08-28-2015 04:20 AM - edited 03-17-2019 04:07 AM
Hay,
I am new to cisco IP telephony.I have just configured a SIP trunk b/w Asterisk and cisco2800 router.Now i want to create another trunk with Avaya with the first one with Asterisk also working i am confused how to do this.
below is my config on router
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname Router
!
boot-start-marker
boot-end-marker
!
enable secret 5 $1$ml.W$gFgeC4R3cZwYLfDULvXx81
enable password xxxx
!
no aaa new-model
!
resource policy
!
no network-clock-participate wic 1
ip subnet-zero
!
!
ip cef
!
!
!
voice-card 0
no dspfarm
!
!
voice rtp send-recv
!
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
redirect ip2ip
signaling forward unconditional
sip
bind control source-interface GigabitEthernet0/0
bind media source-interface GigabitEthernet0/0
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
controller E1 0/1/0
!
controller E1 0/1/1
!
!
!
interface GigabitEthernet0/0
ip address 192.168.200.41 255.255.255.0
duplex auto
speed auto
!
interface GigabitEthernet0/1
no ip address
shutdown
duplex auto
speed auto
!
ip classless
!
!
no ip http server
no ip http secure-server
!
!
!
!
control-plane
!
!
!
voice-port 0/0/0
cptone PK
connection plar opx 7500
caller-id enable
!
voice-port 0/0/1
!
!
no mgcp package-capability res-package
no mgcp package-capability fxr-package
no mgcp timer receive-rtcp
no mgcp explicit hookstate
!
!
!
dial-peer voice 7500 voip
destination-pattern 7500
session protocol sipv2
session target ipv4:192.168.200.170 //ip of astersik server
codec g711alaw
no vad
!
dial-peer voice 100 pots
destination-pattern 0T
port 0/0/0
forward-digits all
!
sip-ua
retry invite 3
retry response 3
retry bye 3
retry cancel 3
timers trying 100
sip-server ipv4:192.168.200.170 //////ip of asterisk server
!
!
!
telephony-service
max-conferences 8 gain -6
!
!
line con 0
line aux 0
line vty 0 4
password xxxx
login
line vty 5 15
password xxxxx
login
!
scheduler allocate 20000 1000
!
end
Router#
Solved! Go to Solution.
08-28-2015 06:11 AM
Simply define the destination on a dial-peer as you can only have one sip-server, i.e.
dial-peer voice 7501 voip
destination-pattern XXXX
session protocol sipv2
session target ipv4:x.x.x.x //ip of Avaya
codec g711alaw
no vad
08-31-2015 05:11 AM
That is no different from what I already provided. The sip-server is nothing more than central place for defining the destination and then pointing to from dial-peers, having a specific destination on the dial-peer overwrites it. So, if you have multiple sip destinations you just define them on appropriate dial peers.
08-28-2015 06:11 AM
Simply define the destination on a dial-peer as you can only have one sip-server, i.e.
dial-peer voice 7501 voip
destination-pattern XXXX
session protocol sipv2
session target ipv4:x.x.x.x //ip of Avaya
codec g711alaw
no vad
08-31-2015 12:50 AM
Thanks Chris,
Bundles of thanks for prompt support.
What if i have both servers on local network/interface .Actually i have two PABX Avaya and Asterisk on local network 192.168.200.0 .I want to connect both two cisco 2800 via sip so that both PABX can be reached via each other.
Can you help configure i dont know how to add two sip servers
08-31-2015 05:11 AM
That is no different from what I already provided. The sip-server is nothing more than central place for defining the destination and then pointing to from dial-peers, having a specific destination on the dial-peer overwrites it. So, if you have multiple sip destinations you just define them on appropriate dial peers.
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