Hi i test a new equipment on my backbone: a Cisco AS5300 with voice dsp ressource connected at a E1 Voice Link.
I want that all call incoming on the cisco 5300 are sent to Asterisk and all Asterisk outgoing call are sent to Cisco AS5300. Actually, i configure the AS5300: isdn switch-type primary-net5 ! voice service voip sip ! voice class codec 400 codec preference 1 g711alaw codec preference 2 g729r8 codec preference 3 g723r63 codec preference 4 g711ulaw ! voice class codec 500 codec preference 1 g729r8 codec preference 2 g723r63 ! controller E1 0 framing NO-CRC4 pri-group timeslots 1-31 description E1 Beta-Test interface Serial0 no ip address shutdown clock rate 2015232 no fair-queue ! interface Serial1 no ip address shutdown clock rate 2015232 no fair-queue ! interface Serial2 no ip address shutdown clock rate 2015232 no fair-queue ! interface Serial3 no ip address shutdown clock rate 2015232 no fair-queue ! interface Serial0:15 no ip address encapsulation ppp isdn switch-type primary-net5 no cdp enable voice-port 0:D ! ! ! dial-peer voice 10 voip destination-pattern .T session protocol sipv2 session target ipv4:IP_OF_ASTERISK:5060 session transport udp dtmf-relay rtp-nte codec g711alaw no vad ! dial-peer voice 42 pots destination-pattern .T direct-inward-dial port 0:D ! sip-ua retry invite 3 retry response 3 retry bye 3 retry cancel 3 timers trying 1000 sip-server ipv4:IP_OF_ASTERISK ! Actually, a Tcpdump on my Asterisk server don't see any trafic between asterisk and cisco and when i call a phone number that arrives on the E1, it's "busy" anyone have a idea ? bye jerome _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
