Summary:ASTERISK-08742: No Idle state change message when call is transferred
Reporter:John Breen (jbreen)Labels:
Date Opened:2007-02-06 21:33:26.000-0600Date Closed:2007-04-08 18:22:42
Versions:Frequency of
Description:We are running 1.4.0 and using Grandstream GXP2000 phones.  The BLF problems are mostly fixed, BUT when a call is transferred from one extension to another, the server does not receive a new state Idle message for the first extension.

As an example, if I dial ex. 108 from 106, and then 108 transfers me to 105 then I talk to 105 and we both hangup, I see messages in the Asterisk console for state changes:

106 goes to InUse
108 goes to Ringing
108 goes to InUse
105 goes to Ringing
105 goes to InUse
105 goes to Idle
106 goes to Idle

There is no message for 108 going to Idle, so the phones don't get notified to turn off the BLF indicator and 108 appears to be busy.  This happens both when the originating caller is a SIP extension and when it is an IAX2 peer.


This behaviour has been reproduced on several servers.  We have used the limitonpeers=yes and applied call-limit=100 on our SIP peers.
Comments:By: Serge Vecher (serge-v) 2007-02-07 13:46:00.000-0600

I think this has been fixed in latest 1.4 branch -- can you please check out 1.4 from svn and report back?

By: John Breen (jbreen) 2007-02-07 15:06:51.000-0600

This _is_ the latest 1.4, isn't it - I did svn checkout http://svn.digium.com/svn/sterisk/trunk asterisk-svn on 7 February and build from that... Or do I have to go to /branches instead of /trunk?

By: Serge Vecher (serge-v) 2007-02-07 16:32:47.000-0600

ok, you've reported the version as 1.4.0, that's why I've asked.

By: Serge Vecher (serge-v) 2007-02-21 11:37:20.000-0600

hi, there have been several changes in 1.4 and trunk that may affect this. Could you please checkout the latest trunk or 1.4 (rev > 55914) and test again? Thanks!

By: Olle Johansson (oej) 2007-02-22 04:26:07.000-0600

This is a duplicate bug report, where the transferer doesn't go to idle quickly enough. It does eventually. This was fixed very recently in svn trunk and 1.4.

By: John Breen (jbreen) 2007-03-08 19:50:03.000-0600

Ok, we have updated to 1.4.1 and to latest SVN immediately afterwards.  The issue does not appear to be fixed, as we still experience the same problems.

By: Serge Vecher (serge-v) 2007-03-09 08:33:32.000-0600

We need to see a sip debug as per following, from 1.4.1 tarball please.
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 (1.4/trunk commands in parenthesis):
set debug 4 (core set debug 4)
set verbose 4 (core set verbose 4)
sip debug (sip set debug)
5) Reproduce the problem
6) Trim startup information and attach verbosedebug.txt to the issue.

By: Serge Vecher (serge-v) 2007-03-26 12:55:26

need those logs, from 1.4.2 now.

By: John Breen (jbreen) 2007-04-07 04:29:35

Ok, we have put 1.4.2 release on a client machine for testing, and it seems now to work fine.

Thanks for your help, you can close this bug now.

By: Joshua C. Colp (jcolp) 2007-04-08 18:22:41

Fixed in latest SVN of things.