[asterisk-users] setting dtmf mode for a particular peer

[Brent Davidson]

Brian J. Murrell wrote:
> On Thu, 2008-04-10 at 11:13 -0500, Brent Davidson wrote:
>   
>> You might also try "canreinvite=no" for both your phone and the sip 
>> peer.
>>     
>
> Yeah, there is definitely no re-inviting going on.  Both Asterisk and
> the local handset are in a local network behind NAT with reference to
> the SIP server that requires INFO.
>
>   
>> I think it's normal procedure for Asterisk to drop out of the 
>> call path once the call is established between two peers.  The 
>> canreinvite directive forces asterisk to remain as an intermediary, and 
>> it will probably do the transcoding that way.
>>     
>
> Indeed, this is my understanding as well, but I am definitely not
> getting a bridging of the sipphone and sip provider through a re-invite.
> The NAT would not facilitate it.
>
>   
>> If I'm not mistaken this 
>> is also useful for making calls between two system that have no common 
>> codecs.
>>     
>
> Right.
>
> I need to use ekiga or one of the ATAs here that I know support rfc2833
> so that I can eliminate this possible need to transcode inband to
> info/rfc2833 in order to narrow down the field.
>
> b.
>   

One more tidbit I just ran across in the upgrade.txt file, since you 
mention NAT:  In 1.4, you need to set canreinvite=nonat to disable 
re-invites when NAT=yes. This is propably what you want.  The settings 
are now: "yes", "no", "nonat", "update". Please consult sip.conf.sample 
for detailed information.

Let us all know how the tests go with the other phone.

-Brent
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