12-06-2006 09:46 AM - edited 03-13-2019 03:50 PM
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.
12-08-2006 06:33 AM
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#
12-08-2006 09:45 AM
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
03-24-2007 04:22 AM
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
03-24-2007 04:32 AM
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.
03-24-2007 08:42 AM
03-24-2007 11:21 AM
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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