What software version do u've, just 12.3T, support IP2IP feature. I suggest you to use * instead
-----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Tuesday, November 30, 2004 10:53 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] cisco dial-peer voip You have 3 dial-peers (40,50,60) all with the same destination-pattern .+ (that means all calls) Think it first tries dial-peer 40 because it has preference 0... And then peers 50 (or) 60 (both preference 5) ... It uses the second preference because the peer 40 just doesn't work.... And that sounds logically because you have "session transport tcp" ... And asterisk doesn't support that... Use "session transport udp" Regards, Niels -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jan Baggen Sent: Tuesday, November 30, 2004 2:36 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] cisco dial-peer voip 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 _______________________________________________ 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
