|
Have you configured;
_ sip.conf_
..add this line:
dtmfmode=inband
..also you have uncomment the right line that
matches your dhcp setup:
localnet=192.168.0.0/255.255.0.0; All RFC 1918
addresses are local networks
;localnet=10.0.0.0/255.0.0.0 ; Also
RFC1918
;localnet=172.16.0.0/12 ; Another RFC1918
with CIDR notation
;localnet=169.254.0.0/255.255.0.0 ;Zero conf local
network
Worked for me ;)
/ Stig Henning
----- Original Message -----
Sent: Friday, September 10, 2004 2:32
PM
Subject: [Asterisk-Users] No DTMF or
Audio
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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