[Asterisk-Users] SIP-DTMF

[Asterisk .]

Hi Alex,

--- Alex Barnes <abarnes at ubiquitysoftware.com> 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:ivr-exten at ip_of_asterisk: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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