[asterisk-bugs] [Asterisk 0013932]: no dialling tone with AMI command originate & 1.6.0.1

Asterisk Bug Tracker noreply at bugs.digium.com
Fri Nov 21 10:20:23 CST 2008


A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=13932 
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Reported By:                scoles
Assigned To:                otherwiseguy
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Project:                    Asterisk
Issue ID:                   13932
Category:                   Core/ManagerInterface
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     assigned
Asterisk Version:           1.6.0.1 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
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Date Submitted:             2008-11-19 11:07 CST
Last Modified:              2008-11-21 10:20 CST
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Summary:                    no dialling tone with AMI command originate &
1.6.0.1
Description: 
when i dial a number using AMI command originate in asterisk 1.6.0.1 my
phone rings when i pick it up it connects to the number i want it to but i
get no dialling tone\noise.

in asterisk 1.4.18.1 when i use originate it works and i get a dialling
tone\noise.

attached is the sip debug from both systems.

to test i set up:

192.168.16.116 - ext:220 my phone that i originate from
192.168.16.162 - ext: 222 the phone that i dial using originate command
192.168.16.4 - asterisk 1.4.18.1 box
192.168.16.8 - asterisk 1.6.0.1 box

i dialled using the originate command setting channel variable to 220 and
exten to 222
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---------------------------------------------------------------------- 
 (0095262) otherwiseguy (administrator) - 2008-11-21 10:20
 http://bugs.digium.com/view.php?id=13932#c95262 
---------------------------------------------------------------------- 
attaching my entire /etc/asterisk directory--although all the pertinent
information in the config files should be above.  The ringing indication on
the phone in this situation, according to my captures is through the
rtp--not through any SIP signaling.  So, when you do your capture, capture
udp traffic--not just port 5060. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2008-11-21 10:20 otherwiseguy   Note Added: 0095262                          
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