For ABE support you really should contact Digium. BTW, there is no such thing as a "sip trunk". It's a sip peer or connection or account.
Noah Miller wrote: > Hi Jared - > >> > For the first time, I'm setting up SIP trunking between two asterisk >> > boxes. The calls themselves work fine, but I'm not able to get DTMF >> > working. >> >> If you're connecting an Asterisk 1.2 box to an Asterisk 1.4 box (as it >> appears that you are), you'll need to set "rfc2833compensate=yes" in the >> peer or friend section of sip.conf on the Asterisk 1.4 box. > > Unfortunately, this didn't work. Maybe rfc2833compensate isn't > available in ABE? > > I think this may require inband signalling anyway, as we'll require > non-sip (zap) devices to be able to use these sip trunks and enter > DTMF. > > Any other ideas? > > Thanks! > Noah > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
