[Asterisk-Dev] sip / h323 transcoding

Ehsanul Karim xxxvoip at gmail.com
Fri Dec 31 17:17:34 MST 2004


 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



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