[Asterisk-Users] Problem with bridging SIP to OH323 and SIP to SIP: Bridge stops bridging

[Lukas Macura]

Hello everybody,
        
        please can somebody help me with my problem ? I have asterisk
        1.0.7 with
        SIP users on one side and OH323 0.6.6pre3 on network side.
        Everything
        works fine except some random errors that sound is working only
        in one
        direction. I think it does not correspondf with nat, it happens
        on SIP
        clients which are not behind nat too.
        
        I found that it happens when communicating through oh323 even
        with sip to sip. When I look into log, I can find (I think only
        this is
        corresponding to my error):
        
        Asked to transmit frame type 8, while native formats is 256
        (read/write
        = 256/256)
        ..
         OH323/R17209: Format changed to g729 (native alaw).
        Ooh, format changed from unknown to g729
        ..
        Invalid data (4 bytes at the end)
        ..
        Didn't get a frame from channel: SIP/591234567-b606
        ..
        Bridge stops bridging channels OH323/R18863 and
        SIP/5971234567-b606

        I bought and use native g.729 codec from digium. 

        Please can somebody point me to right place ? I tried almost
        everything. 
        
        I found in sources that this happens when ast_read fails. There
        is ast_waitfor_n before it. Can it be some problem with
        timeouts? Do you think it is possible to increase some number in
        ast_waitfor_n to wait longer time? Or am I absolutely on bad
        place?
        
        May it be problem with some network connectivity ? Or codec
        incompatibilty ?
        
        I saw this message only here, not on my another asterisk
        installation... 
        
        Thank you very much,
        
        Lukas Macura