The call is inbound on the pots dial-peer, so you should use incoming called-number, as opposed to destination-pattern.
dial-peer voice 1 pots incoming-called number [0-9]T no digit-strip direct-inward-dial port 3/0:D I'm not familiar with the [0-9] syntax, but if it works, ok. I usually use "." Also, you can specify the sip destination directly in the dial-peer, which makes using sip with the cisco's more flexible unless you're using a separate sip proxy. session protocol sipv2 session target ipv4:5.5.5.5 -g On Wed, 2004-07-14 at 07:27, [EMAIL PROTECTED] wrote: > > Hi. > > > If I use a Cisco as a PSTN termination GW and need to route all incoming > isdn calls to my asterisk and all outgoing calls from asterisk via the > cisco out to pstn, how do I do that ? > > > in the cisco I have this: > > dial-peer voice 1 pots > destination-pattern [0-9]T > no digit-strip > direct-inward-dial > port 3/0:D > ! > dial-peer voice 50 voip > destination-pattern [0-9] > voice-class codec 1 > session protocol sipv2 > session target sip-server > no vad > dtmf-relay rtp-nte > ! > > > ------- > > But theese to dialplans seem to interrupt each other. > > When an incoming call from PSTN goes through this the pattern can be > matched by the first, and then be routed ot on the PSTN again, creating > a loop. > > How do I do this in the smartest and easiest way ? > > /Mike > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
