Summary:	ASTERISK-12978: Drop outbound call to IVR in early media.
Reporter:	Sergey G (sgenyuk)	Labels:
Date Opened:	2008-10-29 09:44:30	Date Closed:	2011-06-07 14:03:07
Priority:	Major	Regression?	No
Status:	Closed/Complete	Components:	Channels/chan_sip/General
Versions:		Frequency of Occurrence
Related Issues:
Environment:		Attachments:	( 0) IVRPriblem.pcap
Description:	When I place outbound call to APC IVR (18008004272). Call drops in 20 sec. I have done a capture and have found that APC toll free number use IVR in early media. * does not accept it and timeout call. Sip capture attached. My debug: [Oct 29 10:17:46] VERBOSE[21859] logger.c:     -- Executing [s@macro-dialout:33] NoOp("SIP/350-9f1f3e10", "Finish if-if-if-dialout-7762-7763-7764") in new stack [Oct 29 10:17:46] VERBOSE[21859] logger.c:     -- Executing [s@macro-dialout:34] NoOp("SIP/350-9f1f3e10", "Finish if-if-dialout-7762-7763") in new stack [Oct 29 10:17:46] VERBOSE[21859] logger.c:     -- Executing [s@macro-dialout:35] NoOp("SIP/350-9f1f3e10", "Finish if-dialout-7762") in new stack [Oct 29 10:17:46] VERBOSE[21859] logger.c:     -- Executing [s@macro-dialout:36] GotoIf("SIP/350-9f1f3e10", "0?37:40") in new stack [Oct 29 10:17:46] VERBOSE[21859] logger.c:     -- Executing [s@macro-dialout:40] Dial("SIP/350-9f1f3e10", "SIP/comwave-nextone/18008004272") in new stack [Oct 29 10:17:48] VERBOSE[21859] logger.c:     -- SIP/comwave-nextone-009c96d0 is making progress passing it to SIP/350-9f1f3e10 [Oct 29 10:18:21] VERBOSE[21859] logger.c:     -- Executing [s@macro-dialout:41] NoOp("SIP/350-9f1f3e10", "Finish if-dialout-7767") in new stack [Oct 29 10:18:21] VERBOSE[21859] logger.c:     -- Executing [918008004272@from-admin:2] Hangup("SIP/350-9f1f3e10", "") in new stack [Oct 29 10:18:21] VERBOSE[21859] logger.c:   == Spawn extension (from-admin, 918008004272, 2) exited non-zero on 'SIP/350-9f1f3e10' [Oct 29 10:18:21] VERBOSE[21859] logger.c:     -- Executing [h@from-admin:1] NoOp("SIP/350-9f1f3e10", "global hangup hook") in new stack
Comments:	By: Joshua C. Colp (jcolp) 2008-10-29 09:47:26 I doubt this is a problem with Asterisk but more with the phone you are using. Please attach a sip debug with this so we can see if that is indeed the issue, as well knowing the kind of phone you are using would also be helpful. I can say for sure that if it is a Polycom then it is the Polycom that is hanging up. By: Sergey G (sgenyuk) 2008-10-29 09:52:08 Tested with two phones GS2000 and Aastra 480i. You give me good idea, I just attached a capture with external sip. I will do additional capture on the phone site. By: Jason Parker (jparker) 2009-03-17 17:17:51 Are you able to attach output of "sip debug" from Asterisk, as file requested? By: Leif Madsen (lmadsen) 2009-05-26 10:50:28 Closed due to lack of response from reporter. If you are able to provide the necessary information, then please feel free to reopen this issue. Thanks!