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 -----
From: Huddleston, Robert
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Cc: Patterson, Mike
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
rhuddleston@cavtel.com
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