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Summary:ASTERISK-12540: Problem of call-back
Reporter:wu (pp_zz)Labels:
Date Opened:2008-08-08 03:49:28Date Closed:2011-06-07 14:08:14
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/General
Versions:Frequency of
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Description:my sip-telefon can out-call.wenn i use mobiltelefon ,cannot i call-back. Es exists Error:
[Jan  1 19:15:28] WARNING[6677]: chan_sip.c:6620 determine_firstline_parts: Bad request protocol 1@192.168.178.166:5060 SIP/2.0
   -- Called Teilnehmer 1
[Jan  1 19:15:28] WARNING[6677]: app_dial.c:1106 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
[Jan  1 19:15:28] WARNING[6677]: app_dial.c:1106 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
[Jan  1 19:15:28] WARNING[6677]: app_dial.c:1106 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
[Jan  1 19:15:28] WARNING[6677]: app_dial.c:1106 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
[Jan  1 19:15:28] WARNING[6677]: app_dial.c:1106 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
[Jan  1 19:15:28] WARNING[6677]: app_dial.c:1106 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
[Jan  1 19:15:28] WARNING[6677]: app_dial.c:1106 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
[Jan  1 19:15:28] WARNING[6677]: app_dial.c:1106 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
[Jan  1 19:15:28] WARNING[6677]: chan_sip.c:6620 determine_firstline_parts: Bad request protocol 10@sipgate.de SIP/2.0
   -- Called Teilnehmer 10@sipgate

how can i resolve?
Comments:By: Leif Madsen (lmadsen) 2008-08-11 14:16:37

This appears to be a configuration error on your system. You may want to use the asterisk-users mailing list, or #asterisk IRC channel to get some help on resolving this issue.

Thanks!