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git.ipfire.org Git - thirdparty/asterisk.git/log
Richard Mudgett [Mon, 20 Sep 2010 23:14:42 +0000 (23:14 +0000)]
The inalarm flag was not set in sig_analog struct if the port is initially in alarm.
Fixed initial inalarm value for sig_analog ports.
Along with -r261007, this gets the inalarm flag in sync with chan_dahdi
for sig_analog ports.
(closes issue #16983)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@287683
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Alec L Davis [Mon, 20 Sep 2010 22:21:50 +0000 (22:21 +0000)]
ast_do_masquerade. Keep channels ao2_container locked while unlink and linking channels.
Previously, Masquerade would unlock 'original' and 'clonechan' and allow another masq thread to run.
End result would be corrupted memory, and the frequent report 'Bad Magic Number'.
(closes issue #17801,#17710)
Reported by: notthematrix
Patches:
Based on bug17801.diff1.txt uploaded by alecdavis (license 585)
Tested by: alecdavis
Review: https://reviewboard.asterisk.org/r/928
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@287661
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David Vossel [Mon, 20 Sep 2010 22:09:16 +0000 (22:09 +0000)]
Addition of the FrameHook API (AKA AwesomeHooks)
So far all our tools for viewing and manipulating media streams
within Asterisk have been entirely focused on audio. That made
sense then, but is not scalable now. The FrameHook API lets us
tap into and manipulate _ANY_ type of media or signaling passed
on a channel present today or in the future. This tool is a step
in the direction of expanding Asterisk's boundaries and will help
generate some rather interesting applications in the future.
In addition to the FrameHook API, a simple dialplan function
exercising the api has been included as well. This function
is called FRAME_TRACE(). FRAME_TRACE() allows for the internal
ast_frames read and written to a channel to be output. Filters
can be placed on this function to debug only certain types of frames.
This function could be thought of as an internal way of doing
ast_frame packet captures.
Review: https://reviewboard.asterisk.org/r/925/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@287647
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David Vossel [Mon, 20 Sep 2010 21:34:15 +0000 (21:34 +0000)]
Fixes issue with registrations not working properly with pedantic=yes.
(closes issue #18017)
Reported by: schmidts
Patches:
issues_18017_v1.diff uploaded by dvossel (license 671)
Tested by: schmidts
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@287645
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Jason Parker [Mon, 20 Sep 2010 21:29:46 +0000 (21:29 +0000)]
Merged revisions 287642 via svnmerge from
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r287642 | qwell | 2010-09-20 16:28:32 -0500 (Mon, 20 Sep 2010) | 8 lines
Don't crash when parking a non-bridged call.
(closes issue #17680)
Reported by: jmhunter
Patches:
chan_skinny-park-v1.txt uploaded by DEA (license 3)
Tested by: jmhunter, DEA
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Brett Bryant [Mon, 20 Sep 2010 21:19:12 +0000 (21:19 +0000)]
Fixes an error with the logger that caused verbose messages to be spammed to the
screen if syslog was configured in logger.conf
(closes issue #17974)
Reported by: lmadsen
Review: https://reviewboard.asterisk.org/r/915/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@287639
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Matthew Nicholson [Mon, 20 Sep 2010 15:57:14 +0000 (15:57 +0000)]
Merged revisions 287558 via svnmerge from
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r287558 | mnicholson | 2010-09-20 10:56:21 -0500 (Mon, 20 Sep 2010) | 14 lines
Use ast_str when processing hint state changes
Merged revisions 287555 via svnmerge from
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r287555 | mnicholson | 2010-09-20 10:48:14 -0500 (Mon, 20 Sep 2010) | 5 lines
Use ast_dynamic_str when processing hint state changes
(related to issue #17928)
Reported by: mdu113
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Olle Johansson [Sun, 19 Sep 2010 16:09:28 +0000 (16:09 +0000)]
Merged revisions 287470 via svnmerge from
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r287470 | oej | 2010-09-19 18:06:10 +0200 (Sön, 19 Sep 2010) | 14 lines
Merged revisions 287469 via svnmerge from
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r287469 | oej | 2010-09-19 17:56:50 +0200 (Sön, 19 Sep 2010) | 7 lines
Make sure we always free variables properly in manager originate.
(closes issue #17891)
reported, solved and tested by oej
Review: https://reviewboard.asterisk.org/r/869/
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Tilghman Lesher [Fri, 17 Sep 2010 21:08:54 +0000 (21:08 +0000)]
Merged revisions 287387 via svnmerge from
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r287387 | tilghman | 2010-09-17 16:08:00 -0500 (Fri, 17 Sep 2010) | 14 lines
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r287386 | tilghman | 2010-09-17 16:06:03 -0500 (Fri, 17 Sep 2010) | 7 lines
Blank columns should get set on reload, not ignored.
(closes issue #16893)
Reported by: haakon
Patches:
20100818__issue16893.diff.txt uploaded by tilghman (license 14)
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Matthew Nicholson [Fri, 17 Sep 2010 13:37:10 +0000 (13:37 +0000)]
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r287308 | mnicholson | 2010-09-17 08:36:07 -0500 (Fri, 17 Sep 2010) | 12 lines
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r287307 | mnicholson | 2010-09-17 08:34:34 -0500 (Fri, 17 Sep 2010) | 5 lines
Use ast_strdup() instead of ast_strdupa() while processing in ast_hint_state_changed().
(related to issue #17928)
Reported by: mdu113
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Jan Kalab [Fri, 17 Sep 2010 08:44:28 +0000 (08:44 +0000)]
Events are visible after they were removed from EWS calendar
Because we must merge calendar even when it's empty.
(closes issue #17786)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@287271
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Jan Kalab [Fri, 17 Sep 2010 08:42:37 +0000 (08:42 +0000)]
Asterisk crashing because of double free when EWS request fails
The free is done later in code. I think ast_free() should have built in checks for double free.
(closes issue #17782)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@287270
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Jan Kalab [Fri, 17 Sep 2010 08:37:49 +0000 (08:37 +0000)]
Support for HTTP redirects in calendar's URL
libneon does not support HTTP redirects (3xx responses) by default. You must tell it to follow them.
Also, another little unsigned int fix.
