|Summary:||ASTERISK-07283: Cisco 7912 phone continues ringing after sip call has finished|
|Date Opened:||2006-07-04 19:37:09||Date Closed:||2011-06-07 14:07:46|
|Environment:||Attachments:||( 0) cisco7912|
|Description:||We had a requirement to ring one group of phones immediately after another. If the second group contains the same 7912 phones as in the first group then these phones will continue ringing even though the call was cancelled for those phones (since it was picked up by somebody else in the group). Occasionally the display on the 7912 phones also locks up and the phone has to be rebooted. (The 7912 phones were running the latest firmware v1.3.1) I was able to simulate the problem with the following simple extensions.conf statements:|
exten => 5353,1,Dial(sip/5353,2)
exten => 5353,2,Dial(sip/5353,2)
Looking at the supplied ethereal trace you can see that the INVITE for the second dial is send before the INVITE for the first dial has completed its cancel. This seems to be what causes the trouble. Simply putting a 2second delay between dials is a simple workaround but doesn't stop the problem completely since it is still possible for this condition to occur when there is >1 caller.
|Comments:||By: Olle Johansson (oej) 2006-07-05 01:09:14|
Thank you for reporting this. You always need to read the bug guidelines, that says that we need debug output from your asterisk, so we can see what's going on inside your Asterisk. Read them again, and add the file as an attachment to this bug report. Thanks. /Olle
By: Olle Johansson (oej) 2006-07-05 01:44:31
Looking at the pcap file, there seems to be proper signalling. What is the bug on the Asterisk side?
By: crossml (crossml) 2006-07-05 01:58:56
I logged this one as an interoperability issue. Problem is that the Cisco 7912 continues to ring despite the call be cancelled. Guess I should report it to Ciso eh?
By: Olle Johansson (oej) 2006-07-05 14:23:28
Yes, I would recommend that. Seems like their phones got a bug here.
By: Olle Johansson (oej) 2006-07-05 14:23:59
Please re-open if you locate a bug in Asterisk signalling here. THanks. /Olle