Emilio Panighetti
2004-Oct-12 13:14 UTC
[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.
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/${EXTEN:1}@8002)
exten => _91NXXNXXXXXX,2,Dial(SIP/${EXTEN:1}@8003)
________________________________
From: asterisk-users-bounces@lists.digium.com 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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