Here is what works for me. It is currently working and in service on an MC3810. plar is needed so incoming calls ring an extension in asterisk. extension 102 sends call to my IVR root. Please remember to configure default gateway. This especially important if you have nat specified in asterisk. This is for an MC3810, but you should be able to get enough out of it to make your AS5300 work. Jojo In IOS: version 12.3 service timestamps debug uptime service timestamps log uptime service password-encryption ! hostname MC3810-1 ! boot-start-marker boot system flash:mc3810-a2isv5-mz.123-10.bin boot-end-marker ! enable password 7 xxxxxxxxxxx ! network-clock base-rate 56k no aaa new-model ip subnet-zero ! no ip domain lookup ! 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 shutdown framing sf linecode ami ! interface Ethernet0 ip address 192.168.1.7 255.255.255.0 ip route-cache same-interface ! interface Serial0 no ip address shutdown ! interface Serial1 no ip address shutdown ! interface FR-ATM20 no ip address shutdown ! ip default-gateway 192.168.1.1 ip classless ip route 0.0.0.0 0.0.0.0 192.168.1.1 no ip http server ! ! ! ! voice-port 1/2 connection plar 102 station-id name FXO2 station-id number 8002 ! voice-port 1/3 connection plar 102 station-id name FXO3 station-id number 8003 ! dial-peer cor custom ! dial-peer voice 1 pots destination-pattern ........... port 1/3 ! dial-peer voice 2 pots destination-pattern ........... port 1/2 ! dial-peer voice 10 voip destination-pattern 102 voice-class codec 10 session protocol sipv2 session target sip-server ! sip-ua retry invite 3 retry cancel 2 sip-server ipv4:192.168.1.5:5060 ! ! line con 0 exec-timeout 0 0 logging synchronous 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 password 7 xxxxxxxxx login transport preferred all transport input all transport output all ! end In sip.conf: [8002] type=friend username=8002 host=192.168.1.7 <- IP address of Cisco canreinvite=no qualify=yes nat=no dtmfmode=inband [8003] type=friend username=8003 host=192.168.1.7 <- IP address of Cisco canreinvite=no qualify=yes nat=no dtmfmode=inband In extensions.conf [default] include => 8002
exten => 102,1,Goto(locals,s,1) <-sends to root of my IVR
[8002]
exten => _91NXXNXXXXXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten => _91NXXNXXXXXX,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
________________________________
From: [EMAIL PROTECTED] on behalf of Emilio Panighetti
Sent: Tue 10/12/2004 1:14 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] SIP Connection to a Cisco AS5xxx gateway
Hello,
Does anybody have any experience connecting Asterisk to a Cisco gateway?
I'm trying to terminate calls into this gateway, and then route
incoming DID numbers from the gateway into Asterisk.
So far, Asterisk sends the call to the gateway, and it connects the
call, but there's no audio. I'm using the Cisco gateway with IOS
12.3.10T, connecting as SIP, no registration, and as clients I tried
different SIP Phones including Cisco ATA (which connects to the gateway
just fine without using asterisk), Gandstream ATA and the console. They
all communicate to each other through SIP, but not to the Cisco
gateway. I'm using g.711uLaw as the codec to talk to the gateway.
Thanks,
E.
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