You will probably want canreinvite=yes on your sip entries unless you are going to be using monitoring or some other feature in which asterisk needs to hear the conversation. Also, Is asterisk answering the call from the 7960 or is the 1760 doing it through the dial cmd? If asterisk answers the call, then this could be part of the problem.

Can you send an output of the console for a call from 1760 -> 7960 with a
show channel for each SIP device, and then the same thing for 7960-1760.

-Jon

Eric Bishop wrote:
Yes the 7960 is also set only to use alaw. I was under the impression though that nat=yes did not effect this. And if it does why does it native bridge ok on inbound calls with the same nat=yes




On 1/15/06, Jonathan Feally <[EMAIL PROTECTED]> wrote:
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
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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