[Asterisk-Users] Cisco gateway help needed

[Jason Brockman]

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#