| Summary: | ASTERISK-19004: RTP timeout won't work with locally bridged SIP channels | ||||
| Reporter: | Andrey Solovyev (corruptor) | Labels: | |||
| Date Opened: | 2011-12-12 00:38:13.000-0600 | Date Closed: | 2012-01-24 10:23:52.000-0600 | ||
| Priority: | Minor | Regression? | |||
| Status: | Closed/Complete | Components: | Channels/chan_sip/General | ||
| Versions: | 1.8.7.1 | Frequency of Occurrence | Constant | ||
| Related Issues: |
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| Environment: | Attachments: | ||||
| Description: | It's simple to reproduce. I have rtptimeout=30 in [general] and I have two sip peers 1001 and 1002 [1001] username=1001 type=friend secret=secret qualify=no nat=no host=dynamic context=internal call-limit=10 dtmfmode=rfc2833 callerid=1001<1001> canreinvite=no disallow=all allow=alaw 1002 calls 1001 with simple Dial application without any options and MixMonitor. So we get Locally bridging SIP/1001-00000008 and SIP/1002-00000009 Disconnect 1001 or 1002 from the network. Asterisk won't see any RTP from the phone and won't hangup the call after 30 seconds. If I do channel request hangup SIP/1001-00000008 I get this notice. NOTICE[8378] chan_sip.c: Disconnecting call 'SIP/1001-00000008' for lack of RTP activity in 233 seconds This seems to be unexpected behaviour. | ||||
| Comments: | By: Stefan Schmidt (schmidts) 2011-12-15 07:59:02.028-0600 could you please try to reproduce this with asterisk 10. Rtp timeout is completly handled different there. thanks By: Andrey Solovyev (corruptor) 2011-12-20 03:40:19.790-0600 asterisk 10 successfully disconnects call in the same situation. | ||||