[asterisk-bugs] [Asterisk 0013259]: Problem of call-back
Asterisk Bug Tracker
noreply at bugs.digium.com
Mon Aug 11 14:16:38 CDT 2008
The following issue has been CLOSED
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http://bugs.digium.com/view.php?id=13259
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Reported By: pp_zz
Assigned To:
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Project: Asterisk
Issue ID: 13259
Category: Channels/chan_sip/General
Reproducibility: have not tried
Severity: minor
Priority: normal
Status: closed
Asterisk Version: Older 1.4
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Disclaimer on File?: N/A
Request Review:
Resolution: open
Fixed in Version:
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Date Submitted: 2008-08-08 03:49 CDT
Last Modified: 2008-08-11 14:16 CDT
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Summary: Problem of call-back
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 at 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 at sipgate.de SIP/2.0
-- Called Teilnehmer 10 at sipgate
how can i resolve?
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Issue History
Date Modified Username Field Change
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2008-08-11 14:16 blitzrage Status new => closed
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