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Summary:ASTERISK-06148: Bridged Sipura ATA to SIP provider call generates "503 Server Error" in a loop (Asterisk 1.2)
Reporter:Peter Whisker (whiskerp)Labels:
Date Opened:2006-01-20 13:49:21.000-0600Date Closed:2008-01-15 16:28:31.000-0600
Priority:MajorRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/Interoperability
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) sip.txt
Description:I have an ATA connected as an Asterisk extension. If I call an external number (on a SIP service provider), all is well until Asterisk bridges the two voice circuits on answer. Then a loop of "503 Server Error" - "ACK" - "503 Server Error" occurs with Asterisk and the Sipura until the ATA crashes. The call and audio both work until it crashes. Asterisk seems to think that something is wrong with the "ACK" but I don't think that there is.

Here is a syslog trace generated by the Sipura:

This all worked perfectly for the last 12 months until I upgraded to the latest 1.2 branch last week (I had been running on a May 2005 CVS Head before then).

I commented out the test and generation of the "503 Server error" in chan_sip.c and all is now working fine, but I don't know the underlying cause.

Peter
network:/usr/src/asterisk/asterisk-1.2# svn info asterisk-1.2
Path: asterisk-1.2
URL: http://svn.digium.com/svn/asterisk/branches/1.2
Repository UUID: 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Revision: 8293
Node Kind: directory
Schedule: normal
Last Changed Author: oej
Last Changed Rev: 8281
Last Changed Date: 2006-01-19 19:40:28 +0000 (Thu, 19 Jan 2006)
Properties Last Updated: 2006-01-19 20:30:41 +0000 (Thu, 19 Jan 2006)


****** ADDITIONAL INFORMATION ******

Jan 20 20:26:06 192.168.1.100 SIP/2.0 200 OK^M Via: SIP/2.0/UDP 192.168.1.100:5061;branch=z9hG4bK-ef7b79ea;rport;received=192.168.1.100^M From: 330 <sip:330@192.168.1.250:5052>;tag=ea042a58fb3c9397o1^M To: <sip:02075449494@192.168.1.250:5052>;tag=as199d2858^M Call-ID: d0b6ee2c-b30a81d3@192.168.1.100^M CSeq: 114 INFO^M User-Agent: Asterisk PBX^M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY^M Contact: <sip:02075449494@192.168.1.250:5052>^M Content-Length: 0^M X-Asterisk-HangupCause: Normal Clearing^M ^M
Jan 20 20:26:06 192.168.1.100
Jan 20 20:26:06 192.168.1.100
Jan 20 20:26:06 192.168.1.100 [1:5061]->192.168.1.250:5052
Jan 20 20:26:06 192.168.1.100 [1:5061]->192.168.1.250:5052
Jan 20 20:26:06 192.168.1.100 INFO sip:02075449494@192.168.1.250:5052 SIP/2.0^M Via: SIP/2.0/UDP 192.168.1.100:5061;branch=z9hG4bK-67f24275;rport^M From: 330 <sip:330@192.168.1.250:5052>;tag=ea042a58fb3c9397o1^M To: <sip:02075449494@192.168.1.250:5052>;tag=as199d2858^M Call-ID: d0b6ee2c-b30a81d3@192.168.1.100^M CSeq: 115 INFO^M Max-Forwards: 70^M Proxy-Authorization: Digest username="330",realm="voip.whisker.org.uk",nonce="327d5834",uri="sip:02075449494@192.168.1.250:5052",algorithm=MD5,response="0c01e064373416e513dc70a3a334b5e1"^M User-Agent: Sipura/SPA2000-3.1.5^M Content-Length: 24^M Content-Type: application/dtmf-relay^M ^M Signal=0^M Duration=100^M
Jan 20 20:26:06 192.168.1.100
Jan 20 20:26:06 192.168.1.100
Jan 20 20:26:06 192.168.1.100 [1:5061]<<192.168.1.250:5052
Jan 20 20:26:06 192.168.1.100 [1:5061]<<192.168.1.250:5052

