Why not just have the Asterisk server act as a SIP/H323 gateway instead of the cisco router? You can then send incoming calls to registered Asterisk users via the cisco router and outgoing calls from Asterisk users to the PSTN via the cisco router. You can still use your same config below, but send the VoIP sessions through Asterisk and let it parse out where the calls need to go and send it to the cisco if you want to terminate traffic.
On Tuesday 30 November 2004 01:35 pm, Jan Baggen wrote: > I have 2610 XM with 1 Fastethernet and VIC2-2BRI. Dialin and dialout over > pots is ok. Also inbound pots calls get redirected to Asterisk y.y.y.y > So far so good. > > But I want to setup VOIP sessions with local carrier. I added dial-peer > 40 for this. Session target x.x.x.x But calls will always get routed to > the pstn peers 50 and 60. Peer x.x.x.x is never contacted or tried. > > My situation: > PSTN -> CISCO -> ASTERISK OK > ASTERISK -> CISCO -> PSTN OK > ASTERISK -> CISCO -> VOIP NOT OK (only needs outbound calls) > > > SIP01#sh dial-peer voice summary > dial-peer hunt 0 > TAG TYPE MIN OPER PREFIX DEST-PATTERN FER THRU SESS-TARGET > STAT PORT > 10 pots up up 0 down 1/0/0 > 20 pots up up 0 down 1/0/1 > 30 voip up up 2012345.. 0 syst > ipv4:y.y.y.y:5060 > 40 voip up up .+ 0 syst > ipv4:x.x.x.x:5060 > 50 pots up up .+ 5 up 1/0/0 > 60 pots up up .+ 5 up 1/0/1 > > > > dial-peer voice 10 pots > description INBOUND CALLS PSTN BRI0 > incoming called-number 2012345.. > no digit-strip > direct-inward-dial > port 1/0/0 > ! > dial-peer voice 20 pots > description INBOUND CALLS PSTN BRI1 > incoming called-number 2012345.. > no digit-strip > direct-inward-dial > port 1/0/1 > ! > dial-peer voice 30 voip > description INBOUND CALLS VOIP ASTERISK > destination-pattern 2051860.. > session protocol sipv2 > session target ipv4:y.y.y.y:5060 > session transport udp > dtmf-relay sip-notify > codec g711alaw > no vad > ! > dial-peer voice 40 voip > description OUTBOUND CALLS VOIP CARRIER > destination-pattern .+ > session protocol sipv2 > session target ipv4:x.x.x.x:5060 > session transport tcp > dtmf-relay sip-notify > codec g711alaw > no vad > ! > dial-peer voice 50 pots > tone ringback alert-no-PI > description OUTBOUND CALLS PSTN BRI0 > preference 5 > destination-pattern .+ > no digit-strip > port 1/0/0 > ! > dial-peer voice 60 pots > tone ringback alert-no-PI > description OUTBOUND CALLS PSTN BRI1 > preference 5 > destination-pattern .+ > no digit-strip > port 1/0/1 > > _______________________________________________ > 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 -- Brian Wilkins Software Engineer [EMAIL PROTECTED] Heritage Communications Corporation Melbourne, FL USA 32935 321.308.4000 x33 http://www.hcc.net _______________________________________________ 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
