Age | Commit message (Collapse) | Author | Files | Lines |
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Adds sublines to skinny lines. Each subline can be attached to an
SLA station/trunk combo. Includes the following functionality:
Callid is persistent for both in/out calls on all skinny devices.
Can join, hold, resume.
All sublines appear under a single line button.
See: https://wiki.asterisk.org/wiki/display/~wedhorn/Skinny+SLA for doc.
(closes issue ASTERISK-17947)
Review: https://reviewboard.asterisk.org/r/1239/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@328381 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.10
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r328329 | rmudgett | 2011-07-14 19:19:32 -0500 (Thu, 14 Jul 2011) | 2 lines
Make hint watcher callback take const strings for context and exten parameters.
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@328344 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.10
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r328247 | lmadsen | 2011-07-14 16:25:31 -0400 (Thu, 14 Jul 2011) | 14 lines
Merged revisions 328209 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r328209 | lmadsen | 2011-07-14 16:13:06 -0400 (Thu, 14 Jul 2011) | 6 lines
Introduce <support_level> tags in MODULEINFO.
This change introduces MODULEINFO into many modules in Asterisk in order to show
the community support level for those modules. This is used by changes committed
to menuselect by Russell Bryant recently (r917 in menuselect). More information about
the support level types and what they mean is available on the wiki at
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@328259 f38db490-d61c-443f-a65b-d21fe96a405b
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Autoanswer added to skinny based on incoming chan var SKINNY_AUTOANSWER.
Initial value must be the time to autoanswer in ms, then optionally :BEEP
to play a tone when answered and :MUTE to mute the mic when answering.
eg 3000:MUTE:BEEP will ring for 3 secs, then answer, mute the mic, and
play a beep. just 3000 would answer afer 3 secs of ringing with no
beep and full two way audio.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@322544 f38db490-d61c-443f-a65b-d21fe96a405b
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The do_monitor seemed to be there for task scheduling and network monitoring. However, the network monitoring has a dedicated thread so the ast_io_wait was basically just a usleep as it didn't actually seem to be monitoring anything.
Review: https://reviewboard.asterisk.org/r/1256/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@322381 f38db490-d61c-443f-a65b-d21fe96a405b
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The vars were either explicitly or implicitly not used.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@319470 f38db490-d61c-443f-a65b-d21fe96a405b
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Probably haven't been working for a couple of years. May still need
some more love, but they are now working, both as a hint device and
monitoring a hint. Changes centre around the long ago change
to remove the requirement for a device name in a skinny line, and
changes to the transmit_* functions.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@319316 f38db490-d61c-443f-a65b-d21fe96a405b
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When called, activatesub first cleans up the active sub and then
handles the sub passed. dialandactivatesub first sets sub->exten
and then calls activatesub. Revise handle_offhook to utilise the
callid sent to chan_skinny. Some other minor fixes especially around
d->hookstate (which still needs some more work).
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@319024 f38db490-d61c-443f-a65b-d21fe96a405b
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There were some issues where if a simple switch was cancelled and a
new switch started before the first had timed out where the d->exten
would be used for both subchannels. This was bad leading to possible
invalid extensions if some digits had been entered in the abandoned
simple switch and the second one was completed before the first timed
out, or the second would be cancelled because d->exten would be set to
nothing on the time out of the first.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@318833 f38db490-d61c-443f-a65b-d21fe96a405b
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Consolidate the functions and add some debugging info. Allows to be
able to set a substate without explicitly knowing what the state is.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@318635 f38db490-d61c-443f-a65b-d21fe96a405b
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Add the setsubstate_onhook to complete the initial substate handling
procedures. Added dumpsub(sub, forcehangup) which is the common way of
calling setsubstate_onhook. Dumpsub attempts to activate another sub
after setting the current one onhook.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@318600 f38db490-d61c-443f-a65b-d21fe96a405b
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If a call is made to a line that already has a call and the device is
offhook (ie activeish call), the call is set to CALLWAIT rather than RINGIN.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@318106 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #17901)
Reported by: salecha
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@318019 f38db490-d61c-443f-a65b-d21fe96a405b
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Improve readability of code, eg, (sub->parent == d->activeline) becomes
(sub->line == d->activeline).
