[Asterisk-Dev] sip / h323 transcoding

[Ehsanul Karim]

 Hi,

    I am translating sip any codec to h323 (with OH323) codec 723.1 .
Everything works fine , except there is  a one way audio. It is same
even I recv call on H323 723. If we are using 729 for H323 receiving
or terminating , everything works fine.

  Only if you are using 723 at the H323 , you are lost with one way
audio. 723 works fine with other SIP & iAX.

thanks.

Ehsan