Asterisk unable to receive DTMF tone from sip client. Im using the (d) flag in dial application to perfume one digit exit during ringing state. But unfortunately doesnt work. Here is my sip configuration :-[100]type=friend username0host=dynamic nat=..

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How can I update asterisk to send back move temporarily with updated IPaddress to incoming INVITE.i.e, Incoming call from ITSP to server 1 with x DID and there is a need to update the ITSP that the specified x DID number is allocated in server 2. Thanks,..

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