SIP Trunk - show codec?

mutso
Posts: 6
Member Since:
2007-09-17

I hope this is an easy one. I tried searching but it was too broad to find the right answer.

I just upgrade Trixbox (Kernel Version 2.6.18-53.1.4.el5 (SMP)).

I also upgraded the FreePBX ( to core 2.4.0.1 from 2.4.0.0).

It broke my INBOUND calls. I would get a fast busy after I upgraded.

I seem to have found part of the fix. I added this to my SIP_CUSTOM.CONF:

disallow=all
allow=g729
allow=g723
allow=ulaw
allow=alaw
insecure=port,invite
useragent=PAP2T

It no longer "Fast Busies". This now RINGs for an INBOUND call. I can hear the RING, but when either voicemail or when I switch to another DID and the IVR is supposed to come on, I hear NOTHING. I am assuming this is a product of not having the right CODEC?

The odd part is I had those setting above before the UPGRADE and it worked. I am using WorldDialPoint as my provider and as I understand I should be using alaw and g723 or g729.

Is there a place in the Freepbx web GUI to see what my SIP trunk are using for codecs?

Is there a CLI command I use to show what codec the SIP trunk has registered with?



skykingoh
Posts: 996
Member Since:
2007-12-17
Quote: Is there a CLI
Quote:
Is there a CLI command I use to show what codec the SIP trunk has registered with?

CODEC's are negotiated on a per call bases.

SIP show peers, show channels and SIP show peer
should get you started from the Asterisk CLI.

Are you sure this is not a NAT problem? Did you have NAT setup before you upgraded? What version did you upgrade from?

Scott



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