We have a problem connecting to a Cisco AS5300 trunk.
We have set in freepbx the sip peer trunk to allow only g729. The call attempt is able to connect, but when answered, no audio is heard or transmitted. The only codec that can be used for this trunk is ulaw.
Our asterisk version is 1.6.2.14 . Codec is licensed, bought from Digium.
We do not have this problem on our other providers using asterisk and other non-cisco systems.
Anyone else having the same problem?
Here is our sip config for this trunk:
[xxx]
type=peer
disallow=all
allow=g729 host=xxx.xxx.xxx.xxx fromdomain=xxx.xxx.xxx.xxx
dtmfmode=rfc2833
nat=no
canreinvite=yes
context=from-trunk-sip-xxx
Thanks! We can now hear audio in our calls after adding the settings above.
The audio is not too clear though, but the provider said they will try rerouting us to the best TDM route. Any way we can improve it in the freepbx side?
Assuming you are running IOS version Version 12.3(18) or higher you need to add the lines below to your Cisco config. Cisco defaults to g729b and Asterisk uses g729a. The code below fixes that.
That depends. Is it ALL audio or just audio to specific destinations? Is it better on other routes (with other providers)? What type of Internet connection, handsets, VoIP adapters do you use? You will need to provide much more info here.
The problem seems to happen if the A channel (caller’s channel) is a landline/mobile number contacted using the same outgoing provider (as a local channel). It sounds like listening to a mono speaker on an old radio on low volume.
If I use a softphone that is directly registered to our asterisk box the audio quality improves, the words come out more clearer and louder.
I also asked my provider to test call me using their as5300 system and g729 codec and compared it with ulaw. The difference is unnoticable.