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r115548 | mattf | 2008-05-08 10:04:45 -0500 (Thu, 08 May 2008) | 1 line
Remove unused code as well as demote an error message to a debug message
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r115519 | russell | 2008-05-07 13:24:51 -0500 (Wed, 07 May 2008) | 2 lines
Let chan_h323 build in dev mode
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r115513 | russell | 2008-05-07 12:28:19 -0500 (Wed, 07 May 2008) | 19 lines
Merged revisions 115512 via svnmerge from
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r115512 | russell | 2008-05-07 11:24:09 -0500 (Wed, 07 May 2008) | 11 lines
Merged revisions 115511 via svnmerge from
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r115511 | russell | 2008-05-07 11:22:49 -0500 (Wed, 07 May 2008) | 3 lines
Remove remnants of dlinkedlists. I didn't actually use them in the final version
of my IAX2 improvements.
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r115315 | russell | 2008-05-05 15:28:17 -0500 (Mon, 05 May 2008) | 2 lines
Remove my rant, since I have now replaced the rant with code.
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r115305 | russell | 2008-05-05 14:50:24 -0500 (Mon, 05 May 2008) | 13 lines
Merged revisions 115304 via svnmerge from
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r115304 | russell | 2008-05-05 14:49:25 -0500 (Mon, 05 May 2008) | 5 lines
Avoid putting opaque="" in Digest authentication. This patch came from switchvox.
It fixes authentication with Primus in Canada, and has been in use for a very long
time without causing problems with any other providers.
(closes issue AST-36)
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r114922 | qwell | 2008-05-01 11:49:24 -0500 (Thu, 01 May 2008) | 10 lines
Allow dringXrange to properly default to 10, as was done in 1.4.
dringXrange is a new feature that was added, and it attempted to default, but only when the option was specified.
(closes issue #12536)
Reported by: bjm
Patches:
12536-dringXrange.diff uploaded by qwell (license 4)
Tested by: bjm
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r114899 | oej | 2008-04-30 18:55:49 +0200 (Ons, 30 Apr 2008) | 15 lines
Merged revisions 114890 via svnmerge from
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r114890 | oej | 2008-04-30 18:23:17 +0200 (Ons, 30 Apr 2008) | 7 lines
Don't crash on bad SIP replys.
Fix created in Huntsville together with Mark M (putnopvut)
(closes issue #12363)
Reported by: jvandal
Tested by: putnopvut, oej
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r114892 | russell | 2008-04-30 11:34:24 -0500 (Wed, 30 Apr 2008) | 36 lines
Merged revisions 114891 via svnmerge from
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r114891 | russell | 2008-04-30 11:30:01 -0500 (Wed, 30 Apr 2008) | 28 lines
Merge changes from team/russell/iax2_find_callno and iax2_find_callno_1.4
These changes address a critical performance issue introduced in the latest
release. The fix for the latest security issue included a change that made
Asterisk randomly choose call numbers to make them more difficult to guess by
attackers. However, due to some inefficient (this is by far, an understatement)
code, when Asterisk chose high call numbers, chan_iax2 became unusable after
just a small number of calls. On a small embedded platform, it would not be
able to handle a single call. On my Intel Core 2 Duo @ 2.33 GHz, I couldn't
run more than about 16 IAX2 channels. Ouch.
These changes address some performance issues of the find_callno() function
that have bothered me for a very long time. On every incoming media frame,
it iterated through every possible call number trying to find a matching
active call. This involved a mutex lock and unlock for each call number
checked. So, if the random call number chosen was 20000, then every media
frame would cause 20000 locks and unlocks. Previously, this problem was
not as obvious since Asterisk always chose the lowest call number it could.
A second container for IAX2 pvt structs has been added. It is an astobj2
hash table. When we know the remote side's call number, the pvt goes into
the hash table with a hash value of the remote side's call number. Then,
lookups for incoming media frames are a very fast hash lookup instead of an
absolutely insane array traversal.
