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

Asterisk Bug Tracker noreply at bugs.digium.com
Thu Nov 20 16:38:48 CST 2008


A NOTE has been added to this issue. 
====================================================================== 
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:              
====================================================================== 
Date Submitted:             2008-11-19 11:07 CST
Last Modified:              2008-11-20 16:38 CST
====================================================================== 
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
====================================================================== 

---------------------------------------------------------------------- 
 (0095221) scoles (reporter) - 2008-11-20 16:38
 http://bugs.digium.com/view.php?id=13932#c95221 
---------------------------------------------------------------------- 
have set the system up like you said above - all i had was extensions.conf,
sip.conf & manager.conf (so i could log onto the manager) in my config dir

still the same, even tried your exact originate command with and without
events.

tried it on the aastra 55i firmware: 2.4.0.96
and on the polycom ip soundpoint 320 firmware: 3.1.1.0137

i ported my set up to the asterisk now 1.02  (asterisk version 1.4.18.1)
and i got a ringing sound on my phone

you gave me your sip.conf and extension.conf file what other conf file
were in the asterisk dir can you show me them so i can use thos as well.

how does the phone get the dialling sound, i guess asterisk must tell it
to give out a dialling sound using SIP signalling. if so whats the
different between 1.4.18.1 and 1.6.0.1 way it tells a phone to give out a
dialling sound. could the phone not be understanding the SIP packet to give
out a dialling sound?

also i found i needed to mute the other phone ringing noise as not to
mistake it for my phones dialling noise

are you trying it on softphones as i'm doing this on a hardphone?  will
try a softphone tomorrow, retry it on the polycom and setup you test system
on my old 1.4.18.1 box 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2008-11-20 16:38 scoles         Note Added: 0095221                          
======================================================================




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