|Summary:||ASTERISK-08202: Transfer on a Polycom phone does not set hint to Idle on transfer completion|
|Reporter:||Anthony Rodgers (cunningpike)||Labels:|
|Date Opened:||2007-01-18 16:01:31.000-0600||Date Closed:||2007-06-30 09:20:05|
|Environment:||Attachments:||( 0) verbosedebug.txt|
( 1) verbosedebug-53434.txt
|Description:||4010 is monitoring hints for 4011 and 4012|
4011 calls 4012 - both BLFs light
4011 presses 'Transfer' and calls another number
The other number answers and 4011 presses 'Transfer' again
The BLF on 4010 for 4011 stays lit when the call on that phone ends. CLI does not show a message to say 'Extension Changed 4011 new state Idle for Notify User 4010'
The BLF for 4012 behaves normally (i.e. goes out) when the call eventually terminates, but 4011 stays lit until asterisk is restarted (not even a subsequent regular call resets it).
Regular calls work fine.
****** ADDITIONAL INFORMATION ******
During the post-transfer call, this message repeats on the CLI:
Packet2Packet bridging SIP/4011-0973aa70 and SIP/4012-09823e60
These are the parties to the _original_ pre-transfer call.
|Comments:||By: Anthony Rodgers (cunningpike) 2007-01-18 16:02:36.000-0600|
Summary should be:
Transfer on a Polycom phone does NOT set hint to Idle on transfer completion
By: Serge Vecher (serge-v) 2007-01-25 12:23:39.000-0600
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) Trim startup information and attach verbosedebug.txt to the issue.
By: Anthony Rodgers (cunningpike) 2007-01-25 18:28:28.000-0600
Will do - I'm off until Monday, so I'll attach the info then.
By: Olle Johansson (oej) 2007-01-26 02:32:55.000-0600
please include "show hints" and "sip show inuse" before and after the transfer. thanks.
By: Olle Johansson (oej) 2007-02-01 15:31:49.000-0600
Please check with latest 1.4 from svn. Thanks.
By: Serge Vecher (serge-v) 2007-02-05 13:35:45.000-0600
CunningPike: is this from the absolute latest revision of 1.4 as requested by oej (hint: always specify revision when posting debugs).
By: Anthony Rodgers (cunningpike) 2007-02-05 13:50:49.000-0600
My bad - I meant to post the SVN version - 53099
By: Serge Vecher (serge-v) 2007-02-07 10:25:38.000-0600
ok, there were some changes after that, please test with rev at least 53350 or greater.
By: Anthony Rodgers (cunningpike) 2007-02-07 12:29:49.000-0600
Much better with SVN-branch-1.4-r53434M - however, the changing of the state of the transferor (4011 in our scenario) doesn't take place until after the transfered call ends, and is delayed for a few seconds after that.
Need another trace?
By: Serge Vecher (serge-v) 2007-02-07 16:26:36.000-0600
By: Serge Vecher (serge-v) 2007-02-08 16:23:15.000-0600
interesting ... we accept transfer (REFER) from 4011 at 12:30:12, we BYE 4011 at 12:30:12 the first time around, but NOTIFY to 4010 is not triggered until after we send the second BYE to 4011 out of the blue at 12:30:44.
By: John Laur (gork) 2007-02-13 10:38:52.000-0600
As hints with SIP devices seem to be currently totally broken (see http://bugs.digium.com/view.php?id=8800) I would be very curious to know how you have asterisk 1.4 or trunk configured to even update the hints of BOTH the called and the calling phones. The best I can currently do is to get the hint to update for only the CALLED phone. The calling phone never shows InUse in 'show hints' despite asterisk indicating '1' under the user heading of 'sip show inuse'
I do have this transfer problem, but I would be almost positive that fixing hints on SIP devices on a larger scale would correct it. More to the point the problem seems to be that state changes on sip USERS do not affect the hint state but sip PEERS do. Shame really -- everything worked so well on 1.2 ... well except for that nagging "500 Internal Server Error" problem in the polycom phones themselves...
By: Olle Johansson (oej) 2007-02-13 10:48:11.000-0600
Saying that it's totally broken when it's working for a lot of other people does not help.
The bug tracker is not for discussion, if you add stuff here, please relate it to the reported bug and help us fix that. General issues is what we have mailing lists for.
By: Olle Johansson (oej) 2007-02-13 10:50:38.000-0600
In ASterisk 1.2, only peer states was published in a subscription, exactly as it is in 1.4.
By: John Laur (gork) 2007-02-13 10:54:53.000-0600
I apologize -- it's just the result of frustration after messing with the problem for 6 hours and seeing no result. I should have said instead that none of the documented possible fixes for any hint problems having to do with SIP channels appear to work for me under 1.4 or trunk. I am of course continuing to work on it and hopefully instead of making unfair criticisms I will instead be able to provide something more useful.
By: Olle Johansson (oej) 2007-02-13 10:59:12.000-0600
That debug file is really something to dig into :-)
By: Olle Johansson (oej) 2007-02-14 10:52:08.000-0600
Created a new branch with a possible fix, please try
svn checkout http://svn.digium.com/svn/asterisk/team/oej/bug8848-blinktransfer/ transferblink
By: Olle Johansson (oej) 2007-02-16 06:22:28.000-0600
Have you tested the branch yet?
By: Francesco Romano (francesco_r) 2007-02-16 08:05:32.000-0600
I had the same problem of CunningPike using Snom phones, but with oej test-branch hints now works well and are properly refreshed after a transfer.
By: Anthony Rodgers (cunningpike) 2007-02-16 12:39:51.000-0600
Hi Olle - sorry, not yet, and I'm not back in the office until Tuesday. I'll test it then.
By: Olle Johansson (oej) 2007-02-18 04:23:19.000-0600
francesco-r: Thanks for the response.
CunningPike: You're on hold until tuesday :-)
By: Anthony Rodgers (cunningpike) 2007-02-20 18:19:11.000-0600
Asterisk SVN-oej-bug8848-blinktransfer-r55691M works great - thanks, oej!!
By: Olle Johansson (oej) 2007-02-21 02:33:46.000-0600
Fixed in 1.4 rev 55834 and svn trunk.
Thanks for testing CunningPike and francesco-r !