[Asterisk-Users] No DTMF or Audio

[Stig Thune]

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 at cavtel.com




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