Summary:ASTERISK-06949: Set TIMEOUT(absolute) breaks DIAL timeout parameter when dialing from SIP to SIP
Reporter:Ted Ritchie (tritchie)Labels:
Date Opened:2006-05-10 13:44:51Date Closed:2011-06-07 14:08:13
Versions:Frequency of
Environment:Attachments:( 0) NoOp_Timeout.log
( 1) Timeout_Set.log

I think I have identified a bug when setting an absolute timeout. It would seem that when dialing from a SIP phone to another SIP phone with an absolute timeout set, the timeout in the dial command never gets executed. If I stay on the line until the receiving phone rejects the call I will then see the dial timeout message but only because it has jumped out of the dial command.

Our usual dialplan is much more complex than what I referenced in the additional information, but I have been able to recreate the problem on a test machine with only the dialplan referenced.

I have the dialplan set an absolute timeout of 9 hours for every call using Set TIMEOUT(absolute)=32400). Usually we then spilt out our calls to our legacy Nitsuko 384i via a T1 or to our new Polycom phones. But the problem is pretty clearly just with the absolute timeout because if I remove this line the dial command starts to work properly again.

If the call comes in on a Zap channel to a SIP phone the dial command will still execute its timeout, but if the originating call was from a SIP phone to another SIP phone the dial command stops to execute its timeout.



Tested on Asterisk SVN-trunk-r14785 & SVN-trunk-r25094
Running on Linux CentOS 4.2

Two different systems each with a TE411P and using Polycom 301 and 501 phones. Configured with Realtime using PostgreSQL.

example extensions.conf

exten => _XXX,1,Set(TIMEOUT(absolute)=32400)
exten => _XXX,n(dial),Dial(SIP/${EXTEN},20)
exten => _XXX,n,Voicemail(${EXTEN},u)
exten => _XXX,n,Playback(goodbye)
exten => _XXX,n,Hangup
exten => _XXX,n+101,Voicemail(${EXTEN},b)
exten => _XXX,n,Playback(goodbye)
exten => _XXX,n,Hangup
Comments:By: Serge Vecher (serge-v) 2006-05-10 13:55:17

can you please enable sip debug and attach two different logs: one when timeout works (zap -> sip in your description) and one that doesn't (sip -> sip)


By: Ted Ritchie (tritchie) 2006-05-10 16:00:49

I have uploaded the files as requested.

By: Joshua C. Colp (jcolp) 2006-05-17 18:59:01

I was unable to reproduce this using latest trunk. Any other hints?

By: Joshua C. Colp (jcolp) 2006-05-22 10:39:44

We've gotten no feedback about this in 5 days - if something does come up please feel free to reopen it. Tootles!