Jan 20 20:26:06 192.168.1.100 SIP/2.0 200 OK^M Via: SIP/2.0/UDP 192.168.1.100:5061;branch=z9hG4bK-67f24275;rport;received=192.168.1.100^M From: 330 <sip:330@192.168.1.250:5052>;tag=ea042a58fb3c9397o1^M To: <sip:02075449494@192.168.1.250:5052>;tag=as199d2858^M Call-ID: d0b6ee2c-b30a81d3@192.168.1.100^M CSeq: 115 INFO^M User-Agent: Asterisk PBX^M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY^M Contact: <sip:02075449494@192.168.1.250:5052>^M Content-Length: 0^M X-Asterisk-HangupCause: Normal Clearing^M ^M
Comments:By: Olle Johansson (oej) 2006-01-20 14:45:28.000-0600

Please upload a full output of SIP debug including output from Asterisk, debug level 4 and verbosity set to 4. Add that as an attachment to this bug report. Thank you.
By: Peter Whisker (whiskerp) 2006-01-20 21:08:59.000-0600

Here it is (in uploaded file sip.txt). The problem follows the line:
-- Attempting native bridge of SIP/330-847a and SIP/VS-d787

Why does Asterisk generate all these "SIP/2.0 503 Server error" messages?


By: Olle Johansson (oej) 2006-01-24 06:11:29.000-0600

Ok. We should never send any response to an ACK. Will fix in 1.2 and svn trunk.
By: Olle Johansson (oej) 2006-01-24 06:21:46.000-0600

Fixed in 1.2 rev 8537, rev 8538 in trunk. Thank you for reporting this!
By: Digium Subversion (svnbot) 2008-01-15 16:23:03.000-0600

Repository: asterisk
Revision: 8537

U   branches/1.2/channels/chan_sip.c

------------------------------------------------------------------------
r8537 | oej | 2008-01-15 16:23:03 -0600 (Tue, 15 Jan 2008) | 2 lines

Issue ASTERISK-6148 - never send response to ACK. (Reported by whiskerp)

------------------------------------------------------------------------

http://svn.digium.com/view/asterisk?view=rev&revision=8537
By: Digium Subversion (svnbot) 2008-01-15 16:23:04.000-0600

Repository: asterisk
Revision: 8538

U   trunk/channels/chan_sip.c

------------------------------------------------------------------------
r8538 | oej | 2008-01-15 16:23:04 -0600 (Tue, 15 Jan 2008) | 2 lines

Importing rev ASTERISK-8302 from 1.2, never send response to ACK (issue ASTERISK-6148)

------------------------------------------------------------------------

http://svn.digium.com/view/asterisk?view=rev&revision=8538
By: Digium Subversion (svnbot) 2008-01-15 16:25:13.000-0600

Repository: asterisk
Revision: 8679

_U  team/oej/astum/
D   team/oej/astum/ChangeLog
U   team/oej/astum/apps/app_dial.c
U   team/oej/astum/asterisk.c
U   team/oej/astum/cdr/cdr_pgsql.c
U   team/oej/astum/channel.c
U   team/oej/astum/channels/chan_agent.c
U   team/oej/astum/channels/chan_features.c
U   team/oej/astum/channels/chan_iax2.c
U   team/oej/astum/channels/chan_sip.c
U   team/oej/astum/configs/sip.conf.sample
U   team/oej/astum/contrib/scripts/safe_asterisk
U   team/oej/astum/include/asterisk/channel.h
U   team/oej/astum/rtp.c
U   team/oej/astum/utils/astman.c

------------------------------------------------------------------------
r8679 | oej | 2008-01-15 16:25:13 -0600 (Tue, 15 Jan 2008) | 230 lines

Merged revisions 8517,8523-8524,8531,8538-8539,8548,8554,8560-8561,8563,8571-8572,8574,8582,8587,8589-8597,8599,8609-8610,8618,8620,8633,8642-8643,8654,8664-8665,8667,8676,8678 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
r8517 | oej | 2006-01-24 11:36:45 +0100 (Tue, 24 Jan 2006) | 2 lines

Whitespace change, extra <tab> added from my tab storage.

................
r8523 | oej | 2006-01-24 12:42:09 +0100 (Tue, 24 Jan 2006) | 2 lines

Declaring conn and result static to avoid collission with realtime driver (issue 6336, pressureman)

................
r8524 | oej | 2006-01-24 12:46:29 +0100 (Tue, 24 Jan 2006) | 3 lines

- Adding whitespace that I found unused outside
- Adding "if (option_debug)" before outputting to DEBUG channel

................
r8531 | oej | 2006-01-24 13:48:44 +0100 (Tue, 24 Jan 2006) | 2 lines

- Report SIP reload in manager (issue 5742 with small changes)

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r8538 | oej | 2006-01-24 14:21:13 +0100 (Tue, 24 Jan 2006) | 2 lines

Importing rev ASTERISK-8302 from 1.2, never send response to ACK (issue ASTERISK-6148)

................
r8539 | oej | 2006-01-24 14:53:45 +0100 (Tue, 24 Jan 2006) | 2 lines

Issue ASTERISK-6163, FreeBSD compatibility with compilation of func_odbc.c (reported by nulbyte)

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r8548 | oej | 2006-01-24 18:47:41 +0100 (Tue, 24 Jan 2006) | 2 lines

Reverting change in revision 8539 - fixed wrong problem. Sorry.

................
r8554 | oej | 2006-01-24 19:15:20 +0100 (Tue, 24 Jan 2006) | 2 lines

Make it clear that caller ID in sip.conf is used only on incoming calls (inspired by bug ASTERISK-6026)

................
r8560 | oej | 2006-01-24 20:08:44 +0100 (Tue, 24 Jan 2006) | 2 lines

Issue ASTERISK-5935: Match realtime non-dynamic peers by IP. (siacali).

................
r8561 | oej | 2006-01-24 20:19:20 +0100 (Tue, 24 Jan 2006) | 2 lines

Issue 6114: Don't hangup on bye/also if there's no channel. (gst)

................
r8563 | oej | 2006-01-24 20:29:32 +0100 (Tue, 24 Jan 2006) | 2 lines

Blocking fix from 1.2 from being applied again.

................
r8571 | russell | 2006-01-24 21:20:05 +0100 (Tue, 24 Jan 2006) | 2 lines

convert ast_channel list to use linked list macros (issue ASTERISK-6178)

................
r8572 | russell | 2006-01-24 21:27:09 +0100 (Tue, 24 Jan 2006) | 2 lines

store the list of 'atexit' functions using linked list macros (issue ASTERISK-6169)

................
r8574 | oej | 2006-01-24 21:41:08 +0100 (Tue, 24 Jan 2006) | 2 lines

Don't reset scheduled ID until we actually end the scheduled event.

................
r8582 | mattf | 2006-01-24 22:45:42 +0100 (Tue, 24 Jan 2006) | 2 lines

Updates from royk to safe_asterisk (ASTERISK-5069) Thanks!

................
r8587 | mattf | 2006-01-24 23:06:37 +0100 (Tue, 24 Jan 2006) | 2 lines

Make sure safe_asterisk retains previous script defaults

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r8589 | kpfleming | 2006-01-24 23:33:58 +0100 (Tue, 24 Jan 2006) | 1 line


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r8590 | kpfleming | 2006-01-24 23:34:06 +0100 (Tue, 24 Jan 2006) | 1 line


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r8591 | kpfleming | 2006-01-24 23:38:17 +0100 (Tue, 24 Jan 2006) | 10 lines

Merged revisions 8588 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2

........
r8588 | kpfleming | 2006-01-24 16:32:09 -0600 (Tue, 24 Jan 2006) | 2 lines

ensure that channel cannot become zombie after we check but before we try to start indications

........

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r8592 | kpfleming | 2006-01-24 23:40:20 +0100 (Tue, 24 Jan 2006) | 1 line


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r8593 | kpfleming | 2006-01-24 23:40:57 +0100 (Tue, 24 Jan 2006) | 1 line


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r8594 | kpfleming | 2006-01-24 23:41:45 +0100 (Tue, 24 Jan 2006) | 1 line


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r8595 | kpfleming | 2006-01-24 23:42:43 +0100 (Tue, 24 Jan 2006) | 10 lines

Merged revisions 8173 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2

........
r8173 | russell | 2006-01-17 20:49:21 -0600 (Tue, 17 Jan 2006) | 2 lines

remove ChangeLog from the 1.2 branch.  It will only be present in the tags.

........

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r8596 | kpfleming | 2006-01-24 23:43:30 +0100 (Tue, 24 Jan 2006) | 1 line


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r8597 | kpfleming | 2006-01-24 23:43:57 +0100 (Tue, 24 Jan 2006) | 2 lines

clean up remaining already-merged revisions

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r8599 | kpfleming | 2006-01-24 23:45:41 +0100 (Tue, 24 Jan 2006) | 2 lines

remove extraneous characters from property

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r8609 | kpfleming | 2006-01-25 02:52:58 +0100 (Wed, 25 Jan 2006) | 10 lines

Merged revisions 8608 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r8608 | kpfleming | 2006-01-24 19:50:52 -0600 (Tue, 24 Jan 2006) | 2 lines

ensure hangup cause code is handled properly when channel does not return a frame (issue ASTERISK-6186)

........

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r8610 | kpfleming | 2006-01-25 02:53:15 +0100 (Wed, 25 Jan 2006) | 1 line


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r8618 | russell | 2006-01-25 06:37:29 +0100 (Wed, 25 Jan 2006) | 3 lines

don't leak almost 200 bytes for each new channel and store the active
channel list using the linked list macros (issue ASTERISK-6170)

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r8620 | russell | 2006-01-25 06:39:25 +0100 (Wed, 25 Jan 2006) | 1 line


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r8633 | oej | 2006-01-25 10:50:28 +0100 (Wed, 25 Jan 2006) | 2 lines

Issue ASTERISK-6189 - patch by markster, imported from 1.2

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r8642 | oej | 2006-01-25 13:01:07 +0100 (Wed, 25 Jan 2006) | 3 lines

From now on, apply maxexpiry and minexpiry to all subscriptions. Thanks to fourcheeze in the IRC channel
for pointing this out.

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r8643 | oej | 2006-01-25 13:11:30 +0100 (Wed, 25 Jan 2006) | 3 lines

- Remove unused option to transmit_state_notify
- Allow for expiry=0 in subscription requests that only wants *one* update and that's it.

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r8654 | kpfleming | 2006-01-25 15:52:43 +0100 (Wed, 25 Jan 2006) | 3 lines

don't queue a congestion frame on a channel that will be immediately hung up anyway
clean up/organize code block

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r8664 | russell | 2006-01-25 19:12:55 +0100 (Wed, 25 Jan 2006) | 2 lines

store agent_pvt list using linked list macros (issue ASTERISK-6182)

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r8665 | russell | 2006-01-25 19:24:32 +0100 (Wed, 25 Jan 2006) | 3 lines

store feature_pvt list using linked list macros
(issue ASTERISK-6190, with additional changes to prevent a memory leak in unload_module)

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r8667 | russell | 2006-01-25 19:41:12 +0100 (Wed, 25 Jan 2006) | 1 line


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r8676 | russell | 2006-01-25 20:06:37 +0100 (Wed, 25 Jan 2006) | 2 lines

use arg parsing macros in the AGENT dialplan function (issue ASTERISK-6078, with small mods)

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r8678 | russell | 2006-01-25 20:16:14 +0100 (Wed, 25 Jan 2006) | 11 lines

Merged revisions 8677 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r8677 | russell | 2006-01-25 14:14:43 -0500 (Wed, 25 Jan 2006) | 3 lines

don't call ast_update_realtime with uninitialized variables if we get a
registration with an expirey of 0 seconds (issue ASTERISK-6016)

........

................

------------------------------------------------------------------------

http://svn.digium.com/view/asterisk?view=rev&revision=8679
By: Digium Subversion (svnbot) 2008-01-15 16:28:25.000-0600

Repository: asterisk
Revision: 8891

_U  team/oej/managerstuff/
D   team/oej/managerstuff/ChangeLog
U   team/oej/managerstuff/apps/app_dial.c
U   team/oej/managerstuff/apps/app_festival.c
U   team/oej/managerstuff/apps/app_meetme.c
U   team/oej/managerstuff/apps/app_milliwatt.c
U   team/oej/managerstuff/apps/app_queue.c
U   team/oej/managerstuff/ast_expr2.c
U   team/oej/managerstuff/ast_expr2.fl
U   team/oej/managerstuff/ast_expr2.h
U   team/oej/managerstuff/ast_expr2.y
U   team/oej/managerstuff/ast_expr2f.c
U   team/oej/managerstuff/asterisk.c
U   team/oej/managerstuff/channel.c
U   team/oej/managerstuff/channels/chan_features.c
U   team/oej/managerstuff/channels/chan_iax2.c
U   team/oej/managerstuff/channels/chan_sip.c
U   team/oej/managerstuff/channels/chan_zap.c
U   team/oej/managerstuff/loader.c
U   team/oej/managerstuff/logger.c
U   team/oej/managerstuff/pbx.c
U   team/oej/managerstuff/res/res_features.c
U   team/oej/managerstuff/utils/astman.c

------------------------------------------------------------------------
r8891 | oej | 2008-01-15 16:28:24 -0600 (Tue, 15 Jan 2008) | 202 lines

Merged revisions 8112,8122,8124,8134,8140,8162,8173,8194,8232,8242,8276,8281,8320,8347,8394,8412,8414,8418,8429,8433,8437,8445,8537,8562,8573,8588,8600,8608,8619,8632,8666,8677,8710,8729,8758,8785,8808 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r8112 | kpfleming | 2006-01-17 00:51:37 +0100 (Tue, 17 Jan 2006) | 2 lines

do rlimit check _after_ reading config file, in case 'dumpcore' is specified there

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r8122 | kpfleming | 2006-01-17 14:11:55 +0100 (Tue, 17 Jan 2006) | 2 lines

update CLI copyright notice

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r8124 | mogorman | 2006-01-17 17:55:30 +0100 (Tue, 17 Jan 2006) | 3 lines

Fixed code ordering of logger_init and queue_log_init
bug 6263

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r8134 | mattf | 2006-01-17 19:29:57 +0100 (Tue, 17 Jan 2006) | 2 lines

Backport of fix for ASTERISK-5936

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r8140 | mogorman | 2006-01-17 21:10:29 +0100 (Tue, 17 Jan 2006) | 3 lines

Stop any generators running on a channel when
festival is called as described in 5996

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r8162 | mogorman | 2006-01-18 01:47:04 +0100 (Wed, 18 Jan 2006) | 4 lines

Changed order of autoload so that pbx_ comes before
channels, and in doing so cause bug 6002 to not
be an issue

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r8173 | russell | 2006-01-18 03:49:21 +0100 (Wed, 18 Jan 2006) | 2 lines

remove ChangeLog from the 1.2 branch.  It will only be present in the tags.

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r8194 | mogorman | 2006-01-18 22:02:06 +0100 (Wed, 18 Jan 2006) | 3 lines

Solves issue with the login proccess in meetme
patch from 6136

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r8232 | russell | 2006-01-19 05:17:45 +0100 (Thu, 19 Jan 2006) | 3 lines

fix a seg fault due to assuming that space gets allocatted on the stack in the
same order that we declare the variables (issue ASTERISK-6130)

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r8242 | russell | 2006-01-19 05:56:48 +0100 (Thu, 19 Jan 2006) | 3 lines

fix Message-Account header to use the ip address if the fromdomain
isn't set (issue ASTERISK-6118)

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r8276 | tilghman | 2006-01-19 20:14:37 +0100 (Thu, 19 Jan 2006) | 2 lines

Bug 6072 - Memory leaks in the expression parser

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r8281 | oej | 2006-01-19 20:40:28 +0100 (Thu, 19 Jan 2006) | 2 lines

Enable "musicclass" setting for sip peers as per the config sample.

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r8320 | mogorman | 2006-01-20 02:00:46 +0100 (Fri, 20 Jan 2006) | 3 lines

solved problem with delayreject and iax trunking
bug 4291

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r8347 | russell | 2006-01-20 19:34:42 +0100 (Fri, 20 Jan 2006) | 2 lines

fix invalid value of prev_q (issue ASTERISK-6142)

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r8394 | tilghman | 2006-01-21 19:29:39 +0100 (Sat, 21 Jan 2006) | 2 lines

Bug 5936 - AddQueueMember fails on realtime queue, if queue not yet loaded

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r8412 | russell | 2006-01-22 00:17:06 +0100 (Sun, 22 Jan 2006) | 2 lines

prevent the possibility of writing outside of the available workspace (issue ASTERISK-6111)

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r8414 | russell | 2006-01-22 00:43:14 +0100 (Sun, 22 Jan 2006) | 2 lines

temporarily revert substring fix pending the result of the discussion in issue ASTERISK-6111

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r8418 | russell | 2006-01-22 03:05:41 +0100 (Sun, 22 Jan 2006) | 3 lines

add a modified fix to prevent writing outside of the provided workspace when
calculating a substring (issue ASTERISK-6111)

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r8429 | tilghman | 2006-01-22 09:52:49 +0100 (Sun, 22 Jan 2006) | 2 lines

Bug 6281 - Cannot set more than a single header with SIPAddHeader

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r8433 | bweschke | 2006-01-22 16:13:41 +0100 (Sun, 22 Jan 2006) | 3 lines

Bug fix: Correct some scenarios where CALL_LIMIT could not be getting adjusted properly allowing chan_sip to send calls when it really shouldn't. Bug ASTERISK-5953


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r8437 | russell | 2006-01-22 18:47:13 +0100 (Sun, 22 Jan 2006) | 2 lines

fix MixMonitor crash (issue ASTERISK-6161, probably others)

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r8445 | russell | 2006-01-22 20:03:53 +0100 (Sun, 22 Jan 2006) | 2 lines

fix memory leak from not freeing the queue member list when freeing an old queue

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r8537 | oej | 2006-01-24 14:15:13 +0100 (Tue, 24 Jan 2006) | 2 lines

Issue ASTERISK-6148 - never send response to ACK. (Reported by whiskerp)

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r8562 | oej | 2006-01-24 20:21:15 +0100 (Tue, 24 Jan 2006) | 2 lines

Issue 6114: Don't hangup on BYE/ALSO with no channel.

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r8573 | mattf | 2006-01-24 21:37:30 +0100 (Tue, 24 Jan 2006) | 2 lines

Backport fix for ASTERISK-6071, hangup on polarity reversal

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r8588 | kpfleming | 2006-01-24 23:32:09 +0100 (Tue, 24 Jan 2006) | 2 lines

ensure that channel cannot become zombie after we check but before we try to start indications

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r8600 | russell | 2006-01-24 23:55:32 +0100 (Tue, 24 Jan 2006) | 2 lines

completely arbitrary whitespace change for testing something with svnmerge ...

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r8608 | kpfleming | 2006-01-25 02:50:52 +0100 (Wed, 25 Jan 2006) | 2 lines

ensure hangup cause code is handled properly when channel does not return a frame (issue ASTERISK-6186)

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r8619 | russell | 2006-01-25 06:38:36 +0100 (Wed, 25 Jan 2006) | 2 lines

don't leak almost 200 bytes for each new channel (issue ASTERISK-6170)

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r8632 | oej | 2006-01-25 10:46:43 +0100 (Wed, 25 Jan 2006) | 2 lines

Issue ASTERISK-6276 - the "timebomb" bug. Patch by Markster over GPRS

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r8666 | russell | 2006-01-25 19:39:44 +0100 (Wed, 25 Jan 2006) | 2 lines

fix memory leak (inspired by issue ASTERISK-6190)

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r8677 | russell | 2006-01-25 20:14:43 +0100 (Wed, 25 Jan 2006) | 3 lines

don't call ast_update_realtime with uninitialized variables if we get a
registration with an expirey of 0 seconds (issue ASTERISK-6016)

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r8710 | oej | 2006-01-26 15:39:36 +0100 (Thu, 26 Jan 2006) | 2 lines

Issue 5898: Registrations does not get deleted if there's an active SIP dialog

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r8729 | russell | 2006-01-26 20:42:35 +0100 (Thu, 26 Jan 2006) | 2 lines

fix problem with dtmf on e&m (issue ASTERISK-6203)

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r8758 | tilghman | 2006-01-27 01:52:12 +0100 (Fri, 27 Jan 2006) | 2 lines

Bug 6072 - Revisions to the source bison and flex files don't auto-regenerate these files

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r8785 | oej | 2006-01-27 09:02:16 +0100 (Fri, 27 Jan 2006) | 2 lines

Issue 6362 - Register without Contact: and Expires: fails (reporter: op)

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r8808 | oej | 2006-01-28 14:52:15 +0100 (Sat, 28 Jan 2006) | 3 lines

Issue 6182 - Don't remove scheduled event until it's really done.
(reported by malverian)

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------------------------------------------------------------------------

http://svn.digium.com/view/asterisk?view=rev&revision=8891
By: Digium Subversion (svnbot) 2008-01-15 16:28:31.000-0600

Repository: asterisk
Revision: 8894

_U  team/oej/moduletest/
U   team/oej/moduletest/apps/app_dial.c
U   team/oej/moduletest/apps/app_queue.c
U   team/oej/moduletest/ast_expr2.c
U   team/oej/moduletest/ast_expr2.h
U   team/oej/moduletest/ast_expr2f.c
U   team/oej/moduletest/asterisk.c
U   team/oej/moduletest/channel.c
U   team/oej/moduletest/channels/chan_features.c
U   team/oej/moduletest/channels/chan_iax2.c
U   team/oej/moduletest/channels/chan_sip.c
U   team/oej/moduletest/channels/chan_zap.c
U   team/oej/moduletest/pbx.c
U   team/oej/moduletest/utils/astman.c

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r8894 | oej | 2008-01-15 16:28:30 -0600 (Tue, 15 Jan 2008) | 135 lines

Merged revisions 8320,8347,8394,8412,8414,8418,8429,8433,8437,8445,8537,8562,8573,8588,8600,8608,8619,8632,8666,8677,8710,8729,8758,8785,8808 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r8320 | mogorman | 2006-01-20 02:00:46 +0100 (Fri, 20 Jan 2006) | 3 lines

solved problem with delayreject and iax trunking
bug 4291

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r8347 | russell | 2006-01-20 19:34:42 +0100 (Fri, 20 Jan 2006) | 2 lines

fix invalid value of prev_q (issue ASTERISK-6142)

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r8394 | tilghman | 2006-01-21 19:29:39 +0100 (Sat, 21 Jan 2006) | 2 lines

Bug 5936 - AddQueueMember fails on realtime queue, if queue not yet loaded

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r8412 | russell | 2006-01-22 00:17:06 +0100 (Sun, 22 Jan 2006) | 2 lines

prevent the possibility of writing outside of the available workspace (issue ASTERISK-6111)

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r8414 | russell | 2006-01-22 00:43:14 +0100 (Sun, 22 Jan 2006) | 2 lines

temporarily revert substring fix pending the result of the discussion in issue ASTERISK-6111

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r8418 | russell | 2006-01-22 03:05:41 +0100 (Sun, 22 Jan 2006) | 3 lines

add a modified fix to prevent writing outside of the provided workspace when
calculating a substring (issue ASTERISK-6111)

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r8429 | tilghman | 2006-01-22 09:52:49 +0100 (Sun, 22 Jan 2006) | 2 lines

Bug 6281 - Cannot set more than a single header with SIPAddHeader

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r8433 | bweschke | 2006-01-22 16:13:41 +0100 (Sun, 22 Jan 2006) | 3 lines

Bug fix: Correct some scenarios where CALL_LIMIT could not be getting adjusted properly allowing chan_sip to send calls when it really shouldn't. Bug ASTERISK-5953


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r8437 | russell | 2006-01-22 18:47:13 +0100 (Sun, 22 Jan 2006) | 2 lines

fix MixMonitor crash (issue ASTERISK-6161, probably others)

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r8445 | russell | 2006-01-22 20:03:53 +0100 (Sun, 22 Jan 2006) | 2 lines

fix memory leak from not freeing the queue member list when freeing an old queue

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r8537 | oej | 2006-01-24 14:15:13 +0100 (Tue, 24 Jan 2006) | 2 lines

Issue ASTERISK-6148 - never send response to ACK. (Reported by whiskerp)

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r8562 | oej | 2006-01-24 20:21:15 +0100 (Tue, 24 Jan 2006) | 2 lines

Issue 6114: Don't hangup on BYE/ALSO with no channel.

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r8573 | mattf | 2006-01-24 21:37:30 +0100 (Tue, 24 Jan 2006) | 2 lines

Backport fix for ASTERISK-6071, hangup on polarity reversal

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r8588 | kpfleming | 2006-01-24 23:32:09 +0100 (Tue, 24 Jan 2006) | 2 lines

ensure that channel cannot become zombie after we check but before we try to start indications

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r8600 | russell | 2006-01-24 23:55:32 +0100 (Tue, 24 Jan 2006) | 2 lines

completely arbitrary whitespace change for testing something with svnmerge ...

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r8608 | kpfleming | 2006-01-25 02:50:52 +0100 (Wed, 25 Jan 2006) | 2 lines

ensure hangup cause code is handled properly when channel does not return a frame (issue ASTERISK-6186)

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r8619 | russell | 2006-01-25 06:38:36 +0100 (Wed, 25 Jan 2006) | 2 lines

don't leak almost 200 bytes for each new channel (issue ASTERISK-6170)

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r8632 | oej | 2006-01-25 10:46:43 +0100 (Wed, 25 Jan 2006) | 2 lines

Issue ASTERISK-6276 - the "timebomb" bug. Patch by Markster over GPRS

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r8666 | russell | 2006-01-25 19:39:44 +0100 (Wed, 25 Jan 2006) | 2 lines

fix memory leak (inspired by issue ASTERISK-6190)

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r8677 | russell | 2006-01-25 20:14:43 +0100 (Wed, 25 Jan 2006) | 3 lines

don't call ast_update_realtime with uninitialized variables if we get a
registration with an expirey of 0 seconds (issue ASTERISK-6016)

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r8710 | oej | 2006-01-26 15:39:36 +0100 (Thu, 26 Jan 2006) | 2 lines

Issue 5898: Registrations does not get deleted if there's an active SIP dialog

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r8729 | russell | 2006-01-26 20:42:35 +0100 (Thu, 26 Jan 2006) | 2 lines

fix problem with dtmf on e&m (issue ASTERISK-6203)

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r8758 | tilghman | 2006-01-27 01:52:12 +0100 (Fri, 27 Jan 2006) | 2 lines

Bug 6072 - Revisions to the source bison and flex files don't auto-regenerate these files

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r8785 | oej | 2006-01-27 09:02:16 +0100 (Fri, 27 Jan 2006) | 2 lines

Issue 6362 - Register without Contact: and Expires: fails (reporter: op)

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r8808 | oej | 2006-01-28 14:52:15 +0100 (Sat, 28 Jan 2006) | 3 lines

Issue 6182 - Don't remove scheduled event until it's really done.
(reported by malverian)

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http://svn.digium.com/view/asterisk?view=rev&revision=8894