H323 Codec issue between pbxs

Unanswered Question
Jan 15th, 2010

I have the following:

Shoretel ----SIP---3825-----h323------3745----h323---CUCM

For whatever reason, I can get the call to go through, but it seems to drop on the codec after you answer the call.

I have a similar setup...     PBX---H323---h323----CUCM.    This setup, I have no issue.

On the dial peer at the 3825.. I am forcing G711

dial-peer voice 809 voip
destination-pattern 809T
progress_ind setup enable 3
session target ipv4:
dtmf-relay h245-alphanumeric
codec g711ulaw
ip qos dscp cs5 media

Im not super exciting about, but it seems to be the only way the Shortel via SIP will work.  

Where Im cloudy on is if the 3825 router itself can transcode a G711 call to G729 (from SIP to H323)... or is this not possible and we need to keep it G711.

I just added this site and it seems to give me grief with the rest of my point to point H323 calling legs.

When use the CSIM Start and the dial peer, the call goes through still, but then later, it gives

csim err csimDisconnected recvd DISC cid(1401)
csim: loop = 1, failed = 1
csim: call attempted = 1, setup failed = 1, tone failed = 0

On another router, I get it to connect, and I can see th codec of the IP phone sending G729 and receiving G711.  ( even get the cool beep!)

Not sure what I should adjust on here.

I have this problem too.
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lusandi Sat, 01/16/2010 - 14:56

Hello, in order to provide the transcode from g711 to g721, you will need to configure a dsp to do that, so in order to confirm that you will need to verify if you have all the dsp's that you need.

also the call goes trough because they will use the default call leg in order to proceed.

And you can debug  the conection on the h323 gateways in order to confirm the disconnect cause of the call.

If you need something else please let me know.



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