[asterisk-bugs] [Asterisk 0013259]: Problem of call-back

[Asterisk Bug Tracker]

The following issue has been SUBMITTED. 
====================================================================== 
http://bugs.digium.com/view.php?id=13259 
====================================================================== 
Reported By:                pp_zz
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   13259
Category:                   Channels/chan_sip/General
Reproducibility:            have not tried
Severity:                   minor
Priority:                   normal
Status:                     new
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:              
====================================================================== 
Date Submitted:             2008-08-08 03:49 CDT
Last Modified:              2008-08-08 03:49 CDT
====================================================================== 
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?
====================================================================== 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2008-08-08 03:49 pp_zz          Asterisk Version          => Older 1.4       
2008-08-08 03:49 pp_zz          SVN Branch (only for SVN checkouts, not tarball
releases) => N/A             
======================================================================