Asterisk-1.4 beta2 outbound DTMF doesn't work

jrigg · September 25, 2006, 6:00pm

While testing asterisk-1.4 beta2 I note that dtmf doesn’t get sent to outbound termination provider.

I can call internal IVR and it recognizes dtmf fine. I’ve added dtmf logging in the logger.conf. Again it logs (to my cli console) the dtmf during the internal call, but when I make an outside (SIP termination) call to external IVR such as 1800COMCAST, the dtmf is not recognized and not logged to my cli console.

I’ve tried multiple termination providers (eg. viatalk,SIPPHONE) and multiple SIP phones (eg. softphones and PAP2 ATA). All work correctly with 1.2.12 and all fail with 1.4 beta2.

Also, inbound seems to work fine.

VoIP-Street · September 30, 2006, 6:49am

Works fine for me on our test system.

We push our calls through our own service VoIP Street though, not sure who your provider is.

jrigg · September 30, 2006, 2:30pm

I resolved this issue. Actually, outbound audio was failing. This was a result of the default value for “canreinvite” parameter on SIP user definitions. In 1.2.xx the default was “no”, while in 1.4.xx it appears to be “yes”. This required us to add “canreinvite=no” to all our softphone and ATA user definitions in sip.conf.

thnx