[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
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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
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2008-11-21 10:20 otherwiseguy Note Added: 0095262
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