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I have built latest
Asterisk w/ OpenH323 channel driver. We have a SIP softphone registered to the
Asterisk. We can place outbound calls from the SIP phone to the PSTN via
OpenH323 connection to our gatekeeper. Everything works okay - DTMF and
Audio...
But in the reverse
- if we call from a cellphone or landline the PSTN number we can get the SIP
phone to ring - we answer and can hear the originating party - but the SIP
softphone is not able to transmit DTMF or audio back to the
PSTN...
I'm not sure if
this is an issue w/ converting the signal in asterisk i.e. SIP to H323 -- or if
a problem in the codec or what?
The codec is
G711uLaw..
Help -
thanks
Robert A. Huddleston,
KF4BYY
Cavalier Telephone
LLC.
804.422.4401
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