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 <groberts@interjuncture.com>
wrote:>
> I just put in a new Asterisk@Home 2.8 system. Trunk is connected via SIP
> to ViaTalk.
>
> I had an older Asterisk@Home 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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