[Asterisk-Users] SIP <--> H323 RTP Questions (1 WAY Audio only)

[Ken Chan]

Hello,
I have been trying to get the SIP <--> H323 working in the last few weeks.  I tried different H323 channel drivers.  I need help badly.

I got SIP <--> SIP (with canreinvite=yes) and it was working fine.  So, I believe the problem is not in SIP side.

Here are my problems:

a)  I am currently using Asterisk-Addon ooh323 channel driver.  I dial from SIP to OpenPhone.
I have 1 way audio only (the voice from OpenPhone to SIP is fine.  But there is no voice from SIP to OpenPhone).  The signalling part looks good.  At least I could make a call and hang up the call.  Anyone has any idea why I had 1 way audio?  All the phones and Asterisk are on the same LAN.

Here is part of my ooh323.conf file:
[general]
port=1720
bindaddr=10.3.3.239

[ken_op]
type=peer
context=default
ip=10.1.1.155
port=1720
allow=ulaw
dtmfmode=rfc2833

Here is part of my extensions.conf file:

exten => 6111,1,Dial(SIP/voip6111,20)
exten => 7401,1,Dial(OOH323/ken_op,20)


b)After I established a call, I typed "rtp debug" to enable the debug for RTP.  Seems to me that the RTP packets ARE GOING through Asterisk.  It is normal?

Can someone that had success on setting up SIP <--> h323 (Asterisk-Addon 1.2.3) please provide me some more information (such as conf files and hints how to solve my problems).

Ken

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