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CallManager 4.x to SIP ITSP

darin.pesnell
Level 1
Level 1

Hello,

I am trying to find out if a CallManager 4.x system can communicate to an IP Telephony Service Provider over SIP to their Sessions Border Controller. From what it looks like, I will need an IP2IP gateway to talk H323 to the CCM and SIP to the ITSP. Has anyone successfully done this before?

Any help or experiences would be greatly appreciated.

6 Replies 6

rhoorn
Level 1
Level 1

We have tested this in our lab, and this was working well. An Cisco 2811 with ver. 12.4 IP2IP was used for this test and H323 to SIP, H323 to H323 and SIP to H323 was working well.

Config Cisco router :

Building configuration...

Current configuration : 2983 bytes

!

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 password cisco

!

no aaa new-model

!

!

ip cef

no ip dhcp use vrf connected

no ip dhcp conflict logging

ip dhcp excluded-address 10.193.25.1 10.193.25.65

ip dhcp excluded-address 172.16.1.1 172.16.1.9

ip dhcp excluded-address 10.193.25.70 10.193.25.80

!

ip dhcp pool 10.193.25.0

network 10.193.25.0 255.255.255.0

option 150 ip 10.193.25.113

default-router 10.193.25.111

lease infinite

!

!

!

multilink bundle-name authenticated

!

!

voice-card 0

no dspfarm

dsp services dspfarm

!

!

!

voice service voip

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

h323

call start interwork

sip

!

!

!

!

interface Loopback0

ip address 10.0.0.1 255.255.255.255

!

interface FastEthernet0/0

ip address 10.193.25.111 255.255.255.0

duplex auto

speed auto

h323-gateway voip interface

h323-gateway voip bind srcaddr 10.193.25.111

!

interface FastEthernet0/1

no ip address

shutdown

duplex auto

speed auto

!

interface Content-Engine1/0

no ip address

shutdown

!

ip route 0.0.0.0 0.0.0.0 10.193.25.1

!

!

no ip http server

!

!

!

!

control-plane

!

!

!

!

!

!

!

dial-peer voice 10 voip

description VoIP to live callmanager

destination-pattern 3...

progress_ind connect enable 8

session target ipv4:10.193.1.5

dtmf-relay h245-alphanumeric

codec g711alaw

!

dial-peer voice 20 voip

description VoIP to Test Callmanager

tone ringback alert-no-PI

destination-pattern 2...

progress_ind setup enable 3

progress_ind progress enable 8

progress_ind connect enable 8

session target ipv4:10.193.25.113

dtmf-relay h245-alphanumeric

codec g711alaw

!

dial-peer voice 30 voip

description to VoIP/AA at Test Callmanager

destination-pattern 500.

session target ipv4:10.193.25.113

dtmf-relay h245-alphanumeric

codec g711alaw

!

dial-peer voice 1 voip

description to H323 External GW

destination-pattern 0T

session target ipv4:10.193.1.4

dtmf-relay h245-alphanumeric

codec g711alaw

!

dial-peer voice 200 voip

description to SIP Soft IP-Phone

destination-pattern 1999

session protocol sipv2

session target ipv4:10.193.10.9

dtmf-relay rtp-nte

codec g711alaw

!

dial-peer voice 100 voip

tone ringback alert-no-PI

description 3th party hardware SIP IPPhone

destination-pattern 1...

session protocol sipv2

session target ipv4:10.193.25.200:5060

dtmf-relay rtp-nte h245-alphanumeric

codec g711alaw

no vad

!

!

sip-ua

retry options 0

!

!

gatekeeper

shutdown

!

!

line con 0

line aux 0

line 66

no activation-character

no exec

transport preferred none

transport input all

transport output pad telnet rlogin lapb-ta mop udptn v120

line vty 0 4

password cisco

login

!

scheduler allocate 20000 1000

!

end

Router#

Thanks for the reply! Judging by your descriptions, you tested with SIP softphones. Did you test from CallManager to and ITSP's Session Border controller as well?

Thanks again,

Darin

Hi Darin,

I'm trying to setup the same thing and was wondering whether you managed to get this resolved in the end? I can successfully make outbound calls from CCM4.2(3) but I'm having some problems with the inbound. Any help appreciated.

Johan

Hi J-R,

could you post the output of "debug ccsip message" with "term mon" taken on the router ?

This will help understanding what is going on.

Hi There, Please find attached my config and debug output as requested. Please note that this is when trying an inbound call.

Hi,

you DP 2 should be voip, not pots. In fact you don't need need DP 2 at all.

I seen only 5ms between invite and busy. Not sure if the h.323 is being activated. Check it out with "debug cch323 all". May be CCM is rejecting the call for some reason, unfortunately I'm not good at CM enough to tell you why.

Hope this helps, please rate useful posts!

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