(closes issue #17776)
Review: https://reviewboard.asterisk.org/r/921/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@287269
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Jason Parker [Thu, 16 Sep 2010 22:04:38 +0000 (22:04 +0000)]
Don't fail when running the Debian init script directly (as one would normally do).
readlink apparently returns 1 when the arg isn't a symlink, which caused the script to exit.
(closes issue #17910)
Reported by: wurstsalat
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@287195
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Russell Bryant [Thu, 16 Sep 2010 21:57:51 +0000 (21:57 +0000)]
Set the default for "autofill" and "shared_lastcall" to "yes" in queues.conf.
Review: https://reviewboard.asterisk.org/r/922/
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Matthew Nicholson [Thu, 16 Sep 2010 20:07:38 +0000 (20:07 +0000)]
Merged revisions 287119 via svnmerge from
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r287119 | mnicholson | 2010-09-16 15:06:16 -0500 (Thu, 16 Sep 2010) | 15 lines
Merged revisions 287118 via svnmerge from
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r287118 | mnicholson | 2010-09-16 15:04:46 -0500 (Thu, 16 Sep 2010) | 8 lines
Don't limit hint processing in ast_hint_state_changed() to AST_MAX_EXTENSION length strings.
(closes issue #17928)
Reported by: mdu113
Patches:
20100831__issue17928.diff.txt uploaded by tilghman (license 14)
Tested by: mdu113
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Matthew Nicholson [Thu, 16 Sep 2010 19:54:48 +0000 (19:54 +0000)]
Merged revisions 287115 via svnmerge from
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r287115 | mnicholson | 2010-09-16 14:53:41 -0500 (Thu, 16 Sep 2010) | 15 lines
Merged revisions 287114 via svnmerge from
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r287114 | mnicholson | 2010-09-16 14:52:39 -0500 (Thu, 16 Sep 2010) | 8 lines
Don't stop printing cdr variables if we encounter one with a blank name or value.
(closes issue #17900)
Reported by: under
Patches:
core-show-channel-cdr-fix1.diff uploaded by mnicholson (license 96)
Tested by: mnicholson
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Terry Wilson [Wed, 15 Sep 2010 22:17:17 +0000 (22:17 +0000)]
Don't hang up a call on an SRTP unprotect failure
Also make it more obvious when there is an issue en/decrypting.
(closes issue #17563)
Reported by: Alexcr
Patches:
res_srtp.c.patch uploaded by sfritsch (license 1089)
Tested by: twilson
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@287056
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Jeff Peeler [Wed, 15 Sep 2010 20:58:39 +0000 (20:58 +0000)]
fix uninintialized variable
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@287020
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Richard Mudgett [Wed, 15 Sep 2010 20:53:38 +0000 (20:53 +0000)]
Merged revision 287014 from
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r287014 | rmudgett | 2010-09-15 15:32:24 -0500 (Wed, 15 Sep 2010) | 58 lines
The handling of call transfer signaling for mISDN PTMP is not fully implemented.
The handling of call transfer signaling for mISDN PTMP is not fully
implemented. The signaling of number updates with ISDN/DSS1 ECT
supplementary services (ETS 300 369-1) comes along with a notification
indicator IE and redirection number IE for PTMP. The implementation in
the current Asterisk mISDN channel unfortunately can handle these
information elements only in a NOTIFY message. These information elements
are also signaled in a FACILTY message with a RequestSubaddress facility,
when the subscriber is already in the active state (see 9.2.4 and 9.2.5 of
ETS 300 369-1).
**********
abe_2526_ast.patch
* Added support to handle the notification indicator IE and redirection
number IE with the RequestSubaddress facility.
* Made misdn_update_connected_line() send a NOTIFY message if Asterisk
originated the call and it is not connected yet.
* Made misdn_update_connected_line() send a FACILITY message if the call
is already connected.
This patch requires the presence of the associated mISDN patches to
compile. I had to enhance mISDN to allow the notification indicator IE
and the redirection number IE to be used with a FACILITY message. Earlier
versions of the Digium enhanced mISDN are no longer going to work.
**********
abe_2526_misdn.patch
* Made an incoming FACILITY message allow the presence of the notification
indicator IE and the redirection number IE.
**********
abe_2526_misdnuser_v3.patch
* Added support to send and receive a FACILITY message with the
notification indicator IE and the redirection number IE.
* Added the ability to send a NOTIFY message in PTMP/NT mode to all
responding subcalls in Q.931 states 6, 7, 8, 9, and 25.
**********
Patches:
abe_2526_ast.patch uploaded by rmudgett (license 664)
abe_2526_misdn.patch uploaded by rmudgett (license 664)
abe_2526_misdnuser_v3.patch uploaded by rmudgett (license 664)
Tested by: rmudgett and reporter
JIRA SWP-2146
JIRA ABE-2526
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Jeff Peeler [Wed, 15 Sep 2010 20:32:52 +0000 (20:32 +0000)]
Merged revisions 286998 via svnmerge from
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r286998 | jpeeler | 2010-09-15 15:28:02 -0500 (Wed, 15 Sep 2010) | 14 lines
Merged revisions 286941 via svnmerge from
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r286941 | jpeeler | 2010-09-15 15:08:52 -0500 (Wed, 15 Sep 2010) | 7 lines
Ensure mailbox is not filled to capacity before doing message forwarding.
Specifically, before prompting to record a prepended message the capacity is
checked first. If the mailbox is full the extension will be reprompted.
ABE-2517
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Jeff Peeler [Wed, 15 Sep 2010 19:22:15 +0000 (19:22 +0000)]
Add parking extension for non-default parking lots.
This is a new feature that allows for parking to custom parking lots to be
accessed directly, rather than with channel variables or by changing the
default parking lot. The extension is set with the parkext option just as the
default parking lot is done. Also, the manager action has been updated to
optionally allow a specified parking lot.
(closes issue #14882)
Reported by: vmikhnevych
Patches:
patch_14882.txt uploaded by mnick (license 874)
modified by me
Review: https://reviewboard.asterisk.org/r/884/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@286931
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Richard Mudgett [Wed, 15 Sep 2010 18:29:21 +0000 (18:29 +0000)]
Simplify some code in sig_analog.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@286905
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Richard Mudgett [Wed, 15 Sep 2010 18:28:05 +0000 (18:28 +0000)]
Unable to originate calls using E&M over T1.
