[asterisk-users] H323 situation

[Luis Silva]

Hi all,

I have this installation:

Asterisk 1.6.1.1  with h323 support, pwlib_v1_10_3 and openh323_v1_18_0. 

I have a  problem that is, when a call comes from H323 and goes to a Sip
phone the asterisk sends two rtp streams to the sip. I checked this with
tcpdump, save the payload (voice is in G711u), one is the ringing indication
and the other is the voice coming from the user in h323 side. And worst they
go to the same port. This causes that in the sip phone there are problems,
when the call is answered sometimes we get the riging indication, others a
mix of the two with very bad sound quality and others(few) a god audio call.


The outgoing calls from sip to H323 are ok.

I also tested an incoming call from a dahdi channel and from here everything
is ok, only one rtp stream and a good call.

 

By the way I had other problem that I fixed, but don't know if it was in the
best way. 

The h323 box is a Cisco AS5300 (or 5350?) and when I was making outgoing
calls the AS disconnected all of them after 10 sec. 

 I investigated I noticed that the AS as a limitation to the G711 payload to
20 ms, and asterisk was using 150 ms. I resolve this changing in frame.c the
codec value and recompile asterisk. There is simpler way to do this? Like
changing values in codec.conf?...

 

Regards

LS 

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