Summary: | ASTERISK-18991: CLONE - Channel SIP do not get answer | ||||
Reporter: | Vitaliy Vitaliy (vetal) | Labels: | |||
Date Opened: | 2011-12-08 05:10:35.000-0600 | Date Closed: | 2011-12-12 08:37:42.000-0600 | ||
Priority: | Major | Regression? | |||
Status: | Closed/Complete | Components: | Channels/chan_sip/General | ||
Versions: | 1.8.7.1 | Frequency of Occurrence | |||
Related Issues: |
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Environment: | Centos 5.4 and FreeBSD 8.1 | Attachments: | ( 0) bad.dump ( 1) myDebugLog | ||
Description: | I hear 1 word, and then silense, tcpdump look to be OK. This do not happend on Asterisk 1.6 Verbose: -- Executing [74957410001@test:1] Dial("SIP/00002-00000000", "SIP/ez/01#7495 7410001") in new stack == Using SIP RTP CoS mark 5 -- Called SIP/ez/01#74957410001 -- SIP/ez-00000001 is ringing -- SIP/ez-00000001 is making progress passing it to SIP/00002-00000000 [Nov 30 22:43:50] NOTICE[27741]: chan_sip.c:20192 handle_response_peerpoke: Peer 'ez' is now Lagged. (1ms / 0ms) -- No one is available to answer at this time (1:0/0/0) -- Auto fallthrough, channel 'SIP/00002-00000000' status is 'NOANSWER' | ||||
Comments: | By: Vitaliy Vitaliy (vetal) 2011-12-08 05:11:37.877-0600 Dear developers, as for this issue. I also have good, success calls with same parametres. Please help to solve this issue. Thank you By: Matt Jordan (mjordan) 2011-12-12 08:37:36.291-0600 As was previously noted in ASTERISK-18940, this is a configuration issue. Questions or problems regarding configuration should be asked in the #asterisk IRC channel, or the asterisk-users list. |