In a quick test, I was able to get more than 3600% more IAX2 channels
on my machine with these changes.
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r114888 | jpeeler | 2008-04-30 11:14:43 -0500 (Wed, 30 Apr 2008) | 3 lines
Fixes a bug where if a stream monitor thread was not created (caused from failure of opening or starting the stream) pthread_cancel was called with an invalid thread ID.
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r114884 | kpfleming | 2008-04-30 09:49:51 -0500 (Wed, 30 Apr 2008) | 10 lines
Merged revisions 114880 via svnmerge from
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r114880 | kpfleming | 2008-04-30 09:46:57 -0500 (Wed, 30 Apr 2008) | 2 lines
use the ARRAY_LEN macro for indexing through the iaxs/iaxsl arrays so that the size of the arrays can be adjusted in one place, and change the size of the arrays from 32768 calls to 2048 calls when LOW_MEMORY is defined
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r114866 | jpeeler | 2008-04-29 17:54:14 -0500 (Tue, 29 Apr 2008) | 2 lines
Fixes a problem where all the templates were marked as dead no matter what. The templates should only be marked as dead if a configuration file has been successfully loaded and has changes. Bug found while making API documentation for 1.6.0.
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r114776 | mattf | 2008-04-28 12:00:38 -0500 (Mon, 28 Apr 2008) | 1 line
Fix deadlock issue in chan_zap with libss7 due to channel variables being set with the channel pvt lock being held. #12512
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r114709 | tilghman | 2008-04-27 23:53:20 -0500 (Sun, 27 Apr 2008) | 13 lines
Merged revisions 114708 via svnmerge from
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r114708 | tilghman | 2008-04-27 23:47:39 -0500 (Sun, 27 Apr 2008) | 5 lines
When modules are embedded, they take on a different name, without the ".so"
extension. Specifically check for this name, when we're checking if a module
is loaded.
(Closes issue #12534)
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Merged to 1.6 because it fixes a crash.
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r114700 | mvanbaak | 2008-04-27 17:17:18 +0200 (Sun, 27 Apr 2008) | 8 lines
Make MWI in chan_skinny event based modeled after chan_zap and chan_mgcp.
(closes issue #12214)
Reported by: DEA
Patches:
chan_skinny-vm-events-v3.txt uploaded by DEA (license 3)
Tested by: DEA and me
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r114674 | russell | 2008-04-25 17:00:35 -0500 (Fri, 25 Apr 2008) | 11 lines
Merged revisions 114673 via svnmerge from
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r114673 | russell | 2008-04-25 16:54:40 -0500 (Fri, 25 Apr 2008) | 3 lines
Use consistent logic for checking to see if a call number has been chosen yet.
Also, remove some redundant logic I recently added in a fix.
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r114635 | file | 2008-04-24 19:11:46 -0300 (Thu, 24 Apr 2008) | 4 lines
Hey look, it builds.
(closes issue #12519)
Reported by: falves11
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r114633 | mmichelson | 2008-04-24 16:35:39 -0500 (Thu, 24 Apr 2008) | 19 lines
Merged revisions 114632 via svnmerge from
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r114632 | mmichelson | 2008-04-24 16:35:08 -0500 (Thu, 24 Apr 2008) | 11 lines
Re-invite RTP during a masquerade so that, for instance, an AMI
redirect of two channels which are natively bridged will preserve audio
on both channels. This prevents a problem with Asterisk not re-inviting
due to one of the channels having being a zombie.
(closes issue #12513)
Reported by: mneuhauser
Patches:
asterisk-1.4-114602_restore-RTP-on-fixup.patch uploaded by mneuhauser (license 425)
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r114625 | mmichelson | 2008-04-24 15:06:06 -0500 (Thu, 24 Apr 2008) | 18 lines
Merged revisions 114624 via svnmerge from
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r114624 | mmichelson | 2008-04-24 15:04:24 -0500 (Thu, 24 Apr 2008) | 10 lines
Resolve a deadlock in chan_local by releasing the channel lock
temporarily.
