Okay, I have made some progress getting calls in and out of asterisk with the mc3810. I think the problem lies in how the switch is sending me the did digits. I am receiving "*6125551212**4418*" from the switch (the 612... being the caller id, 4418 the did#), the cisco matches *612555 with the pots dial peer I have setup for outgoing calls and tries to make an outgoing call to *612555. Does anyone know how I can build a dial peer with a destination pattern that will strip off all of the extra stuff and just process the 4 digit did?
Thanks Jason ----- Original Message ----- From: "Jason Brockman" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Monday, November 29, 2004 5:47 PM Subject: [Asterisk-Users] Cisco gateway help needed > HI, > > I have been pulling my hair out trying to get a Cisco MC3810 to interface my > Asterisk box with a T1. > I am able to make outgoing calls but incoing calls never reach my Asterisk > box. The cisco give a fast busy when I try to call one of the DID's. When > playing around with the dial-peers I can get the cisco to pick up the call, > but then it forwards the call back to the ANI that is dialing. I know the > T1 is good because I hooked it up to a Nortel KSU and the DID's work fine. > > I am receiving 4 digits and the T1 only has 4 ds0's. > > I have attached a sho run, any help would be appreciated. > > TIA, > Jason > > service timestamps debug uptime > service timestamps log uptime > service password-encryption > ! > hostname gw1 > ! > boot-start-marker > boot system tftp mc3810-a2isv5-mz.123-10a.bin 192.168.5.104 > boot-end-marker > > network-clock base-rate 56k > no aaa new-model > ip subnet-zero > ! > voice class codec 10 > codec preference 1 g711ulaw > codec preference 2 g711alaw > codec preference 4 g729r8 > codec preference 6 g729ar8 > ! > ! > no voice confirmation-tone > ! > controller T1 0 > mode cas > framing esf > linecode b8zs > ds0-group 1 timeslots 1-4 type e&m-wink-start > fdl both > ! > controller T1 1 > mode cas > framing esf > linecode b8zs > ds0-group 1 timeslots 1-4 type e&m-wink-start > ! > ! > ! > interface Tunnel1 > no ip address > ! > interface Ethernet0 > ip address xx.xx.xx.xx 255.255.255.248 > ! > interface Serial0 > no ip address > shutdown > ! > interface Serial1 > no ip address > shutdown > ! > interface FR-ATM20 > no ip address > shutdown > ! > ip default-gateway xx.xx.xx.xx > ip classless > ip route 0.0.0.0 0.0.0.0 xx.xx.xx.xx > no ip http server > ! > ! > ! > snmp-server community xxxxxx RO > ! > voice-port 0:1 > ! > voice-port 1:1 > ! > ! > ! > dial-peer voice 1 voip > destination-pattern T > progress_ind setup enable 3 > progress_ind progress enable 8 > voice-class codec 10 > session protocol sipv2 > session target ipv4:xx.xx.xx.xx > session transport udp > dtmf-relay rtp-nte > no vad > ! > dial-peer voice 110 pots > incoming called-number .... > direct-inward-dial > ! > dial-peer voice 100 pots > destination-pattern ....... > port 0:1 > ! > sip-ua > retry invite 3 > retry cancel 2 > sip-server ipv4:xx.xx.xx.xx:5060 > ! > ! > line con 0 > transport preferred all > transport output all > line aux 0 > transport preferred all > transport output all > line 2 3 > transport preferred all > transport output all > line vty 0 4 > login local > transport preferred all > transport input all > transport output all > ! > ntp server xx.xx.xx.xx > end > > gw1# > > _______________________________________________ > 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
