asterisk users - May 2005 - Strange problem with G711/G729, Cisco and Grandstream

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@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.

Hi, I have an asterisk server without any G729 licenses, and a couple of
BT-100 phones that  actually works already with  G729 passtrought (*)
conf.
My problem, is when the BT-100 try to call to the voicemail application,
It first try G729, and then the call hang up.

My sip.conf:
[general]
disallow=all
allow=g729
allow=ulaw
allow=alaw
canreinvite=yes

snip

[XXXXXX]
type=friend
host=dynamic
secret=XXXXXX
canreinvite=yes
callerid=XXXXXXX
disallow=all
allow=g729
allow=ulaw
allow=alaw


Thanks in advance.

-- 
Ren? Mayorga
Internet & Data 
El Salvador Telecom S.A. de S.V.
Tel:(503) 247-7246
    (503) 247-7156
Cel:(503) 962-8205

> 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