[Asterisk-Users] No "native bridge" on outbound SIP channels

[Jonathan Feally]

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 <vulture at netvulture.com 
> <mailto:vulture at netvulture.com>> 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 <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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