I make a call in the following way
H323 Softphone–>Asterisk–>Sip Operator
For this call flow I get one way audio (H323 client doesn’t hear remote side). As debug shows (rtp debug) Asterisk doesn’t receive RTP stream from Sip operator. I see only packets from H323 and to Sip Operator.
For H323 client I use openphone and SJphone. Asterisk uses ooh323 which is in Asterisk-addons. I tried both 1.2 version and 1.4.
As I see from sip debug Sip Operator uses SER.
I tried to use different codecs ( gsm, g711 etc.). As I see from sip debug codecs are supported by remote side.
if I use these schemes everything is ok:
H323 Softphone–>Asterisk–>Sip Phone
Sip Phone–>Asterisk–>Sip Operator
I don’t use firewalls.
Can anybody help ?
I can show any debugs if u need.