Ooh323 configuration- ooh323 phone rings but no audio when calling from pjsip device

Asterisk Asterisk Support
1

I am trying to make a call from pjsip device=> h323 device vice versa. I can dial from the sip to h323 and the h323 rings but no audio is heard. The other way around the h323 does not dial out at all. I am new to asterisk. Any help would be much appreciated .

this is the content of my ooh323.conf file
[general]
port=1720
bindaddr=0.0.0.0
gatekeeper=DISABLE
context=sets
disallow=all
allow=ulaw
dtmfmode=rfc2833
fastStart=yes
h245Tunneling=yes
h323id=PHONE_073DC9

[myfriend1]
type=peer
ip=192.168.1.100 ; UPDATE with appropriate ip address
port=1720 ; UPDATE with appropriate port
allow=ulaw
e164=1001
dtmfmode=rfc2833

Also in the ooh323 log file i got this error ERROR:Failed to connect to remote destination for transmit H2250 channel(outgoing, ooh323c_o_1, 45, 0.0.0.0)

2

Maybe try binding on the public IP instead of 0.0.0.0 ?

On the Asterisk CLI you can run “rtp set debug on” to see more details of the audio stream.

4

Port 1720 is typically the H323 signalling port, not one of the media ports.

It looks like you are only getting RTP from the PJSIP endpoint, and then sending RTP to the H323 endpoint, but it does not appear that the H323 end point is sending back any media at all. You might want to post some of the H323 signalling messages to check where it is actually going. A local firewall on your Asterisk PBX may also be the issue. Are you opening up all the RTP port range in iptables ?

5

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