When originating a call from Unit Under Test to Reference Unit using E&M
RBS signaling mode, I get the following warning message: "Ring/Off-hook in
strange state 3 on channel 1".
Fixed the sig_analog outgoing flag. It was never set when sig_analog was
extracted from chan_dahdi.
JIRA SWP-2191
JIRA AST-408
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Matthew Nicholson [Wed, 15 Sep 2010 13:05:52 +0000 (13:05 +0000)]
Set tohost to the domain specified in the configuration file instead of the IP address of the host we are calling.
This fixes a regression introduced in r274783.
(closes issue #17960)
Reported by: adriavidal
Patches:
sip-tohost-fix1.diff uploaded by mnicholson (license 96)
Tested by: mich, mnicholson, adriavidal
(closes issue #17676)
Reported by: outcast
Patches:
sip-tohost-fix1.diff uploaded by mnicholson (license 96)
Tested by: mnicholson
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@286868
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David Vossel [Tue, 14 Sep 2010 21:57:35 +0000 (21:57 +0000)]
Sets subscribed type for outgoing MWI subscriptions so correct Event header is used.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@286834
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Matthew Nicholson [Tue, 14 Sep 2010 19:28:38 +0000 (19:28 +0000)]
Merged revisions 286757 via svnmerge from
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r286757 | mnicholson | 2010-09-14 14:27:28 -0500 (Tue, 14 Sep 2010) | 20 lines
Merged revisions 286756 via svnmerge from
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r286756 | mnicholson | 2010-09-14 14:26:18 -0500 (Tue, 14 Sep 2010) | 13 lines
Don't clear the username from a realtime database when a registration expires.
Non-realtime chan_sip does not clear the username from memory when a registration expiries so realtime probably shouldn't either.
(closes issue #17551)
Reported by: ricardolandim
Patches:
reg-expiry-username-1.4-fix1.diff uploaded by mnicholson (license 96)
reg-expiry-username-1.6.2-fix1.diff uploaded by mnicholson (license 96)
reg-expiry-username-1.8-fix1.diff uploaded by mnicholson (license 96)
reg-expiry-username-trunk-fix1.diff uploaded by mnicholson (license 96)
Tested by: ricardolandim, mnicholson
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Matthew Nicholson [Tue, 14 Sep 2010 18:04:21 +0000 (18:04 +0000)]
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r286681 | mnicholson | 2010-09-14 13:02:24 -0500 (Tue, 14 Sep 2010) | 14 lines
Merged revisions 286679 via svnmerge from
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r286679 | mnicholson | 2010-09-14 13:00:01 -0500 (Tue, 14 Sep 2010) | 7 lines
Only drop duplicate answer frames if the channel is bridged.
Back in r3710 ast_read() was modified to drop answer frames on channels that were in the UP state. This modification prevented bridges that were up before the answer from being broken and reestablished by an ANSWER control frame. That change also prevents pickup of channels called from the ast_dial framework from working properly. The ast_dial framework expects to see an ANSWER frame after dialing and the pickup code queues one but ast_read() drops it. This new change only drops ANSWER frames when the channel is bridged, allowing the answer queued by the pickup code to properly pass through ast_read() on to the ast_dial framework.
ABE-2473
(related to issue #2342)
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Richard Mudgett [Tue, 14 Sep 2010 15:30:49 +0000 (15:30 +0000)]
Corrected documented CONNECTED_LINE and REDIRECTING party manipulation macro names.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@286647
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Jan Kalab [Tue, 14 Sep 2010 06:55:44 +0000 (06:55 +0000)]
Merging events for Exchange web service doesn't work as expected, resulting in only one event in calendar
The solution is to use "global" counter of events, since we do new requests for every event and calendar sync after every request. So now we do sync only after last request.
(closes issue #17877)
Review: https://reviewboard.asterisk.org/r/916/
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Tilghman Lesher [Tue, 14 Sep 2010 05:07:16 +0000 (05:07 +0000)]
Merged revisions 286587 via svnmerge from
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r286587 | tilghman | 2010-09-14 00:06:05 -0500 (Tue, 14 Sep 2010) | 2 lines
Add documentation on missing backend tables for Voicemail
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Tilghman Lesher [Mon, 13 Sep 2010 23:50:34 +0000 (23:50 +0000)]
Merged revisions 286557 via svnmerge from
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r286557 | tilghman | 2010-09-13 18:48:51 -0500 (Mon, 13 Sep 2010) | 2 lines
C precedence got me
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Tilghman Lesher [Mon, 13 Sep 2010 23:12:21 +0000 (23:12 +0000)]
Merged revisions 286527 via svnmerge from
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r286527 | tilghman | 2010-09-13 18:03:26 -0500 (Mon, 13 Sep 2010) | 2 lines
Refactor conversion to ast_poll() to fix callparking regression.
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Jason Parker [Mon, 13 Sep 2010 19:40:05 +0000 (19:40 +0000)]
Merged revisions 286456 via svnmerge from
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r286456 | qwell | 2010-09-13 14:38:35 -0500 (Mon, 13 Sep 2010) | 5 lines
Remove "Internal IP" from sip show settings, as it's not at all useful to display.
(closes issue #17840)
Reported by: oej
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Richard Mudgett [Mon, 13 Sep 2010 15:52:14 +0000 (15:52 +0000)]
Update chan_dahdi.conf.sample to reflect new libpri T309 default value.