(closes issue #11712)
Reported by: callguy
Patches:
11712.patch uploaded by putnopvut (license 60)
Tested by: acunningham
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r114622 | tilghman | 2008-04-24 14:54:57 -0500 (Thu, 24 Apr 2008) | 12 lines
Merged revisions 114621 via svnmerge from
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r114621 | tilghman | 2008-04-24 14:53:36 -0500 (Thu, 24 Apr 2008) | 4 lines
Ensure that when we set the accountcode, it actually shows up in the CDR.
(Fix for AMI Originate)
(Closes issue #12007)
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r114612 | qwell | 2008-04-24 11:47:01 -0500 (Thu, 24 Apr 2008) | 17 lines
Merged revisions 51989 via svnmerge from
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(closes issue #12496)
Reported by: daniele
Patches:
misdn-moh-1.6.0-beta7.1.patch uploaded by daniele (license 471)
Tested by: daniele
Technically, I didn't use the patch above except to find out what revision to merge - but it's the same thing as this revision.
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r51989 | crichter | 2007-01-24 06:57:22 -0600 (Wed, 24 Jan 2007) | 1 line
added fix from #8899
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r114609 | russell | 2008-04-24 10:56:55 -0500 (Thu, 24 Apr 2008) | 12 lines
Merged revisions 114608 via svnmerge from
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r114608 | russell | 2008-04-24 10:55:21 -0500 (Thu, 24 Apr 2008) | 4 lines
Fix a silly mistake in a change I made yesterday that caused chan_iax2 to blow
up very quickly.
(issue #12515)
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r114606 | oej | 2008-04-24 16:59:05 +0200 (Tor, 24 Apr 2008) | 11 lines
Merged revisions 114603 via svnmerge from
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r114603 | oej | 2008-04-24 16:55:18 +0200 (Tor, 24 Apr 2008) | 3 lines
Only have one max-forwards header in outbound REFERs.
Discovered in the Asterisk SIP Masterclass in Orlando. Thanks Joe!
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r114604 | russell | 2008-04-24 09:55:21 -0500 (Thu, 24 Apr 2008) | 3 lines
Change a verbose message to debug.
(closes issue #12514)
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r114588 | russell | 2008-04-23 12:18:29 -0500 (Wed, 23 Apr 2008) | 10 lines
Merged revisions 114587 via svnmerge from
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r114587 | russell | 2008-04-23 12:16:32 -0500 (Wed, 23 Apr 2008) | 2 lines
Fix find_callno_locked() to actually return the callno locked in some more cases.
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r114585 | oej | 2008-04-23 18:53:34 +0200 (Ons, 23 Apr 2008) | 10 lines
Merged revisions 114584 via svnmerge from
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r114584 | oej | 2008-04-23 18:51:41 +0200 (Ons, 23 Apr 2008) | 2 lines
Add 502 support for both directions, not only one... (see r114571)
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r114572 | tilghman | 2008-04-22 18:58:19 -0500 (Tue, 22 Apr 2008) | 10 lines
Merged revisions 114571 via svnmerge from
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r114571 | tilghman | 2008-04-22 18:51:44 -0500 (Tue, 22 Apr 2008) | 2 lines
Treat a 502 just like a 503, when it comes to processing a response code
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r114559 | russell | 2008-04-22 17:17:31 -0500 (Tue, 22 Apr 2008) | 13 lines
Merged revisions 114558 via svnmerge from
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r114558 | russell | 2008-04-22 17:15:36 -0500 (Tue, 22 Apr 2008) | 5 lines
When we receive a full frame that is supposed to contain our call number,
ensure that it has the correct one.
(closes issue #10078)
(AST-2008-006)
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r114538 | russell | 2008-04-22 13:04:39 -0500 (Tue, 22 Apr 2008) | 17 lines
Merged revisions 114537 via svnmerge from
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r114537 | russell | 2008-04-22 13:03:33 -0500 (Tue, 22 Apr 2008) | 9 lines
If the dial string passed to the call channel callback does not indicate an
extension, then consider the extension on the channel before falling back
to the default.