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@318018 f38db490-d61c-443f-a65b-d21fe96a405b
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Long time coming, finally moving the hookstate from line to device.
This may fix some issues where a device has multiple lines. Previously
we had to run through all lines on a device to see if it was actually
onhook or not.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@317996 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.8
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r317478 | russell | 2011-05-05 17:53:45 -0500 (Thu, 05 May 2011) | 12 lines
Fix some consistency issues with jitterbuffer config.
Store the defaults noted in the sample config files in the jitterbuffer config
data structure. This makes the CLI commands that output these settings show
the right thing. Also only show the settings that are relevant in the settings
CLI commands, based on which jitterbuffer is selected and whether it's enabled.
(closes issue #19083)
Reported by: rgagnon
Patches:
issue-19083-trunk-r313139.diff uploaded by rgagnon (license 1202)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@317479 f38db490-d61c-443f-a65b-d21fe96a405b
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skinny_hold moved to setsubstate_hold and skinny_unhold integrated into
setsubstate_connected. Removed sub->onhold and replaced with
SUBSTATE_HOLD.
Also fixed inbound call answering by queueing an AST_CONTROL_ANSWER on
answering a SUBSTATE_RINGIN sub (was a typo).
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@317382 f38db490-d61c-443f-a65b-d21fe96a405b
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Move handling of both state handling from skinny_indicate to it's own sub.
Also, modified behaviour to not hangup the sub and let the dialplan
have a chance in doing what it wants for congestion. Added various states to
substate2str and added these states where applicable for other set_substate_
procs.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@317026 f38db490-d61c-443f-a65b-d21fe96a405b
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Move handling of setting busy state from skinny_indicate to it's own sub.
Also, modified behaviour to not hangup the sub and let the dialplan
have a chance in doing what it wants (eg busy(10); hangup() in the dialplan
now gives a busy indication for 10 secs and then hangs up.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@316994 f38db490-d61c-443f-a65b-d21fe96a405b
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Renamed previous setsubstate_ringout to setsubstate_dialing for a state
when attempting to dial a number, substate ringout now for when core
has indicated that the channel is actually ringing on the other end.
Also added substate2str for debugging purposes.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@316885 f38db490-d61c-443f-a65b-d21fe96a405b
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Added setsubstate_ringin. skinny_call now calls sss_ringin rather than inline.
Fixed previous issue so that setsubstate_connected now use SUBSTATE_RINGIN
to determine is an AST_CONTROL_ANSWER should be queued.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@316584 f38db490-d61c-443f-a65b-d21fe96a405b
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Cosolidated the code so that skinny_answer now uses the setsubstate procedures
rather than doing the handling inline.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@316552 f38db490-d61c-443f-a65b-d21fe96a405b
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Cosolidated the working out of the callinfo to be sent into
transmit_callinfo. Replaced ambiguous sub->outgoing with calldirection
which can be SKINNY_INCOMING or SKINNY_OUTGOING (same value as the
skinny protocol).
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@316519 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.8
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r316336 | russell | 2011-05-03 17:13:31 -0500 (Tue, 03 May 2011) | 8 lines
Use htons() instead of ntohs() in some places.
(closes issue #19200)
Reported by: wdoekes
Patches:
issue19200-trunk.patch uploaded by wdoekes (license 717)
issue19200-1.8.x.patch uploaded by wdoekes (license 717)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@316337 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.8
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r316265 | russell | 2011-05-03 14:55:49 -0500 (Tue, 03 May 2011) | 5 lines
Fix a bunch of compiler warnings generated by gcc 4.6.0.
Most of these are -Wunused-but-set-variable, but there were a few others
mixed in here, as well.