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Olle Johansson [Sat, 11 Sep 2010 17:09:22 +0000 (17:09 +0000)]
Merged revisions 286268 via svnmerge from
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r286268 | oej | 2010-09-11 19:05:16 +0200 (Lör, 11 Sep 2010) | 11 lines
Merged revisions 286267 via svnmerge from
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r286267 | oej | 2010-09-11 18:59:20 +0200 (Lör, 11 Sep 2010) | 4 lines
Handle error response when we can't make file compatible
Review: https://reviewboard.asterisk.org/r/911/
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Terry Wilson [Fri, 10 Sep 2010 22:04:53 +0000 (22:04 +0000)]
Merged revisions 286115 via svnmerge from
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r286115 | twilson | 2010-09-10 15:35:25 -0500 (Fri, 10 Sep 2010) | 23 lines
Merged revisions 286059 via svnmerge from
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r286059 | twilson | 2010-09-10 14:25:08 -0500 (Fri, 10 Sep 2010) | 16 lines
Inherit CHANNEL() writes to both sides of a Local channel
Having Local (/n) channels as queue members and setting the language in the
extension with Set(CHANNEL(language)=fr) sets the language on the Local/...,2
channel. Hold time report playbacks happen on the Local/...,1 channel and
therefor do not play in the specified language.
This patch modifies func_channel_write to call the setoption callback and pass
the CHANNEL() write info to the callback. chan_local uses this information to
look up the other side of the channel and apply the same changes to it.
(closes issue #17673)
Reported by: Guggemand
Review: https://reviewboard.asterisk.org/r/903/
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Paul Belanger [Fri, 10 Sep 2010 21:11:08 +0000 (21:11 +0000)]
Merged revisions 286117 via svnmerge from
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r286117 | pabelanger | 2010-09-10 16:55:06 -0400 (Fri, 10 Sep 2010) | 11 lines
Merged revisions 286114 via svnmerge from
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r286114 | pabelanger | 2010-09-10 16:35:08 -0400 (Fri, 10 Sep 2010) | 4 lines
Load iax.conf before registering any functions/applications/actions.
Review: https://reviewboard.asterisk.org/r/914/
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Richard Mudgett [Fri, 10 Sep 2010 20:55:37 +0000 (20:55 +0000)]
Merged revisions 286116 via svnmerge from
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r286116 | rmudgett | 2010-09-10 15:42:44 -0500 (Fri, 10 Sep 2010) | 18 lines
Merged revisions 286113 via svnmerge from
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r286113 | rmudgett | 2010-09-10 15:33:16 -0500 (Fri, 10 Sep 2010) | 11 lines
An outgoing call may not get hung up if a pre-connect incoming ISDN call is disconnected.
If the ISDN link a pre-connect incoming call is using fails or is reset,
the outgoing leg may not hang up or be delayed in hanging up. (Causes:
PRI_CAUSE_NETWORK_OUT_OF_ORDER, PRI_CAUSE_DESTINATION_OUT_OF_ORDER, and
PRI_CAUSE_NORMAL_TEMPORARY_FAILURE.)
Just hang up the call if the incoming call leg hangs up before connecting
for any reason. It makes no sense to send a BUSY or CONGESTION control
frame to the outgoing call leg under these circumstances.
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Russell Bryant [Fri, 10 Sep 2010 20:31:58 +0000 (20:31 +0000)]
Rate limit calls to fsync() to 1 per second after astdb updates.
Astdb was determined to be one of the most significant bottlenecks in SIP
registration processing. This patch improved the speed of an astdb load
test by 50000% (yes, Fifty-Thousand Percent). On this particular load test
setup, this doubled the number of SIP registrations the server could handle.
Review: https://reviewboard.asterisk.org/r/825/
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Tilghman Lesher [Fri, 10 Sep 2010 18:31:12 +0000 (18:31 +0000)]
Merged revisions 286024 via svnmerge from
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r286024 | tilghman | 2010-09-10 13:30:21 -0500 (Fri, 10 Sep 2010) | 9 lines
Merged revisions 286023 via svnmerge from
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r286023 | tilghman | 2010-09-10 13:22:04 -0500 (Fri, 10 Sep 2010) | 2 lines
Missing newline
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David Ruggles [Fri, 10 Sep 2010 13:13:16 +0000 (13:13 +0000)]
Added missing documentation for ExternalIVR feature added in January 2010
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@285992
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Tilghman Lesher [Fri, 10 Sep 2010 05:32:18 +0000 (05:32 +0000)]
Merged revisions 285961 via svnmerge from
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r285961 | tilghman | 2010-09-10 00:31:31 -0500 (Fri, 10 Sep 2010) | 6 lines
Another fix for Mac OS X.
While trying to fix this the "right" way, I wandered into dependency hell. Two
hours later, I backed out, and just removed the offending code. ast_inline_api
only goes one level deep and then it breaks. Ouch.
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Tilghman Lesher [Fri, 10 Sep 2010 01:25:50 +0000 (01:25 +0000)]
Merged revisions 285930 via svnmerge from
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r285930 | tilghman | 2010-09-09 20:16:32 -0500 (Thu, 09 Sep 2010) | 14 lines
Merged revisions 285889 via svnmerge from
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r285889 | tilghman | 2010-09-09 19:13:45 -0500 (Thu, 09 Sep 2010) | 7 lines
Fix Mac OS X build.
This also fixes a rather grievous calculation error for the offset of
ast_fdset, which was masked on Linux and FreeBSD, because these platforms
check the first 256 FDs regardless of the bitmask setting (due to backwards
compatibility).
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Paul Belanger [Thu, 9 Sep 2010 22:52:31 +0000 (22:52 +0000)]
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r285818 | pabelanger | 2010-09-09 18:49:19 -0400 (Thu, 09 Sep 2010) | 15 lines
Merged revisions 285817 via svnmerge from
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r285817 | pabelanger | 2010-09-09 18:34:35 -0400 (Thu, 09 Sep 2010) | 8 lines
GCC 4.2.x optimizations result in improper behavior of GSM codec
(closes issue #17688)
Reported by: pprindeville
Patches:
asterisk-trunk-bugid11243.patch uploaded by pprindeville (license 347)
Tested by: mkeuter, pprindeville
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Jason Parker [Thu, 9 Sep 2010 20:11:06 +0000 (20:11 +0000)]
Merged revisions 285744 via svnmerge from
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r285744 | qwell | 2010-09-09 15:09:23 -0500 (Thu, 09 Sep 2010) | 16 lines
Merged revisions 285742 via svnmerge from
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r285742 | qwell | 2010-09-09 15:06:31 -0500 (Thu, 09 Sep 2010) | 9 lines
Transmit silence when reading DTMF in ast_readstring.
Otherwise, you could get issues with DTMF timeouts causing hangups.
