On Thu, Oct 15, 2009 at 6:27 AM, Phibee Network Operation Center <n...@phibee.net> wrote: > 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
What does "destination-pattern .T" mean? I'm not familiar with what ".T" would match. I would suggest using a more specific pattern that you expect to be coming down the line. > 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" Doesn't see any traffic when? When the asterisk tries to call the Cisco? That would suggest you have a sip.conf misconfiguration on asterisk. No traffic when the Cisco tries to call the asterisk? That could be for a number of reasons. I would suggest your destination-pattern could be bad, since I don't know what that syntax means. If an E1 works like a PRI T1, when you dial in, a DNIS is getting pushed to the Cisco, and that's what you should match your destination-pattern on. Regardless, SOMETHING is getting pushed down the wire when an E1 call comes in, and you're getting a busy because Cisco has no matching dial-peers. Finally, it's rather embarrassing that you're asking this from a 'network operations center' email address. How about using a personal email address with your real name? _______________________________________________ -- 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