Hi,

With Asterisk and SIP phones (Cisco ATA186, Grandstream BT102), I'm having an issue with DTMF passing correctly to IVR systems like customer support phone numbers, voicemail, etc.:

1) If I set DTMF to SIP INFO, DTMF works for ISDN4LINUX calls to IVR systems with the CiscoATA186, but not with the Grandstream (this could be a bug) -- BUT the Asterisk Voicemail doesn't detect the DTMFs. SIP calls to gateways and other SIP phones work fine.
2) If I set DTMF to INBAND, there are problems with double-detection and it doesn't work with LBR codecs. However, in this mode, calls to IVR systems via I4L work with the Grandstream.
3) If I set DTMF to rfc2833, voicemail works, but ISDN4LINUX calls don't pass the DTMF to IVR systems from either the Cisco or the Grandstream.


Has anyone had any luck getting one setting to work across all channels? The problem is not in the initial call, but usually after the call is connected, the DTMF doesn't get passed.

What baffles me is why SIP INFO doesn't work with Asterisk AppVoicemail.

Any hints, suggestions, experiences would be greately appreciated.

_______________________________________________
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Reply via email to