(closes issue #17370)
Reported by: makoto
Patches:
channel-readstring-silence-generator.patch uploaded by makoto (license 38)
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Brett Bryant [Thu, 9 Sep 2010 18:51:52 +0000 (18:51 +0000)]
Merged revisions 285710 via svnmerge from
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r285710 | bbryant | 2010-09-09 14:50:13 -0400 (Thu, 09 Sep 2010) | 8 lines
Fixes an issue with dialplan pattern matching where the specificity for pattern ranges and pattern special characters was inconsistent.
(closes issue #16903)
Reported by: Nick_Lewis
Patches:
pbx.c-specificity.patch uploaded by Nick Lewis (license 657)
Tested by: Nick_Lewis
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Brett Bryant [Thu, 9 Sep 2010 17:23:28 +0000 (17:23 +0000)]
Merged revisions 285639 via svnmerge from
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r285639 | bbryant | 2010-09-09 13:22:25 -0400 (Thu, 09 Sep 2010) | 14 lines
Merged revisions 285638 via svnmerge from
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r285638 | bbryant | 2010-09-09 13:20:17 -0400 (Thu, 09 Sep 2010) | 7 lines
Fixes an issue with MOH where it doesn't recover cleanly when it can't play a file and would just stop, instead of continuing to find the next playable file in the MOH class.
(closes issue #17807)
Reported by: kshumard
Review: https://reviewboard.asterisk.org/r/910/
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David Vossel [Wed, 8 Sep 2010 22:14:19 +0000 (22:14 +0000)]
Merged revisions 285567 via svnmerge from
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r285567 | dvossel | 2010-09-08 17:11:28 -0500 (Wed, 08 Sep 2010) | 9 lines
Merged revisions 285566 via svnmerge from
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r285566 | dvossel | 2010-09-08 17:07:31 -0500 (Wed, 08 Sep 2010) | 2 lines
In retrans_pkt, do not unlock pvt until the end of the function on a transmit failure.
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David Vossel [Wed, 8 Sep 2010 21:48:37 +0000 (21:48 +0000)]
Merged revisions 285563 via svnmerge from
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r285563 | dvossel | 2010-09-08 16:47:29 -0500 (Wed, 08 Sep 2010) | 54 lines
Fixes interoperability problems with session timer behavior in Asterisk.
CHANGES:
1. Never put "timer" in "Require" header. This is not to our benefit
and RFC 4028 section 7.1 even warns against it. It is possible for one
endpoint to perform session-timer refreshes while the other endpoint does
not support them. If in this case the end point performing the refreshing
puts "timer" in the Require field during a refresh, the dialog will
likely get terminated by the other end.
2. Change the behavior of 'session-timer=accept' in sip.conf (which is
the default behavior of Asterisk with no session timer configuration
specified) to only run session-timers as result of an incoming INVITE
request if the INVITE contains an "Session-Expires" header... Asterisk is
currently treating having the "timer" option in the "Supported" header as
a request for session timers by the UAC. I do not agree with this. Session
timers should only be negotiated in "accept" mode when the incoming INVITE
supplies a "Session-Expires" header, otherwise RFC 4028 says we should
treat a request containing no "Session-Expires" header as a session with
no expiration.
Below I have outlined some situations and what Asterisk's behavior is.
The table reflects the behavior changes implemented by this patch.
SITUATIONS:
-Asterisk as UAS
1. Incoming INVITE: NO "Session-Expires"
2. Incoming INVITE: HAS "Session-Expires"
-Asterisk as UAC
3. Outgoing INVITE: NO "Session-Expires". 200 Ok Response HAS "Session-Expires" header
4. Outgoing INVITE: NO "Session-Expires". 200 Ok Response NO "Session-Expires" header
5. Outgoing INVITE: HAS "Session-Expires".
Active - Asterisk will have an active refresh timer regardless if the other endpoint does.
Inactive - Asterisk does not have an active refresh timer regardless if the other endpoint does.
XXXXXXX - Not possible for mode.
______________________________________
|SITUATIONS | 'session-timer' MODES |
|___________|________________________|
| | originate | accept |
|-----------|------------|-----------|
|1. | Active | Inactive |
|2. | Active | Active |
|3. | XXXXXXXX | Active |
|4. | XXXXXXXX | Inactive |
|5. | Active | XXXXXXXX |
--------------------------------------
(closes issue #17005)
Reported by: alexrecarey
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Brett Bryant [Wed, 8 Sep 2010 20:58:43 +0000 (20:58 +0000)]
Merged revisions 285532 via svnmerge from
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r285532 | bbryant | 2010-09-08 16:56:12 -0400 (Wed, 08 Sep 2010) | 8 lines
Fixes a bug with MeetMe where after announcing the amount of time left in a conference, if music on hold was playing, it doesn't restart.
(closes issue #17408)
Reported by: sysreq
Patches:
asterisk-issue-17408_fixed.patch uploaded by sysreq (license 1009)
Tested by: sysreq
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Jason Parker [Wed, 8 Sep 2010 20:43:10 +0000 (20:43 +0000)]
Merged revisions 285529 via svnmerge from
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r285529 | qwell | 2010-09-08 15:42:44 -0500 (Wed, 08 Sep 2010) | 1 line
Follow coding guidelines in moh rescan fix. Also fix the documentation that got me in trouble.
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Jason Parker [Wed, 8 Sep 2010 20:32:13 +0000 (20:32 +0000)]
Merged revisions 285526 via svnmerge from
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r285526 | qwell | 2010-09-08 15:31:43 -0500 (Wed, 08 Sep 2010) | 8 lines
Fixes issue where moh files were no longer rescanned during a reload.
(closes issue #16744)
Reported by: pj
Patches:
16744-reload.diff uploaded by qwell (license 4)
Tested by: qwell
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Tilghman Lesher [Wed, 8 Sep 2010 07:14:17 +0000 (07:14 +0000)]
Documentation only
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@285484
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Jason Parker [Tue, 7 Sep 2010 22:22:14 +0000 (22:22 +0000)]
Don't automatically add domains for wildcard bindaddrs.
