Hi,

Can anyone please comment on this?

--- "Asterisk ." <[EMAIL PROTECTED]> wrote:

> Hi Alex,
> 
> --- Alex Barnes <[EMAIL PROTECTED]> wrote:
> > You should set the type of DTMF on a per SIP PEER basis (sip.conf).
> > Then simply set the SJPhone peer to use dtmfmode=inband.
> > I have used SJPhone without problems along side Snoms that use
> > dtmfmode=rfc2833.
> 
> Thanks for the response. I know this will work, if the UACs are registered with 
> Asterisk. But
> none
> of the UACs that dial this number are registered with Asterisk. They just use the 
> sip uri to
> dial
> to that number. ie, like this: <sip:[EMAIL PROTECTED]:port>. I was trying to make 
> any sip
> client to reach this number and to the desired extension just by dialing using the 
> sip uri. 
> 
> Hope that explains the problem. Any help appreciated.
> 
> Thanks again, Girish
> 
> > 
> > HTH
> > 
> > Alex
> > 
> > -----Original Message-----
> > 
> > I have mapped a number in the default context of my dialplan. When
> > someone dials that number, it plays an IVR message and allows the caller
> > to enter 4 digit extensions. If the extension is a valid one, the call
> > wll be routed to that particular extension. 'INFO' is set as the dtmf
> > mode. This works fine if i call from a SIP UAC which sends dtmf as INFO.
> > But When i dial using SJPhone, call doesn't get routed, because SJPhone
> > uses inband dtmf. So, my problem is only people who use UACs that send
> > dtmf using the INFO method can reach the desired extension, where as
> > people who use SJPhone cannot do this. Can i make Asterisk to receive
> > both info and inband dtmf for the same number? Is this possible? If so,
> > can anyone tell me how to do that? 
> > 
> > Thanks, Girish
> > 
> 
> 
> 
>               
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