| Summary: | ASTERISK-16024: Call gets disconnected after approx 20 seconds of ringing | ||
| Reporter: | Gareth Blades (gblades_skymarket) | Labels: | |
| Date Opened: | 2010-04-28 07:18:04 | Date Closed: | 2011-06-07 14:01:03 |
| Priority: | Major | Regression? | No |
| Status: | Closed/Complete | Components: | Channels/chan_sip/General |
| Versions: | Frequency of Occurrence | ||
| Related Issues: | |||
| Environment: | Attachments: | ( 0) sip_debug_-_call_hagup_afer_brief_ringing | |
| Description: | We have a system where calls come in over ISDN and then get routed out over SIP. The SIP carrier gives a SIP/183 response to indicate call progress but after approx 20 seconds of ringing asterisk disconnects the call. I am not specifying a duration to the dial command so it should not timeout. We were running an older version of asterisk (1.4.22) and at that time we manually modified sip.conf to make it consider a 183 to be the same as a 180 response which fixed this problem but it can cause other issues itself) | ||
| Comments: | By: David Woolley (davidw) 2010-04-28 08:15:26 -- Channel 0/31, span 1 got hangup request, cause 19 This was closed from the other end of the circuit switched connection, with a NO ANSWER status. Moreover, as the trace shows that: [Apr 28 12:10:08] DEBUG[22107]: chan_dahdi.c:5547 dahdi_indicate: Received AST_CONTROL_PROGRESS on Zap/31-1 this has to be a dahdi problem, or remote network problem. To be clear. Asterisk is not initiating the close; whatever is at the other end of the ISDN is doing so. I tend to suspect the remote network, making this a support question, not one for this bug tracker. By: Paul Belanger (pabelanger) 2010-04-28 08:29:53 I agree with davidw, this is not a bug, but a support issue. ISDN Cause Code 19: No answer from user (user alerted) The destination responds to the connection request but fails to complete the connection within the prescribed time. The problem is at the remote end of the connection. --- Thanks for your comments. This does not appear to be a bug report and we are closing it. We appreciate the difficulties you are facing, but it would make more sense to raise your question in the support tracker, http://www.asterisk.org/support | ||