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
[EMAIL PROTECTED]
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/ANRCall ID Seq (Tx/Rx) Format Last Msg
67.126.23.2513710118e46ce79a 00103/0 g729Tx: ACK
192.168.128.171 170700070ef7-36 00102/00101 g729Tx: 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/ANRCall ID Seq (Tx/Rx) Format Last Msg
192.168.128.171 170702fff7f7169 00102/0 ulawTx: ACK
67.126.23.2513710b5d3f977ea1 00101/52181 g729Rx: ACK
When this bug is gonna be fixed?
I.N.
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