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@316293 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.8
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r314628 | mnicholson | 2011-04-21 13:24:05 -0500 (Thu, 21 Apr 2011) | 27 lines
Merged revisions 314620 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r314620 | mnicholson | 2011-04-21 13:22:19 -0500 (Thu, 21 Apr 2011) | 20 lines
Merged revisions 314607 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r314607 | mnicholson | 2011-04-21 13:19:21 -0500 (Thu, 21 Apr 2011) | 14 lines
Added limits to the number of unauthenticated sessions TCP based protocols are allowed to have open simultaneously. Also added timeouts for unauthenticated sessions where it made sense to do so.
Unrelated, the manager interface now properly checks if the user has the "system" privilege before executing shell commands via the Originate action.
AST-2011-005
AST-2011-006
(closes issue #18787)
Reported by: kobaz
(related to issue #18996)
Reported by: tzafrir
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@314666 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #17907)
Reported by: wedhorn
Patches:
cleanup.stateringout.diff uploaded by wedhorn (license 30)
Tested by: salecha, wedhorn
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@313980 f38db490-d61c-443f-a65b-d21fe96a405b
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audio ConfBridge, and other stuff
-Functional changes
1. Dynamic global format list build by codecs defined in codecs.conf
2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf
3. Negotiation of SILK attributes in chan_sip.
4. SPEEX 32khz with translation
5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation
using codec_resample.c
6. Various changes to RTP code required to properly handle the dynamic format list
and formats with attributes.
7. ConfBridge now dynamically jumps to the best possible sample rate. This allows
for conferences to take advantage of HD audio (Which sounds awesome)
8. Audiohooks are no longer limited to 8khz audio, and most effects have been
updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT.
9. codec_resample now uses its own code rather than depending on libresample.
-Organizational changes
Global format list is moved from frame.c to format.c
Various format specific functions moved from frame.c to format.c
Review: https://reviewboard.asterisk.org/r/1104/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@308582 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #18556)
Reported by: kkm
Review: https://reviewboard.asterisk.org/r/1071/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@306258 f38db490-d61c-443f-a65b-d21fe96a405b
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This patch is the foundation of an entire new way of looking at media in Asterisk.
The code present in this patch is everything required to complete phase1 of my
Media Architecture proposal. For more information about this project visit the link below.
https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal
The primary function of this patch is to convert all the usages of format
bitfields in Asterisk to use the new format and format_cap APIs. Functionally
no change in behavior should be present in this patch. Thanks to twilson
and russell for all the time they spent reviewing these changes.
Review: https://reviewboard.asterisk.org/r/1083/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@306010 f38db490-d61c-443f-a65b-d21fe96a405b
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Previously, I had added the ast_sched_thread stuff that was a generic scheduler
thread implementation. However, if you used it, it required using different
functions for modifying scheduler contents. This patch reworks how this is
done and just allows you to optionally start a thread on the original scheduler
context structure that has always been there. This makes it trivial to switch
to the generic scheduler thread implementation without having to touch any of
the other code that adds or removes scheduler entries.
In passing, I made some naming tweaks to add ast_ prefixes where they were not
there before.
Review: https://reviewboard.asterisk.org/r/1007/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@299091 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.8
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r287643 | qwell | 2010-09-20 16:29:46 -0500 (Mon, 20 Sep 2010) | 15 lines
Merged revisions 287642 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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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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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@287644 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #17887)
Reported by: salecha
Patches:
skinny.redial.diff uploaded by wedhorn (license 30)
Tested by: wedhorn, salecha
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@283495 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@283289 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@283288 f38db490-d61c-443f-a65b-d21fe96a405b
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Also correct the callinfo provided in skinny_answer.
(closes issue #17876)
Reported by: salecha
Patches:
skinny_cnd3.diff uploaded by wedhorn (license 30)
Tested by: salecha, wedhorn
Review: NA
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@283287 f38db490-d61c-443f-a65b-d21fe96a405b
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Consolidates all offhook (new call with dialtone) to setsubstate_offhook. This should be roughly equivalent to existing code, although a couple of calls now run through the full offhook sequence rather than an abbreviated one.
