asterisk users - Jan 2006 - No "native bridge" on outbound SIP channels

Hi all,

I have a Cisco 1760 gateway and and Cisco 7960 VoIP phone running via
Asterisk. Both are running g711A codecs and SIP. On inbound calls I get a
native bridge, however on outbound calls I never get a native bridge. With
other SIP gateways I do get a native bridge on the outbound call. My
sip.conf is as follows:

[cisco1760]
type=friend
context=incoming
host=192.168.0.55
insecure=yes
nat=no
canreinvite=no
dtmfmode=rfc2833
disallow=all
allow=alaw

I have also confirmed while on an outbound calls that both are using the
exact same codecs. sip show channels shows

pbx*CLI> sip show channels
Peer             User/ANR    Call ID      Seq (Tx/Rx)  Form  Hold     Last
Message
192.168.0.55         123456789 4ea2e1314cd  00102/00000  alaw  No       Tx:
ACK
192.168.0.58     200         0013c427-f4  00101/00102  alaw  No       Rx:
ACK
2 active SIP channels


Anyone have an idea what's going on?
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I'm guessing that you have a similar entry in your sip.conf for the 
7960?? The 7960 has a setting for preferred codec. It defaults to g711 
U-Law. You might try changing this setting also as the 7960 doesn't know 
that you only want to speak A-Law. You will also want to make sure that 
the nat settings are disabled on both devices as they are on the same 
network. nat=never is a better choice than nat=no. You might also check 
your extensions.conf to verify that the calling from 1760 to 7960 is the 
same as from 7960 to 1760. You could also try moving both devices to 
using U-Law instead.

-Jon

Eric Bishop wrote:
> Hi all,
>
> I have a Cisco 1760 gateway and and Cisco 7960 VoIP phone running via 
> Asterisk. Both are running g711A codecs and SIP. On inbound calls I 
> get a native bridge, however on outbound calls I never get a native 
> bridge. With other SIP gateways I do get a native bridge on the 
> outbound call. My sip.conf is as follows:
>
> [cisco1760]
> type=friend
> context=incoming
> host=192.168.0.55 <http://192.168.0.55>
> insecure=yes
> nat=no
> canreinvite=no
> dtmfmode=rfc2833
> disallow=all  
> allow=alaw
>
> I have also confirmed while on an outbound calls that both are using 
> the exact same codecs. sip show channels shows
>
> pbx*CLI> sip show channels
> Peer             User/ANR    Call ID      Seq (Tx/Rx)  Form  Hold     
> Last Message  
> 192.168.0.55 <http://192.168.0.55>         123456789 4ea2e1314cd  
> 00102/00000  alaw  No       Tx: ACK       
> 192.168.0.58 <http://192.168.0.58>     200         0013c427-f4  
> 00101/00102  alaw  No       Rx: ACK       
> 2 active SIP channels
>
>
> Anyone have an idea what's going on?
>
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