Summary:ASTERISK-18991: CLONE - Channel SIP do not get answer
Reporter:Vitaliy Vitaliy (vetal)Labels:
Date Opened:2011-12-08 05:10:35.000-0600Date Closed:2011-12-12 08:37:42.000-0600
Versions: Frequency of
is a clone ofASTERISK-18940 Channel SIP do not get answer
Environment:Centos 5.4 and FreeBSD 8.1Attachments:( 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


   -- 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.