[Asterisk-Users] Strange problem with G711/G729, Cisco and Grandstream

[Jason Williams]

> But when BT-100 calls 7960 the following is happening:
> 
>    -- Executing Dial("SIP/3710-8f2b", "SIP/1707|15") in new stack
>    -- Called 1707
>    -- SIP/1707-e96a is ringing
>    -- SIP/1707-e96a answered SIP/3710-8f2b
>    -- Attempting native bridge of SIP/3710-8f2b and SIP/1707-e96a
> 
> May  4 16:46:58 WARNING[5220]: rtp.c:1545 ast_rtp_bridge: codec0 = 256 is
> not codec1 = 4, cannot native bridge.
> 
> sipsrv1*CLI> sip show channels
> 
> Peer             User/ANR    Call ID      Seq (Tx/Rx)   Format  Last Msg
> 192.168.128.171  1707        02fff7f7169  00102/00000   ulaw    Tx: ACK
> 67.126.23.251    3710        b5d3f977ea1  00101/52181   g729    Rx: ACK
> 
> When this bug is gonna be fixed?
> 

Change the codec order in the phone configuration and place g729
higher it is not asterisk doing this