(closes issue #12479)
Reported by: darren1713
Patches:
exten_dial_fix_chan_iax2.c.patch uploaded by darren1713 (license 116)
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r114389 | mattf | 2008-04-21 13:44:35 -0500 (Mon, 21 Apr 2008) | 1 line
Add support for generic name transmission (#12484) on SS7 in chan_zap
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r114323 | file | 2008-04-21 11:40:33 -0300 (Mon, 21 Apr 2008) | 12 lines
Merged revisions 114322 via svnmerge from
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r114322 | file | 2008-04-21 11:39:32 -0300 (Mon, 21 Apr 2008) | 4 lines
Only drop audio if we receive it without a progress indication. We allow other frames through such as DTMF because they may be needed to complete the call.
(closes issue #12440)
Reported by: aragon
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r114271 | file | 2008-04-18 16:35:33 -0300 (Fri, 18 Apr 2008) | 4 lines
Make sure ADSI is marked as unavailable on Unistim channels so voicemail does not try to do some ADSI jazz.
(closes issue #12460)
Reported by: PerryB
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r114259 | mmichelson | 2008-04-18 13:03:06 -0500 (Fri, 18 Apr 2008) | 14 lines
Merged revisions 114257 via svnmerge from
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r114257 | mmichelson | 2008-04-18 12:44:29 -0500 (Fri, 18 Apr 2008) | 6 lines
Clearing up error messages so they make a bit more sense. Also removing a redundant error
message.
Issue AST-15
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r114246 | seanbright | 2008-04-18 09:38:07 -0400 (Fri, 18 Apr 2008) | 9 lines
Merged revisions 114245 via svnmerge from
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r114245 | seanbright | 2008-04-18 09:33:32 -0400 (Fri, 18 Apr 2008) | 1 line
Only complete the SIP channel name once for 'sip show channel <channel>'
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r114151 | oej | 2008-04-15 15:39:29 -0500 (Tue, 15 Apr 2008) | 10 lines
Merged revisions 114148 via svnmerge from
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r114148 | oej | 2008-04-15 22:26:05 +0200 (Tis, 15 Apr 2008) | 2 lines
Handle subscribe queues in all situations... Thanks to festr_ on irc for telling me about this bug.
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r114150 | oej | 2008-04-15 15:31:08 -0500 (Tue, 15 Apr 2008) | 2 lines
Adding chanvar to SIPPEER from 1.4 branch
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r114185 | kpfleming | 2008-04-16 15:47:30 -0500 (Wed, 16 Apr 2008) | 14 lines
Merged revisions 114184 via svnmerge from
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r114184 | kpfleming | 2008-04-16 15:46:38 -0500 (Wed, 16 Apr 2008) | 6 lines
use the ZT_SET_DIALPARAMS ioctl properly by initializing the structure to all zeroes in case it contains fields that we don't write values into (which it does as of Zaptel 1.4.10)
(closes issue #12456)
Reported by: fnordian
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r114141 | qwell | 2008-04-15 12:21:58 -0500 (Tue, 15 Apr 2008) | 8 lines
Shorten the mac address pattern, since some phones use different identifiers (such as the i2050 softphone).
(closes issue #12398)
Reported by: c_hans
Patches:
chan_unistim_svn.diff uploaded by c (license 460)
Tested by: c_hans
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r114121 | qwell | 2008-04-14 13:34:17 -0500 (Mon, 14 Apr 2008) | 15 lines
Merged revisions 114120 via svnmerge from
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r114120 | qwell | 2008-04-14 13:31:57 -0500 (Mon, 14 Apr 2008) | 7 lines
The call_token on the pvt can occasionally be NULL, causing a crash.
If it is NULL, we can skip this channel, since it can't the one we're looking for.
