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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r277568 | tilghman | 2010-07-16 16:54:29 -0500 (Fri, 16 Jul 2010) | 8 lines
Since we split values at the semicolon, we should store values with a semicolon as an encoded value.
(closes issue #17369)
Reported by: gkservice
Patches:
20100625__issue17369.diff.txt uploaded by tilghman (license 14)
Tested by: tilghman
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@277773 f38db490-d61c-443f-a65b-d21fe96a405b
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xmllint seems to be more commonly available since it comes with libxml2.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@277703 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issues #17667)
Reported by: snuffy
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@277667 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r277625 | tringenbach | 2010-07-16 17:43:39 -0500 (Fri, 16 Jul 2010) | 9 lines
Save and restore AST_FLAG_BRIDGE_HANGUP_DONT on attended transfer.
ast_bridge_call() clears AST_FLAG_BRIDGE_HANGUP_DONT. But during an attended
transfer, ast_bridge_call() is called for a second bridge on the same channel,
and it clears that flag, which still needs to get set for when the original
ast_bridge_call() gets control back and checks it.
Review: https://reviewboard.asterisk.org/r/741
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@277657 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r277497 | mnicholson | 2010-07-16 16:18:38 -0500 (Fri, 16 Jul 2010) | 4 lines
Default to no udptl error correction so that error correction will be disabled in the event that the remote end indicates that they do not support the error correction mode we requested.
FAX-128
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@277530 f38db490-d61c-443f-a65b-d21fe96a405b
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Just say the number of seconds (after minutes) rather than doing some incorrect
calculation with respect to minutes.
(closes issue #17498)
Reported by: corruptor
Patches:
holdesecs_bug.diff uploaded by corruptor (license 253)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@277488 f38db490-d61c-443f-a65b-d21fe96a405b
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cancelling lagid.
No, replacing usleep(1) with sched_yield() did not have an effect.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@277484 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r277419 | rmudgett | 2010-07-16 15:18:54 -0500 (Fri, 16 Jul 2010) | 15 lines
priexclusive in chan_dahdi.conf ignored when reloading dahdi module
During a reload, the priexclusive and outsignalling parameters are not
read in from the config file as intended. Unfortunately, they get set to
defaults as a result. This patch makes sure that they do not get set to
defaults during a reload.
(closes issue #17441)
Reported by: mtryfoss
Patches:
issue17441_v1.4.patch uploaded by rmudgett (license 664)
issue17441_v1.6.2.patch uploaded by rmudgett (license 664)
issue17441_trunk.patch uploaded by rmudgett (license 664)
Tested by: rmudgett
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@277467 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@277452 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@277409 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #17471)
Reported by: jazzy
Patches:
app_queue.c.diff uploaded by jazzy (license 1056)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@277366 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r277327 | mnicholson | 2010-07-16 13:30:22 -0500 (Fri, 16 Jul 2010) | 8 lines
Interpret device state AST_DEVICE_UNKNOWN as extension state AST_EXTENSION_NOT_INUSE.
(closes issue #16035)
Reported by: francesco_r
Patches:
pbx.c.patch uploaded by viniciusfontes (license 978)
Tested by: francesco_r, agx, lawbar
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@277331 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r277261 | tilghman | 2010-07-16 13:04:11 -0500 (Fri, 16 Jul 2010) | 5 lines
If variable gotten is not set, will segfault on Solaris.
(closes issue #17636)
Reported by: bklang
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@277263 f38db490-d61c-443f-a65b-d21fe96a405b
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(fix build breakage introduced in r277250)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@277262 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r277247 | mnicholson | 2010-07-16 12:29:57 -0500 (Fri, 16 Jul 2010) | 4 lines
For pass through DTMF tones, measure the actual duration between the begin and end packets on the wire. If it is detected to be less than AST_MIN_DTMF_DURATION, trigger dtmf emulation.
AST-362
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@277250 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r277182 | pabelanger | 2010-07-16 13:10:36 -0400 (Fri, 16 Jul 2010) | 8 lines
Total analysis time error with SIP and silence suppression
When using app_amd with SIP providers that have silence
suppression on, the iTotalTime count increases exponentially.
(closes issue #17656)
Reported by: juls
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@277183 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@277175 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #17092)
Reported by: moy
Patches:
translate.rev254273.patch uploaded by moy (license 222)
Tested by: moy
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@277143 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@277103 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@277102 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@277065 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@277028 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@277027 f38db490-d61c-443f-a65b-d21fe96a405b
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Found a unused bag of curly brackets under my table. I always wondered where
they had gone. They where indeed needed in chan_sip.c
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276989 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276952 f38db490-d61c-443f-a65b-d21fe96a405b
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sip.conf configuration for the channel and for devices.
