[Asterisk-Users] SIP-DTMF

[Alex Barnes]

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.

HTH

Alex

-----Original Message-----
From: Asterisk . [mailto:asterisk_in at yahoo.com] 
Sent: 27 October 2004 13:49
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] SIP-DTMF


Hi,

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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