I am taking over an asterisk system from another person and having an issue
where a sip trunk is restricting the outgoing codecs to just g.729
I am dialing in from a Cisco 7960. The Invite from the Cisco has the
folowing M line:
m=audio 17022 RTP/AVP 18 0 8 101.
So it is allowing g.729, ulaw and alaw.
Asterisk is tandeming the call out over a SIP trunk
sip.conf tandem trunk config:
But the outgoing Invite has the following m line:
m=audio 17064 RTP/AVP 18 101.
This system does realtime which I am not really familiar with but the only
stuff that seems relivent is one table called sip_devices with 2 columns
disallowd and allowed. I think this should only affect the phones though.
For this extension the values are disallowed=all and allowed=g729;ulaw;alaw
I did try to search here and Google but I am not sure what to use for a
I turned on debug to level 3 :