|Summary:||ASTERISK-08898: Phone based forward will cause Asterisk to stop responding|
|Reporter:||Phil Smith (philsmith)||Labels:|
|Date Opened:||2007-02-27 15:27:04.000-0600||Date Closed:||2007-02-27 16:30:55.000-0600|
|Environment:||Attachments:||( 0) verbosedebug-trimmed.txt|
|Description:||Polycom phones running sip software 1.6.6.0036|
Asterisk 1.4 from svn Asterisk SVN-trunk-r56952
If the user uses the softkeys on the Polycom phone to set a forward to another extension, anytime a user calls that number it will cause Asterisk to stop responding.
The log capture below shows extension 234 calling 213. 213 has a forward set on the Polycom phone to extension 211. When the forward is set Asterisk will stop responding after the last line, after removing the forward from the phone the call will process as expected
-- Executing [213@iax-trunk:1] Macro("SIP/234-b7ddd1f0", "stdexten|213|SIP/213") in new stack
-- Executing [s@macro-stdexten:1] GotoIf("SIP/234-b7ddd1f0", "0?s|2:s|3") in new stack
-- Goto (macro-stdexten,s,3)
-- Executing [s@macro-stdexten:3] Dial("SIP/234-b7ddd1f0", "SIP/213|20|t") in new stack
-- Called 213
-- Got SIP response 302 "Moved Temporarily" back from 10.1.3.113
-- Now forwarding SIP/234-b7ddd1f0 to 'Local/211@iax-trunk' (thanks to SIP/213-0850a5c0)
The forward from the Polycom phone works on Asterisk 1.2.15 which was running on this server before the move to 1.4
|Comments:||By: Serge Vecher (serge-v) 2007-02-27 15:37:27.000-0600|
As per bug guidelines, you need to attach a SIP debug trace illustrating the problem. Please do the following:
1) Prepare test environment (reduce the amount of unrelated traffic on the server);
2) Make sure your logger.conf has the following line:
console => notice,warning,error,debug
3) restart Asterisk with the following command:
'asterisk -Tvvvvvdddddngc | tee /tmp/verbosedebug.txt'
4) Enable SIP transaction logging with the following CLI commands:
set debug 4
set verbose 4
5) Reproduce the problem
6) Trim startup information and attach verbosedebug.txt to the issue.
By: Joshua C. Colp (jcolp) 2007-02-27 15:49:04.000-0600
For future reference trunk is *not* 1.4. You should not be using trink in production. To get 1.4 use svn co http://svn.digium.com/svn/asterisk/branches/1.4 asterisk.
By: Phil Smith (philsmith) 2007-02-27 16:15:31.000-0600
verbosedebug-trimmed.txt is now available per serge's instructions, sorry I did nto get them to you first. In this dump 214 is set with the phone forward to extension 431. Extension 204 calls 214 and Asterisk stops responding.
file - we were mistaken then since some of the commands had changed (no longer sip debug but core set debug) we assumed we were running 1.4 from the trunk not 1.2. Or is trunk something completely different from either of those?
By: Joshua C. Colp (jcolp) 2007-02-27 16:17:34.000-0600
Fixed in trunk as of revision 57011.
By: Joshua C. Colp (jcolp) 2007-02-27 16:19:22.000-0600
trunk is where all new development happens, anything currently flys there. Some days it might work fine, some days it might crash every second. That's why we have:
NOTE: This is a development version of Asterisk, and should not be used in
As part of the banner.
By: Serge Vecher (serge-v) 2007-02-27 16:30:55.000-0600
on the other hand, thanks for trying the trunk out and helping us catch the issues there -- as someone said, trunk tends to become a release ;)