Hello, i know thats is * mailing list but maybe here are cisco guru's which can help.
My network schema is: Softphones <-SIP-> Asterisk <-SIP->\ /<-H323-> WORLD2 > Cisco AS5350 <- E1 -> WORLD (Phones <-> Traditional PBX <-E1->--/) - will be developed soon. Communications beetween WORLD and softphones works well but i have an H323 link to other site and i want to allow calling from softthones (in future from Phones too) calling to the WORLD2. I tried to add dialpeers but this doesnt work - all calls are routed via E1 to WORLD. This is part of my config: ! GK Config: interface FastEthernet0/0 ip address 192.168.X.X 255.255.255.0 duplex auto speed auto h323-gateway voip interface h323-gateway voip id TGK1 ipaddr 194.X.X.X 1719 h323-gateway voip h323-id MYID ! gateway ! For World -> Softphones communication dial-peer voice 14 pots incoming called-number 2323. direct-inward-dial ! dial-peer voice 15 voip destination-pattern 2323. session protocol sipv2 session target ipv4:192.168.X.X codec g711alaw ! For outgoing Softphones - World dial-peer voice 1000 pots application session destination-pattern .T direct-inward-dial port 3/1:D forward-digits all ! i tried to add ! dial-peer voice 999 voip application session destination-pattern 0. target session ras ! but all calls are still routed via dial peer 1000 - why ? I want to pass all calls thru cisco becouse i need one point for billing for asterisk and PBX calls and in future i need to make calls from PBX to the WORLD2 destinantion. PLEASE HELP! Thanks, Adam _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users