(closes issue #17832)
Reported by: oej
Patches:
17832-wildcard.diff uploaded by qwell (license 4)
Tested by: qwell
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@285455
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Tilghman Lesher [Tue, 7 Sep 2010 21:20:16 +0000 (21:20 +0000)]
Don't notify on attribute changes, and change how the queuing mechanism works.
Fixes call spools in 1.8.
(closes issue #17337)
Reported by: loloski
Patches:
20100827__issue17337.diff.txt uploaded by tilghman (license 14)
(closes issue #17924)
Reported by: mkeuter
Tested by: mkeuter
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@285386
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Tilghman Lesher [Tue, 7 Sep 2010 21:14:03 +0000 (21:14 +0000)]
Add CHANNEL(checkhangup) to check whether a channel is in the process of being hanged up.
(closes issue #17652)
Reported by: kobaz
Patches:
func_channel.patch uploaded by kobaz (license 834)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@285373
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Richard Mudgett [Tue, 7 Sep 2010 21:08:35 +0000 (21:08 +0000)]
Fix cut-n-paste error.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@285371
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Jason Parker [Tue, 7 Sep 2010 20:58:34 +0000 (20:58 +0000)]
Add note to 'sip show settings' regarding dual-stack support, and a :: bindaddress.
(closes issue #17831)
Reported by: oej
Patches:
17831-v6wildcardbind.diff uploaded by qwell (license 4)
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Tilghman Lesher [Tue, 7 Sep 2010 20:56:07 +0000 (20:56 +0000)]
Merged revisions 285366 via svnmerge from
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r285366 | tilghman | 2010-09-07 15:31:41 -0500 (Tue, 07 Sep 2010) | 16 lines
Merged revisions 285365 via svnmerge from
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r285365 | tilghman | 2010-09-07 15:30:22 -0500 (Tue, 07 Sep 2010) | 9 lines
Catch invalid extensions at the parser, instead of making the core deal with them.
(closes issue #17794)
Reported by: PavelL
Patches:
20100820__issue17794__1.6.2.diff.txt uploaded by tilghman (license 14)
20100820__issue17794__1.4.diff.txt uploaded by tilghman (license 14)
Tested by: PavelL
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Tilghman Lesher [Tue, 7 Sep 2010 19:38:12 +0000 (19:38 +0000)]
Fix build on FreeBSD 8.0, take 2.
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Tilghman Lesher [Tue, 7 Sep 2010 19:08:09 +0000 (19:08 +0000)]
Merged revisions 285267 via svnmerge from
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r285267 | tilghman | 2010-09-07 14:07:17 -0500 (Tue, 07 Sep 2010) | 11 lines
Merged revisions 285266 via svnmerge from
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r285266 | tilghman | 2010-09-07 14:04:50 -0500 (Tue, 07 Sep 2010) | 4 lines
Use poll, if indicated to do so, in the ast_poll2 implementation.
This fixes the unit tests on FreeBSD 8.0.
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Brett Bryant [Tue, 7 Sep 2010 17:54:21 +0000 (17:54 +0000)]
Merged revisions 285196 via svnmerge from
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r285196 | bbryant | 2010-09-07 13:49:07 -0400 (Tue, 07 Sep 2010) | 17 lines
Merged revisions 285194 via svnmerge from
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r285194 | bbryant | 2010-09-07 13:45:41 -0400 (Tue, 07 Sep 2010) | 10 lines
Fixes voicemail.conf issues where mailboxes with passwords that don't precede a comma would throw unnecessary error messages.
(closes issue #15726)
Reported by: 298
Patches:
M15726.diff uploaded by junky (license 177)
Tested by: junky
Review: [full review board URL with trailing slash]
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Richard Mudgett [Tue, 7 Sep 2010 17:47:34 +0000 (17:47 +0000)]
Merged revisions 285193 via svnmerge from
https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
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Merged revisions 285192 via svnmerge from
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r285192 | rmudgett | 2010-09-07 11:58:57 -0500 (Tue, 07 Sep 2010) | 8 lines
COLP/CONP and chan_misdn missing update
chan_misdn does not update the caller id of the channel if a new connected
number or ECT-INFORM (w/ new peer number on call transfer) is received.
JIRA ABE-2502
JIRA SWP-2058
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Russell Bryant [Mon, 6 Sep 2010 20:10:24 +0000 (20:10 +0000)]
regenerate configure script.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@285162
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Russell Bryant [Mon, 6 Sep 2010 20:10:03 +0000 (20:10 +0000)]
Fix libsrtp -fPIC check for when non-standard prefix is used.
Thanks to loompek in #asterisk for reporting the issue and testing this patch.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@285161
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Tilghman Lesher [Mon, 6 Sep 2010 06:56:07 +0000 (06:56 +0000)]
Merged revisions 285089 via svnmerge from
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r285089 | tilghman | 2010-09-06 01:55:17 -0500 (Mon, 06 Sep 2010) | 9 lines
Merged revisions 285088 via svnmerge from
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r285088 | tilghman | 2010-09-06 01:54:18 -0500 (Mon, 06 Sep 2010) | 2 lines
Silly convenience script for BSD platforms.
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Russell Bryant [Sat, 4 Sep 2010 18:08:19 +0000 (18:08 +0000)]
Add a C++ compatible version of AST_CLI_DEFINE().
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@285057
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Terry Wilson [Fri, 3 Sep 2010 23:19:54 +0000 (23:19 +0000)]
Call correct lock function as transferer is a sip_pvt not a channel
Both functions are #defined to ao2_lock, but still...
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@285017
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David Vossel [Fri, 3 Sep 2010 22:21:50 +0000 (22:21 +0000)]
Disables auth_options_request option by default.
The auth_options_request option was created to do authentication
on OPTIONS request just like INVITES are done. Since it has been
noted that some endpoints use OPTIONS requests as a way of qualifying
a peer and that a 401 authentication response could result in
interoperability issues, this option has been disabled by default.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@285006
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Brett Bryant [Fri, 3 Sep 2010 18:19:53 +0000 (18:19 +0000)]
Merged revisions 284958 via svnmerge from
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r284958 | bbryant | 2010-09-03 14:15:49 -0400 (Fri, 03 Sep 2010) | 8 lines
This is a patch provided for issue #17935 to add the ActionID to the IAXregistry AMI response.
(closes issue #17935)
Reported by: alexkuklin
Patches:
iaxshowreg uploaded by alexkuklin (license 1115)
Tested by: alexkuklin
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David Vossel [Fri, 3 Sep 2010 18:03:23 +0000 (18:03 +0000)]
During OPTIONS authentication, the authpeer does not need to be returned for any reason.
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David Vossel [Fri, 3 Sep 2010 17:29:02 +0000 (17:29 +0000)]
authenticate OPTIONS requests just like we would an INVITE
OPTIONS requests should be treated the same as an INVITE
This includes authentication. This patch adds the ability for
incoming out of dialog OPTION requests to be authenticated
before providing a response indicating whether an extension
is available or not. The authentication routine works the
exact same way as it does for incoming INVITEs. This means
that if a peer has 'insecure=invite' in their peer definition,
the same will be true for the processing of the OPTIONS request.
Review: https://reviewboard.asterisk.org/r/881/
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Terry Wilson [Fri, 3 Sep 2010 16:28:18 +0000 (16:28 +0000)]
Merged revisions 284897 via svnmerge from
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r284897 | twilson | 2010-09-03 11:20:45 -0500 (Fri, 03 Sep 2010) | 12 lines
Merged revisions 284881 via svnmerge from
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r284881 | twilson | 2010-09-03 11:10:23 -0500 (Fri, 03 Sep 2010) | 5 lines
Properly detect when a sound file doesn't exist
ast_fileexists returns -1 for error and 0 for a non-existant file. The existing
code treated missing files as though they existed.
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Jan Kalab [Fri, 3 Sep 2010 13:07:39 +0000 (13:07 +0000)]
Calendar categories and priorities: strdupa() fix
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Jan Kalab [Fri, 3 Sep 2010 12:48:59 +0000 (12:48 +0000)]
Fix for calendar categories and priorities according to ISO C90
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Jan Kalab [Fri, 3 Sep 2010 11:27:04 +0000 (11:27 +0000)]
Support for calendar events priorities and categories
Review 880
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Brett Bryant [Thu, 2 Sep 2010 21:04:51 +0000 (21:04 +0000)]
Merged revisions 284778 via svnmerge from
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r284778 | bbryant | 2010-09-02 16:54:33 -0400 (Thu, 02 Sep 2010) | 14 lines
Merged revisions 284777 via svnmerge from
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r284777 | bbryant | 2010-09-02 16:25:03 -0400 (Thu, 02 Sep 2010) | 7 lines
Fixes a bug in manager.c where the default configuration values weren't reset when the manager configuration was reloaded.
(closes issue #17917)
Reported by: lmadsen
Review: https://reviewboard.asterisk.org/r/883/
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Richard Mudgett [Thu, 2 Sep 2010 21:02:54 +0000 (21:02 +0000)]
Simplified pri_dchannel() poll timeout duration code.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@284780
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Richard Mudgett [Thu, 2 Sep 2010 20:59:12 +0000 (20:59 +0000)]
Made output libpri event names if pri debugging is enabled when sig_pri processes them.
* Simplified CLI "pri debug xx span xx" command code and removed redundant
debugging enabled messages.
* Made CLI "pri debug xx span xx" command only close the debugging log
file if it was opened.
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David Vossel [Thu, 2 Sep 2010 16:56:43 +0000 (16:56 +0000)]
Merged revisions 284704 via svnmerge from
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r284704 | dvossel | 2010-09-02 11:48:51 -0500 (Thu, 02 Sep 2010) | 13 lines
Merged revisions 284703 via svnmerge from
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r284703 | dvossel | 2010-09-02 11:47:15 -0500 (Thu, 02 Sep 2010) | 7 lines
Removed relatedpeer code from sip_autodestruct
Handling of the relatedpeer structure associated with a
sip_pvt should be done during the final sip_destruction
function, not in sip_autodestruct.
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Jason Parker [Thu, 2 Sep 2010 16:43:09 +0000 (16:43 +0000)]
Add slin16 support for format_wav (new wav16 file extension)
(closes issue #15029)
Reported by: andrew
Patches:
wav16.patch uploaded by andrew (license 240)
Tested by: qwell, andrew
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@284701
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Richard Mudgett [Thu, 2 Sep 2010 16:34:32 +0000 (16:34 +0000)]
Added documentation for CONNECTEDLINE and REDIRECTING functions.
(closes issue #17808)
Reported by: jtodd
Review: https://reviewboard.asterisk.org/r/875/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@284698
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Tilghman Lesher [Thu, 2 Sep 2010 16:27:52 +0000 (16:27 +0000)]
Fixing build
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Tilghman Lesher [Thu, 2 Sep 2010 16:11:15 +0000 (16:11 +0000)]
Merged revisions 284665 via svnmerge from
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r284665 | tilghman | 2010-09-02 11:07:19 -0500 (Thu, 02 Sep 2010) | 2 lines
Fixing build.
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Tilghman Lesher [Thu, 2 Sep 2010 05:31:02 +0000 (05:31 +0000)]
Merged revisions 284631 via svnmerge from
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r284631 | tilghman | 2010-09-02 00:30:16 -0500 (Thu, 02 Sep 2010) | 7 lines
Don't reset queue stats on a module reload.
(closes issue #17535)
Reported by: raarts
Patches:
20100819__issue17535.diff.txt uploaded by tilghman (license 14)
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Tilghman Lesher [Thu, 2 Sep 2010 05:20:59 +0000 (05:20 +0000)]
When optional_api is non-optional, force dependent modules to be loaded.
(closes issue #17707)
Reported by: ira
Patches:
20100819__issue17707__asterisk1.8.diff.txt uploaded by tilghman (license 14)
Tested by: tilghman
Review: https://reviewboard.asterisk.org/r/876/
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Tilghman Lesher [Thu, 2 Sep 2010 05:00:34 +0000 (05:00 +0000)]
Merged revisions 284593,284595 via svnmerge from
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r284593 | tilghman | 2010-09-01 17:59:50 -0500 (Wed, 01 Sep 2010) | 18 lines
Merged revisions 284478 via svnmerge from
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r284478 | tilghman | 2010-09-01 13:49:11 -0500 (Wed, 01 Sep 2010) | 11 lines
Ensure that all areas that previously used select(2) now use poll(2), with implementations that need poll(2) implemented with select(2) safe against 1024-bit overflows.
This is a followup to the fix for the pthread timer in 1.6.2 and beyond, fixing
a potential crash bug in all supported releases.
(closes issue #17678)
Reported by: russell
Branch: https://origsvn.digium.com/svn/asterisk/team/tilghman/ast_select
Review: https://reviewboard.asterisk.org/r/824/
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r284595 | tilghman | 2010-09-01 22:57:43 -0500 (Wed, 01 Sep 2010) | 2 lines
Failed to rerun bootstrap.sh after last commit
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David Vossel [Wed, 1 Sep 2010 21:47:01 +0000 (21:47 +0000)]
During request to dialog matching, verify init_ruri is present before comparing.
During request to dialog matching, we attempt a best effort routine for fork
detection which requires several elements to be in place. The dialog's
initial request uri is one of those elements. Since it is best effort,
if the init_ruri is not present for some reason we can not proceed with that
routine.
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Terry Wilson [Wed, 1 Sep 2010 18:44:36 +0000 (18:44 +0000)]
Fix SRTP for changing SSRC and multiple a=crypto SDP lines
Adding code to Asterisk that changed the SSRC during bridges and masquerades
broke SRTP functionality. Also broken was handling the situation where an
incoming INVITE had more than one crypto offer. This patch caches the SRTP
policies the we use so that we can change the ssrc and inform libsrtp of the
new streams. It also uses the first acceptable a=crypto line from the incoming
INVITE.
(closes issue #17563)
Reported by: Alexcr
Patches:
srtp.diff uploaded by twilson (license 396)
Tested by: twilson
Review: https://reviewboard.asterisk.org/r/878/
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Tilghman Lesher [Wed, 1 Sep 2010 18:16:37 +0000 (18:16 +0000)]
Merged revisions 284472 via svnmerge from
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r284472 | tilghman | 2010-09-01 13:13:35 -0500 (Wed, 01 Sep 2010) | 5 lines
Don't warn on floats and timestamps
(closes issue #17082)
Reported by: coolmig
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Tilghman Lesher [Tue, 31 Aug 2010 20:22:10 +0000 (20:22 +0000)]
Merged revisions 284399 via svnmerge from
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r284399 | tilghman | 2010-08-31 15:18:32 -0500 (Tue, 31 Aug 2010) | 14 lines
Merged revisions 284393 via svnmerge from
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r284393 | tilghman | 2010-08-31 15:13:21 -0500 (Tue, 31 Aug 2010) | 7 lines
Don't send a devstate change on poke_noanswer if the state did not change.
(closes issue #17741)
Reported by: schmidts
Patches:
chan_sip.c.patch uploaded by schmidts (license 1077)
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Leif Madsen [Tue, 31 Aug 2010 19:00:15 +0000 (19:00 +0000)]
Merged revisions 284317 via svnmerge from
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r284317 | lmadsen | 2010-08-31 13:59:31 -0500 (Tue, 31 Aug 2010) | 15 lines
Merged revisions 284316 via svnmerge from
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r284316 | lmadsen | 2010-08-31 13:57:59 -0500 (Tue, 31 Aug 2010) | 7 lines
Update say.conf.sample to match the rules in say.c
(closes issue #17835)
Reported by: RoadKill
Patches:
say.conf.sample.patch.rules uploaded by RoadKill (license 933)
Tested by: RoadKill
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Tilghman Lesher [Mon, 30 Aug 2010 22:28:47 +0000 (22:28 +0000)]
Merged revisions 284280 via svnmerge from
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r284280 | tilghman | 2010-08-30 17:27:06 -0500 (Mon, 30 Aug 2010) | 11 lines
Fix 3 coding errors:
1) After we close FD, we should not be trying to write to it.
2) Call _exit(0), not exit(0), to avoid running shutdown routines in a child.
3) Use endian, not processor, detection to ensure bytes are written in the correct order.
(closes issue #15706)
Reported by: modelnine
Patches:
asterisk-1.6.1.1-festival-debug.patch uploaded by modelnine (license 865)
Tested by: gmartinez
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Tilghman Lesher [Sun, 29 Aug 2010 07:05:27 +0000 (07:05 +0000)]
Missed adding this file
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Tilghman Lesher [Sun, 29 Aug 2010 05:17:37 +0000 (05:17 +0000)]
Also ignore the checksums
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Tilghman Lesher [Sun, 29 Aug 2010 02:51:14 +0000 (02:51 +0000)]
Rename CEL adaptive driver to plain driver, since there isn't another ODBC driver
(and the other CEL drivers have adaptive capabilities, anyway).
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@284096
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Russell Bryant [Sat, 28 Aug 2010 21:29:45 +0000 (21:29 +0000)]
Be more flexible with whitespace on AMI action headers.
Previously, this code required exactly one space to be after the ':' in headers
for an AMI action. This now makes whitespace optional, and allows whitespace that
is there to vary in amount.
(closes issue #17862)
Reported by: cmoye
Patches:
manager.c.patch_trunk uploaded by cmoye (license 858)
manager.c.patch_1.8 uploaded by cmoye (license 858)
Tested by: cmoye
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David Vossel [Fri, 27 Aug 2010 22:37:11 +0000 (22:37 +0000)]
Merged revisions 284002 via svnmerge from
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r284002 | dvossel | 2010-08-27 17:27:50 -0500 (Fri, 27 Aug 2010) | 14 lines
Merged revisions 283960 via svnmerge from
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r283960 | dvossel | 2010-08-27 17:17:26 -0500 (Fri, 27 Aug 2010) | 8 lines
Parse all "Accept" headers for SIP SUBSCRIBE requests.
(closes issue #17758)
Reported by: ibc
Patches:
multiple_accept_headers_1.4.diff uploaded by dvossel (license 671)
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Russell Bryant [Fri, 27 Aug 2010 21:33:55 +0000 (21:33 +0000)]
Print exten@context:priority in verbose messages from pbx_realtime.
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projects / thirdparty / asterisk.git / log
