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

[Asterisk Bug Tracker]

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:                
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Project:                    Asterisk
Issue ID:                   13932
Category:                   Core/ManagerInterface
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     new
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:              2008-11-19 18:40 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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 (0095145) otherwiseguy (administrator) - 2008-11-19 18:40
 http://bugs.digium.com/view.php?id=13932#c95145 
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I am unable to duplicate the lack of ringing tone in 1.6.0.1 with:
extensions.conf
------------
[default]
exten => _600X,1,Dial(SIP/${EXTEN},30)

sip.conf
------
[test](!)
type=friend
host=dynamic
context=default
secret=test
canreinvite=no
nat=yes

[6001](test)
[6002](test)
[6003](test)

orignate command
---------------
Action: Originate
Channel: SIP/6001
Context: default
Exten: 6002
Priority: 1

After the originate, 6001 rings.  I answer.  I hear ringing on the line
until I answer 6002.  I tested on 1.4, 1.6.0 SVN, 1.6.0.1 tarball and
trunk.  Can you reduce this to the simplest extensions.conf and sip.conf
settings that will reproduce the issue and post back with that info? 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2008-11-19 18:40 otherwiseguy   Note Added: 0095145                          
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