(closes issue #9299)
Reported by: vazir
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r114104 | file | 2008-04-14 11:53:33 -0300 (Mon, 14 Apr 2008) | 12 lines
Merged revisions 114103 via svnmerge from
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r114103 | file | 2008-04-14 11:52:46 -0300 (Mon, 14 Apr 2008) | 4 lines
It is possible for the remote side to say they want T38 but not give any capabilities.
(closes issue #12414)
Reported by: MVF
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Also make sure when completing a COT we call ss7_start_call with the proper locks held. Lastly, make sure if we fail to get a channel from zt_new that we don't assume it's there.
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r114084 | twilson | 2008-04-11 17:48:52 -0500 (Fri, 11 Apr 2008) | 15 lines
Merged revisions 114083 via svnmerge from
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r114083 | twilson | 2008-04-11 17:32:51 -0500 (Fri, 11 Apr 2008) | 7 lines
Several places in the code called find_callno() (which releases the lock on the pvt structure) and then immediately locked the call and did things with it. Unfortunately, the call can disappear between the find_callno and the lock, causing Bad Stuff(tm) to happen.
Added find_callno_locked() function to return the callno withtout unlocking for instances that it is needed.
(issue #12400)
Reported by: ztel
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r114046 | mmichelson | 2008-04-10 14:58:36 -0500 (Thu, 10 Apr 2008) | 14 lines
Merged revisions 114045 via svnmerge from
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r114045 | mmichelson | 2008-04-10 14:55:33 -0500 (Thu, 10 Apr 2008) | 6 lines
Be sure that we're not about to set bridgepvt NULL prior to dereferencing it.
(closes issue #11775)
Reported by: fujin
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r114024 | file | 2008-04-10 10:45:45 -0300 (Thu, 10 Apr 2008) | 4 lines
Fix spelling of existent in a few places.
(closes issue #12409)
Reported by: candlerb
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r114022 | file | 2008-04-10 10:28:30 -0300 (Thu, 10 Apr 2008) | 14 lines
Merged revisions 114021 via svnmerge from
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r114021 | file | 2008-04-10 10:27:11 -0300 (Thu, 10 Apr 2008) | 6 lines
Don't add custom URI options if they don't exist OR they are empty.
(closes issue #12407)
Reported by: homesick
Patches:
uri_options-1.4.diff uploaded by homesick (license 91)
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r113928 | mmichelson | 2008-04-09 15:56:14 -0500 (Wed, 09 Apr 2008) | 16 lines
Merged revisions 113927 via svnmerge from
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r113927 | mmichelson | 2008-04-09 15:54:31 -0500 (Wed, 09 Apr 2008) | 8 lines
We need to set the persistant_route [sic] parameter for the sip_pvt
during the initial INVITE, no matter if we're building the route set from
an INVITE request or response.
(closes issue #12391)
Reported by: benjaminbohlmann
Tested by: benjaminbohlmann
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r113785 | file | 2008-04-09 13:52:04 -0300 (Wed, 09 Apr 2008) | 12 lines
Merged revisions 113784 via svnmerge from
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r113784 | file | 2008-04-09 13:50:45 -0300 (Wed, 09 Apr 2008) | 4 lines
If we receive an AUTHREQ from the remote server and we are unable to reply (for example they have a secret configured, but we do not) then queue a hangup frame on the Asterisk channel. This will cause the channel to hangup and a HANGUP to be sent via IAX2 to the remote side which is the proper thing to do in this scenario.
(closes issue #12385)
Reported by: viraptor
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r113682 | mmichelson | 2008-04-09 09:41:58 -0500 (Wed, 09 Apr 2008) | 17 lines
Merged revisions 113681 via svnmerge from
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r113681 | mmichelson | 2008-04-09 09:40:05 -0500 (Wed, 09 Apr 2008) | 9 lines
If Asterisk receives a 488 on an INVITE (not a reinvite), then
we should not send a BYE.
(closes issue #12392)
Reported by: fnordian
Patches:
chan_sip.patch uploaded by fnordian (license 110) with small modification from me
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