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

[Irakli Natsvlishvili]

Hello everybody,

Further interesting details about BT-100, * and Cisco 7960.

Asterisk has G729 installed, on BT-100 there is g729 selected on all codec
selections. On Cisco 7960 preferred codec is g711.

Form sip.conf


[1707]
;---------> Cisco 7960
context=default
type= friend
username=1707
host = dynamic
dtmfmode=rfc2833
qualify=2000
disallow=all
allow=g729
allow=ulaw

[3710]
; -----------------> GrandStream Bt-100
context=default
type=friend
username=3710
user=phone
host=dynamic
dtmfmode=rfc2833
mailbox=1710 at default
qualify=2000
disallow=all
allow=g729
allow=ulaw

When 7960 calls BT-100 there is g729 used on both ends. 

sipsrv1*CLI> sip show channels

Peer             User/ANR    Call ID      Seq (Tx/Rx)   Format  Last Msg
67.126.23.251    3710        118e46ce79a  00103/00000   g729    Tx: ACK
192.168.128.171  1707        00070ef7-36  00102/00101   g729    Tx: ACK

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?

I.N.