Summary:	ASTERISK-05348: Inbound PRI call not handle if called SIP exten is registered but offline
Reporter:	sohosys (sohosys)	Labels:
Date Opened:	2005-10-21 19:10:07	Date Closed:	2011-06-07 14:03:05
Priority:	Major	Regression?	No
Status:	Closed/Complete	Components:	Core/General
Versions:		Frequency of Occurrence
Related Issues:
Environment:		Attachments:
Description:	sip user (friend) registers with asterisk, registraion is recorded in MySQL (realtime), sip user network connection is lost, but registartion is still active. call orignates on PRI for the disconnected, but registered SIP user, dial plan is as follows; exten => s,1,dial(sip/##########|20) exten => s,2,voicemail(u##########) exten => s,3,hangup() call should go to voicemail, since there is no response from the sip extension, instead the inbound zap channel gets 20 seconds of silence, and the telco switch drops the call (no progress) well before the 20 seconds elapses. When calling this same extension from another SIP extensions the behavior is similar, but acceptable - silence for 20 seconds, then voicemail. call progress on the PRI originated call looks like this;    -- Executing Dial("Zap/1-1", "sip/##########|20") in new stack    -- Called ##########    -- Channel 0/1, span 1 got hangup  == Spawn extension (macro-ring-subscriber, s, 6) exited non-zero on 'Zap/1-1' in macro 'ring-subscriber'  == Spawn extension (pri-inbound, 3037687410, 1) exited non-zero on 'Zap/1-1'    -- Hungup 'Zap/1-1' then caller never gets voicemail (channel is hung up), and the telco switch provides ringing to the caller (call not released properly?). Seems like the correct behavior would be that the SIP invite times out before the telco PRI progress timer does and the dial plan progresses to next step (voicemail). ****** ADDITIONAL INFORMATION ****** ISDN PRI connected to LEC Lucent 5ESS, SIP extension is polycom IP 501
Comments:	By: Olle Johansson (oej) 2005-10-22 04:19:12 Try this with a current CVS head and see if the same thing happens. Thank you for testing! By: Olle Johansson (oej) 2005-10-22 04:19:55 You can also use the sippeer dialplan function to check if you have reachability before you place the call with dial, if you have qualify=yes for that peer. By: Olle Johansson (oej) 2005-10-22 04:21:09 To satisfy the telco switch that times out, you could possibly send a playback(<1-second-of-sílence|noanswer) before you dial to send some call progress to them... By: Mark Spencer (markster) 2005-10-22 21:33:26 This is a configuration issue, not a bug.