09-13-2007 06:19 PM - edited 03-14-2019 11:32 PM
I have a CCM in Site A and I have a CME in Site B. I have a VPN link connecting them.
I need 4 digit dialing between them.
How do I integrate them. I tried creating an intercluster trunk but it did not work.
09-13-2007 07:43 PM
Hi Joel,
hope this will help you:
/majdi
09-13-2007 08:00 PM
I literally JUST did this today. Here are the directions and it works GREAT!!!
If you have questions or need help on it please let me know.
Jason
configuration needs to be apply on both ends CME and CCM.
on the CME portion, you need to have the following
dial-peer voice XX voip
destination-pattern XXXX
sesstion target ipv4:
on the call manager's end, you need to add the following
Create an H323 gateway and use the ip address of CME.
Make sure the CSS of this H323 gateway contain the IP phones Partition. This triped me up.
Look down in the config of the GW a little for "Call Routing Info" Inbound make sure this matches your phones
so CCM knows how to route the call to the ip phone.
and on call manager, you have to create a route pattern pointing to this CME router with the 4 digit extention,
and the ip phone have to have the access to this route pattern.
If you are going to enable fax fron site to site over the 4 digit dila you need this also:
voice service voip
modem passthrough nse codec g711ulaw redundancy maximum-session 5
dial-peer voice XX voip
modem passthrough nse codec g711ulaw redundancy
01-03-2008 03:48 PM
Jason, I'm doing exactly this and I can't go on to work this out, I have a question for you, when you create H323 gateway on CCM by default uses port 1720, and in CME by default is 1719, which should I use?? Does it has to be the same port on both CME and CCM??
Regards,
Juan Carlos Arias
09-14-2007 05:27 AM
Hello,
I just finished CCM and CME intergation. I used SIP.
Here is my config:
SIP trunk on CCME router:
dial-peer voice 20 voip
destination-pattern XXXX
session protocol sipv2
session target ipv4:10.181.0.200
session transport tcp
dtmf-relay rtp-nte
codec g711ulaw
no vad
The Trunk on CCM is simple:
- Call Classification - OnNet;
- MTP - checked;
- Destination Address - IP on CCME;
- ports - 5060, 5060, TCP, 711ulaw;
- significant digits - 4;
Let me know if need more help.
10-05-2007 12:00 AM
Hello,
I've configured my SIP connection between CCM 4.2 and CCME 3.3 like well-descripted in precedent posts. It's work fine in CCM->CCME direction (call from CCM IP Phone to CCME IP Phone is established and voice quality is good). In the reverse condition (making a call from CCME to CCM) the call is not established. I've already checked the CSS on trunk and above there is the CCME dial-peer configuration:
dial-peer voice 2001 voip
tone ringback alert-no-PI
destination-pattern 2....
session protocol sipv2
session target ipv4:
session transport tcp
dtmf-relay rtp-nte
codec g711ulaw
no vad
Any hint?
Thanks in advance
Claudio
11-15-2007 02:48 PM
what version of CCM are you using? I don't believe SIP will work with v 3.x or 4.x - you will have to use h.323 in this case.
edit: just notice you mentioned v 4.2, try h.323
12-07-2007 01:14 PM
Hello,
I am also trying to do the exact same thing, question...since I am going over the clients VPN what should the firewall portion look like? I know they have the fixup protocol enabled for voice?
01-03-2008 01:39 PM
you probably need the following command (PIX):
fixup protocol h323 ras 1718-1719
although for VPN I dont think it is required, since the traffic is not NAT'd over VPN
Find answers to your questions by entering keywords or phrases in the Search bar above. New here? Use these resources to familiarize yourself with the community: