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Summary:ASTERISK-22683: SipUpdate on Local Channel will crash asterisk
Reporter:Dirk Wendland (kesselklopfer79)Labels:
Date Opened:2013-10-11 07:17:45Date Closed:2013-10-30 08:19:10
Priority:CriticalRegression?
Status:Closed/CompleteComponents:
Versions:11.5.1 Frequency of
Occurrence
Related
Issues:
is a clone ofASTERISK-22707 SipUpdate on Local Channel will crash asterisk
Environment:Attachments:( 0) backtrace.txt
( 1) gdb.txt
Description:Asterisk crashes if there is an SipUpdate on outher Channel technologies (Local)

Step 1 Atxfer
Step 2 Channel Settings after that is SipAccount/1 <-> Local;1 <-> Local;2 <-> SipAccount/2
Step 3 Send P-Asserted-Identity over SipUpdate on Local;2 ( ok that was my mistake :) )
Step 4 Asterisk crash with [Oct  9 16:56:13] NOTICE[11618] chan_sip.c: No field 'Call-ID' present to copy

Comments:By: Matt Jordan (mjordan) 2013-10-14 12:39:35.795-0500

Thank you for taking the time to report this bug and helping to make Asterisk better. Unfortunately, we cannot work on this bug because your description did not include enough information. You may find it helpful to read the Asterisk Issue Guidelines http://www.asterisk.org/developers/bug-guidelines. We would be grateful if you would then provide a more complete description of the problem. At a minimum, we need:

1. the specific steps or actions you took that caused you to encounter the problem,
2. the behavior you expected, and
3. the behavior you actually encountered (in as much detail as possible).

This likely includes output from the console with debug level logging, a SIP trace (if this is SIP related), and configuration information such as dialplan (e.g. extensions.conf) and channel configuration (e.g. sip.conf). Thanks!



By: Dirk Wendland (kesselklopfer79) 2013-10-15 06:04:38.871-0500

updated Description

By: Matt Jordan (mjordan) 2013-10-15 07:36:51.768-0500

This is not the requested information. Please read the issue reporting guidelines and provide the requested information. At a minimum, this is:

* A backtrace of the crash
* A DEBUG log with SIP trace debugging enabled showing the messages leading up to the crash

By: Dirk Wendland (kesselklopfer79) 2013-10-30 04:33:51.746-0500

Added gdb informations

By: Dirk Wendland (kesselklopfer79) 2013-10-30 05:14:06.698-0500

Online debugging
{NOFORMAT}
Breakpoint 1, get_transport_pvt (p=0x7f21ec127098) at chan_sip.c:3900
3900 if (p->outboundproxy && p->outboundproxy->transport) {
(gdb) p p->outboundproxy
$5 = (struct sip_proxy *) 0xdeaddeaddeaddead
{NOFORMAT}

By: Matt Jordan (mjordan) 2013-10-30 08:19:00.603-0500

What is this:

{noformat}
#3  0x00007ff30b3c1d47 in manager_sipupdate (s=0x7ff2fb0abb40, m=0x7ff2fb0ab200) at chan_sip.c:14940
{noformat}

There is no {{manager_sipupdate}} in Asterisk 11, nor is there an AMI action called {{SIPUpdate}}.

If you are running with custom patches, you should contact the author of those patches for bugs with those features.