(closes issue #17812)
Reported by: wedhorn
Patches:
cleanup.stateoffhook.diff uploaded by wedhorn (license 30)
Tested by: salecha, wedhorn
Review: NA
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@282701 f38db490-d61c-443f-a65b-d21fe96a405b
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Cleaned up handling of onhook indications and added indications if more than one sub on device. Also fixes issue in 12324 so that the phone can call itself without locking up.
(closes issue #17692)
Reported by: jmhunter
Patches:
chan_skinny-transfer-v4.txt uploaded by DEA (license 3)
skinnytransfver.v8.diff uploaded by wedhorn (license 30)
Tested by: jmhunter, salecha, wedhorn
Review: NA
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@281257 f38db490-d61c-443f-a65b-d21fe96a405b
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Move call answering stuff into new setsubstate_connected. Also add sub->substate var and set it to SUBSTATE_CONNECTED in setsubstate_connected.
(closes issue #17772)
Reported by: wedhorn
Patches:
cleanup.stateconnected2.diff uploaded by wedhorn (license 30)
Tested by: wedhorn, salecha
Review: NA
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@281227 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #17770)
Reported by: salecha
Patches:
skinny.answercrash.diff uploaded by wedhorn (license 30)
Tested by: salecha
Review: NA
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@281226 f38db490-d61c-443f-a65b-d21fe96a405b
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Moved inline packet sending to transmit_ subs. Removed handle_keep_alive and handle_register_message to inline in handle_message. Also moved transmit_response(d) to transmit_response_bysessions(s) and created a wrapper transmit_response(d) that calls transmit_response_bysession(d->session).
(closes issue #16980)
Reported by: wedhorn
Patches:
skinny-clean06b.diff uploaded by wedhorn (license 30)
Tested by: wedhorn, DEA
Review: NA
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@280589 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@278132 f38db490-d61c-443f-a65b-d21fe96a405b
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ACLs can now be configured to match IPv6 networks. This is only
relevant for ACLs in chan_sip for now since other channel drivers
do not support IPv6 addressing. However, once those channel drivers
are outfitted to support IPv6 addressing, the ACLs will already be
ready for IPv6 support.
https://reviewboard.asterisk.org/r/791
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@277814 f38db490-d61c-443f-a65b-d21fe96a405b
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Expand the ani field in ast_party_caller and ast_party_connected_line to
an ast_party_id.
This is an extension to the ast_callerid restructuring patch in review:
https://reviewboard.asterisk.org/r/702/
Review: https://reviewboard.asterisk.org/r/744/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276393 f38db490-d61c-443f-a65b-d21fe96a405b
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The purpose of this patch is to eliminate struct ast_callerid since it has
turned into a miscellaneous collection of various party information.
Eliminate struct ast_callerid and replace it with the following struct
organization:
struct ast_party_name {
char *str;
int char_set;
int presentation;
unsigned char valid;
};
struct ast_party_number {
char *str;
int plan;
int presentation;
unsigned char valid;
};
struct ast_party_subaddress {
char *str;
int type;
unsigned char odd_even_indicator;
unsigned char valid;
};
struct ast_party_id {
struct ast_party_name name;
struct ast_party_number number;
struct ast_party_subaddress subaddress;
char *tag;
};
struct ast_party_dialed {
struct {
char *str;
int plan;
} number;
struct ast_party_subaddress subaddress;
int transit_network_select;
};
struct ast_party_caller {
struct ast_party_id id;
char *ani;
int ani2;
};
The new organization adds some new information as well.
* The party name and number now have their own presentation value that can
be manipulated independently. ISDN supplies the presentation value for
the name and number at different times with the possibility that they
could be different.
* The party name and number now have a valid flag. Before this change the
name or number string could be empty if the presentation were restricted.
Most channel drivers assume that the name or number is then simply not
available instead of indicating that the name or number was restricted.
