -0600 (Tue, 04 Mar 2008) | 6 lines
Fix some bugs in the SIP tcp helper thread.
- fix a spot where a lock wouldn't get unlocked in an error condition
- call ast_mutex_destroy() on the lock before freeing its memory
(related to issue #11972)
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r105675 | file | 2008-03-04 12:08:42 -0600 (Tue, 04 Mar 2008) | 16 lines
Merged revisions 105674 via svnmerge from
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r105674 | file | 2008-03-04 14:05:28 -0400 (Tue, 04 Mar 2008) | 8 lines
When a new source of audio comes in (such as music on hold) make sure the marker bit gets set.
(closes issue #10355)
Reported by: wdecarne
Patches:
10355.diff uploaded by file (license 11)
(closes issue #11491)
Reported by: kanderson
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r106040 | kpfleming | 2008-03-05 09:40:40 -0600 (Wed, 05 Mar 2008) | 15 lines
Merged revisions 106038 via svnmerge from
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r106038 | kpfleming | 2008-03-05 09:32:35 -0600 (Wed, 05 Mar 2008) | 7 lines
when a PRI call must be moved to a different B channel at the request of the other endpoint, ensure that any DSP active on the original channel is moved to the new one
(closes issue #11917)
Reported by: mavetju
Tested by: mavetju
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automatically generated file like it used to be. This still needs to be there
for modules that have to check it to compile against multiple asterisk versions.
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r104119 | russell | 2008-02-25 18:25:29 -0600 (Mon, 25 Feb 2008) | 33 lines
Merge changes from team/russell/smdi-1.4
This commit brings in a significant set of changes to the SMDI support in Asterisk.
There were a number of bugs in the current implementation, most notably being that
it was very likely on busy systems to pop off the wrong message from the SMDI message
queue. So, this set of changes fixes the issues discovered as well as introducing
some new ways to use the SMDI support which are required to avoid the bugs with
grabbing the wrong message off of the queue.
This code introduces a new interface to SMDI, with two dialplan functions. First,
you get an SMDI message in the dialplan using SMDI_MSG_RETRIEVE() and then you access
details in the message using the SMDI_MSG() function. A side benefit of this is that
it now supports more than just chan_zap.
For example, with this implementation, you can have some FXO lines being terminated
on a SIP gateway, but the SMDI link in Asterisk.
Another issue with the current implementation is that it is quite common that the
station ID that comes in on the SMDI link is not necessarily the same as the Asterisk
voicemail box. There are now additional directives in the smdi.conf configuration
file which let you map SMDI station IDs to Asterisk voicemail boxes.
Yet another issue with the current SMDI support was related to MWI reporting over
the SMDI link. The current code could only report a MWI change when the change
was made by someone calling into voicemail. If the change was made by some other
entity (such as with IMAP storage, or with a web interface of some kind), then the
MWI change would never be sent. The SMDI module can now poll for MWI changes if
configured to do so.
This work was inspired by and primarily done for the University of Pennsylvania.
(also related to issue #9260)
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r104095 | file | 2008-02-25 17:37:20 -0400 (Mon, 25 Feb 2008) | 6 lines
Make it so a users.conf user creates both a SIP peer and a SIP user. The user will be used for inbound authentication for the device, and peer will be used for placing calls to the device.
(closes issue #9044)
Reported by: queuetue
Patches:
sip-gui-friend.diff uploaded by qwell (license 4)
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r104084 | file | 2008-02-25 12:16:13 -0400 (Mon, 25 Feb 2008) | 6 lines
If a resubscription comes in for a dialog we no longer know about tell the remote side that the dialog does not exist so they subscribe again using a new dialog.
(closes issue #10727)
Reported by: s0l4rb03
Patches:
10727-2.diff uploaded by file (license 11)
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r104082 | file | 2008-02-25 11:17:18 -0400 (Mon, 25 Feb 2008) | 6 lines
Due to recent changes tag will no longer be NULL if not present so we have to use ast_strlen_zero to see if it's actually blank.
(closes issue #12061)
Reported by: flefoll
Patches:
chan_sip.c.br14.patch_pedantic_no_totag uploaded by flefoll (license 244)
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