I have:
 
disallow=all
allow=ulaw
allow=alaw
I don't have the allow=gsm.  What is that for?
 
George


From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Marco Mouta
Sent: Tuesday, May 30, 2006 4:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] No sound?? HELP

check [general] section of your /etc/asterisk/sip.conf

disallow=all
allow=alaw
allow=ulaw
allow=gsm

This codecs depends on of your SIP provider as well as activation in your SIPphone

On 5/30/06, George A. Roberts IV <[EMAIL PROTECTED]> wrote:
I just put in a new [EMAIL PROTECTED] 2.8 system.  Trunk is connected via SIP to ViaTalk.
 
I had an older [EMAIL PROTECTED] system up and running that was working fine and I replicated settings over to the new box.  When I call 7777 from an internal SIP extension I can hear the IVR menu just fine.  However, when I call from a POTS phone to our number and it comes in via ViaTalk over SIP the call connects but I do not get any sound.  I'm sure it's a setting or something I missed, but I'm not sure what it is.  Anyone have any ideas?
 
George

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