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SIP configuration with CUCM 10.2

Mark Verwey
Level 1
Level 1

I am in the process of setting our office with SIP trunks, moving from ISDN/PRI circuits to save money.  The carrier we are using has initially provided us with a 10MB VPN connection.  I am using a Cisco 2951 router as my gateway and have configured this as a gateway on the CUCM and also configured a SIP trunk pointing back to the interface GB0/0 which connects to my network where the CUCM.  I created a DP, Partition, and CSS on my CUCM for just the SIP trunk so that only phones configured to use that DP would be able to hit the SIP trunk.  The carrier requested I setup a /30 subnet which I did.  I connected the GB0/1 interface to the carrier's equipment, the interface came up and I can ping there side of the interface.  My confussion is with the remainder of the configuration.  I thought I had this correct in my mind as I was programming the dial peers for both inbound and outbound calling.  My thought originally was that I needed at a minimum 4 dial peers on the gateway.  The first would be inbound from the PSTN (carrier) to the gateway, then outbound to the CUCM.  The third would be outbound from the CUCM, and fourth would be from gateway to the PSTN.  I thought that even though the carrier hasnt turned up the SIP on their side I would be able to hit the gateway and dial peers, which I can.  Heres where I ran into problems, the dial peers that I thought would be for the outbound from the CUCM to the gateway isnt the dial peer that shows its using in the debug traces.  I expected the call to be using dial peer 201 (from cucm to gw) however the trace shows it hits dial peer 101.  Also, in a pre turnup meeting today they gave me a couple of VPN ip addresses that they said I would need to configure my side, not sure what or where to put them.  Can anyone fill in the blanks for me?  Here is part of the gateway config that I need help with..

 

voice translation-rule 100
rule 1 /904783\(.*\)/ /\1/
rule 2 /904370\(.*\)/ /\1/
!
voice translation-rule 200
rule 1 /^9\(.*\)$/ /\1/
!
voice translation-profile INBOUNDCUCM
translate called 100
!
voice translation-profile OUTBOUNDPSTN
translate called 200
!
license udi pid CISCO2951/K9 sn FTX1750AL5Q
hw-module pvdm 0/0
!
hw-module pvdm 0/1
!
redundancy
!
ip tcp synwait-time 13
ip ssh time-out 60
ip ssh authentication-retries 2
ip ssh version 1
!
interface Embedded-Service-Engine0/0
no ip address
shutdown
!
interface GigabitEthernet0/0
ip address 10.211.110.18 255.255.255.224
duplex auto
speed auto
no mop enabled
!
interface GigabitEthernet0/1
ip address 192.168.3.138 255.255.255.252
duplex auto
speed auto
!
interface GigabitEthernet0/2
no ip address
shutdown
duplex auto
speed auto
!
ip forward-protocol nd
!
no ip http server
no ip http secure-server
!
ip route 0.0.0.0 0.0.0.0 10.211.110.1
!
nls resp-timeout 1
cpd cr-id 1
!
control-plane
!
voice-port 0/1/0
!
voice-port 0/1/1
!
mgcp behavior rsip-range tgcp-only
mgcp behavior comedia-role none
mgcp behavior comedia-check-media-src disable
mgcp behavior comedia-sdp-force disable
!
mgcp profile default
!
sccp local GigabitEthernet0/0
sccp ccm 10.211.110.10 identifier 1 priority 1 version 7.0
sccp ccm 10.211.110.11 identifier 2 priority 2 version 7.0
sccp ccm 10.219.112.150 identifier 3 priority 3 version 7.0
sccp
!

dial-peer voice 1 voip
description inbound from pstn to gw
translation-profile incoming INBOUNDCUCM
session protocol sipv2
incoming called-number .%
voice-class sip dtmf-relay force rtp-nte
codec g711ulaw
no vad
!
dial-peer voice 101 voip
description outbound from gw to cucm
destination-pattern [19]...
session protocol sipv2
session target ipv4:10.211.110.10
voice-class sip bind control source-interface GigabitEthernet0/0
voice-class sip bind media source-interface GigabitEthernet0/0
codec g711ulaw
!
dial-peer voice 201 voip

 description inbound from cucm to gw
session protocol sipv2
incoming called-number 9...........
!
dial-peer voice 301 voip
description outbound from gw to pstn
translation-profile outgoing OUTBOUNDPSTN

voice-class sip bind control source-interface GigabitEthernet0/0
voice-class sip bind media source-interface GigabitEthernet0/0

destination-pattern [91]...
session target ipv4:192.168.3.137
!
sip-ua
!
gatekeeper
shutdown

1 Reply 1

Chris Deren
Hall of Fame
Hall of Fame

What number are you dialing?

One issue with your destination-patterns is that you match the same first digit by both outbound dial peer [91] and [19] as anything in [] represents a single digit, which makes your dial-peers overlapping.