* The party name now has a character set value. SIP and Q.SIG have the
ability to indicate what character set a name string is using so it could
be presented properly.
* The dialed party now has a numbering plan value that could be useful to
have available.
The various channel drivers will need to be updated to support the new
core features as needed. They have simply been converted to supply
current functionality at this time.
The following items of note were either corrected or enhanced:
* The CONNECTEDLINE() and REDIRECTING() dialplan functions were
consolidated into func_callerid.c to share party id handling code.
* CALLERPRES() is now deprecated because the name and number have their
own presentation values.
* Fixed app_alarmreceiver.c write_metadata(). The workstring[] could
contain garbage. It also can only contain the caller id number so using
ast_callerid_parse() on it is silly. There was also a typo in the
CALLERNAME if test.
* Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id
number string. ast_callerid_parse() alters the given buffer which in this
case is the channel's caller id number string. Then using
ast_shrink_phone_number() could alter it even more.
* Fixed caller ID name and number memory leak in chan_usbradio.c.
* Fixed uninitialized char arrays cid_num[] and cid_name[] in
sig_analog.c.
* Protected access to a caller channel with lock in chan_sip.c.
* Clarified intent of code in app_meetme.c sla_ring_station() and
dial_trunk(). Also made save all caller ID data instead of just the name
and number strings.
* Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge()
function.
* Corrected some weirdness with app_privacy.c's use of caller
presentation.
Review: https://reviewboard.asterisk.org/r/702/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276347 f38db490-d61c-443f-a65b-d21fe96a405b
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This adds a generic API for accommodating IPv6 and IPv4 addresses
within Asterisk. While many files have been updated to make use of the
API, chan_sip and the RTP code are the files which actually support
IPv6 addresses at the time of this commit. The way has been paved for
easier upgrading for other files in the near future, though.
Big thanks go to Simon Perrault, Marc Blanchet, and Jean-Philippe Dionne
for their hard work on this.
(closes issue #17565)
Reported by: russell
Patches:
asteriskv6-test-report.pdf uploaded by russell (license 2)
Review: https://reviewboard.asterisk.org/r/743
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@274783 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #17126)
Reported by: wedhorn
Patches:
skinny79xx_redial1.diff uploaded by wedhorn (license 30)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@255851 f38db490-d61c-443f-a65b-d21fe96a405b
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Bulk lot of generally trivial changes for cleaning up the transmit stuff. Line state request has been modified for line only responses.
(closes issue #16994)
Reported by: wedhorn
Patches:
skinny-clean07.diff uploaded by wedhorn (license 30)
Tested by: wedhorn
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@255850 f38db490-d61c-443f-a65b-d21fe96a405b
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This change basically reverts the change reviewed in
https://reviewboard.asterisk.org/r/374/ and instead limits the
updating of the RTP synchronization source to only those times when we
detect that the other side of the conversation has changed the ssrc.
The problem is that SRCUPDATE control frames are sent many times where
we don't want a new ssrc, including whenever Asterisk has to send DTMF
in a normal bridge. This is also not the first time that this mistake
has been made. The initial implementation of the ast_rtp_new_source
function also changed the ssrc--and then it was removed because of
this same issue. Then, we put it back in again to fix a different
issue. This patch attempts to only change the ssrc when we see that
the other side of the conversation has changed the ssrc.
It also renames some functions to make their purpose more clear.
Review: https://reviewboard.asterisk.org/r/540/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@252089 f38db490-d61c-443f-a65b-d21fe96a405b
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Small patch changing skinny_set_rtp_peer to use transmit_stopmediatransmission and to use new transmit_startmediatransmission.
Basic testing on 30VIP's by wedhorn
Basic testing on 7960 by me
(closes issue #16956)
Reported by: wedhorn
Patches:
skinny-clean05b.diff uploaded by wedhorn (license 30)
Tested by: wedhorn,mvanbaak
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@251222 f38db490-d61c-443f-a65b-d21fe96a405b
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