I have a VG200 series gateway that registers to a SIP host for service to an FXS port. There is no Communications Manager or CME involved, this is just a VG200 acting as a SIP gateway/endpoint for a fax machine on an FXS port.
I am trying to put together a dialplan that will allow the fax machine to dial out with 7,10, or 11 digits for US dialing. The client wants these patterns available without an access code. So I used the following dial-peers.
dial-peer voice 100 voip
destination-pattern 1[2-9]..[2-9]......
session protocol sipv2
session target dns:siphost.net
session transport udp
dtmf-relay rtp-nte digit-drop
codec g711ulaw
no vad
!
dial-peer voice 101 voip
destination-pattern [2-9].....T
session protocol sipv2
session target dns:siphost.net
session transport udp
dtmf-relay rtp-nte digit-drop
codec g711ulaw
no vad
!
dial-peer voice 102 voip
destination-pattern 011T
session protocol sipv2
session target dns:siphost.net
session transport udp
dtmf-relay rtp-nte digit-drop
codec g711ulaw
no vad
!
dial-peer voice 103 voip
destination-pattern [2-9]..[2-9]......
session protocol sipv2
session target dns:siphost.net
session transport udp
dtmf-relay rtp-nte digit-drop
codec g711ulaw
no vad
The problem is that 7 digit calls never match and send to the SIP host. The 'debug voip dialpeer all' just indicates that its waiting for more digits. I thought that I could use the 'T' in the dial-peer 101 to allow overlaping dial patterns and let the inter-digit timeout decide, but it never places the call. I have been web searching and found some indication that there is no inter-digit timeout for SIP gateways.
Has anyone else fought this before?
thanks,