The Max-Forwards header is used to prevent loops in a SIP network. Each intermediary,
like SIP proxys and SBCs, decrement this counter and detects when it reaches zero,
at which point the SIP request is nicely killed in a SIP-friendly way.
Review: https://reviewboard.asterisk.org/r/778/
Thanks to dvossel for the review and good advice.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276951 f38db490-d61c-443f-a65b-d21fe96a405b
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Review: https://reviewboard.asterisk.org/r/777/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276950 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276911 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276910 f38db490-d61c-443f-a65b-d21fe96a405b
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Found based on message from Philip Prindeville on the
Asterisk Developers mailing list.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276909 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276908 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276871 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276870 f38db490-d61c-443f-a65b-d21fe96a405b
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Also, update meetme to the full list of supported fields.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276869 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #17654)
Reported by: pprindeville
Patches:
issue17654.diff uploaded by qwell (license 4)
Tested by: qwell, pprindeville
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276830 f38db490-d61c-443f-a65b-d21fe96a405b
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Related to changes from 276571, I was accidentally testing with a port set in
my configuration causing me to miss this. Also moved the TCP handling as well
to occur before build_peer is called.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276788 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #17644)
Reported by: pprindeville
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276769 f38db490-d61c-443f-a65b-d21fe96a405b
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the real fix.
Review: https://reviewboard.asterisk.org/r/790/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276731 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r276652 | jpeeler | 2010-07-15 08:48:58 -0500 (Thu, 15 Jul 2010) | 2 lines
In a perfect world, the frame source would never be NULL. In the meantime, don't crash when it is.
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276653 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276616 f38db490-d61c-443f-a65b-d21fe96a405b
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MWI updates were not being sent if no messages were found in the event cache.
This was corrected since a phone may need to clear its MWI status configured
previously from another mailbox.
Upon module or sip reload, MWI updates could not be sent due to the sipsock
socket not being set early enough in reload_config. The code handling the
descriptor assignment and such has simply been moved before the call to
build_peer.
Issuing a sip reload cleared the IP address of the peer, but skipped checking
the database for registration information. The database is now checked both
for sip reload and actually reloading the module.
If a transmission occurs before the do_monitor thread has started, do not
attempt to send a signal to it.
(closes issue #17398)
Reported by: ip-rob
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276571 f38db490-d61c-443f-a65b-d21fe96a405b
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ast_sockaddr_stringiy_fmt (which is call by all ast_sockaddr_stringify* functions)
uses thread-local storage for storing the string that it creates. In cases where
ast_sockaddr_stringify_fmt was being called twice within the same statement, the
result of one call would be overwritten by the result of the other call. This
usually was happening in printf-like statements and was resulting in the same
stringified addressed being printed twice instead of two separate addresses.
I have fixed this by using ast_strdupa on the result of stringify functions if
they are used twice within the same statement. As far as I could tell, there were
no instances where a pointer to the result of such a call were saved anywhere, so
this is the only situation I could see where this error could occur.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276570 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276531 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276493 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #17475)
Reported by: tilghman
Review: https://reviewboard.asterisk.org/r/695/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276490 f38db490-d61c-443f-a65b-d21fe96a405b
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after confirming it exists.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276441 f38db490-d61c-443f-a65b-d21fe96a405b
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Unlike most responses, the 200 Ok to a pending INVITE Request is
acknowledged by an ACK Request. If the ACK Request for this Response is not received
the previous behavior was to immediately destroy the dialog and hangup
the channel. Now in an effort to be more RFC compliant, instead of immediately
destroying the dialog during this special case, termination is done with a BYE Request
as the dialog is technically confirmed when the 200 Ok is sent even if the ACK is
never received. The behavior of immediately hanging up the channel remains.
This only affects how dialog termination proceeds for this one special case.
RFC 3261 section 13.3.1.4
"If the server retransmits the 2xx response for 64*T1 seconds without receiving
an ACK, the dialog is confirmed, but the session SHOULD be terminated. This is
accomplished with a BYE, as described in Section 15."
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276439 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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I've been working in this function a bunch lately, and
these huge debug strings are getting annoying.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276392 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276391 f38db490-d61c-443f-a65b-d21fe96a405b
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