Called & Calling Number Transformation with MGCP

[Nightwolf_82]

Hi guys,

Below is my diagram:

3XXX [CUCM] --- [MGCP GTW] 61 2 33333XXX === WAN/PSTN === 1 212 5555XXX [CME] 5XXX

I have this translation rule on my CME to translate called and calling parties for incoming and outgoing calls.

voice translation-rule 1
 rule 1 /^1212555\(5...\)$/ /\1/
!
voice translation-rule 2
 rule 1 /^.......$/ /9&/
 rule 2 /^[2-9]..[2-9]......$/ /91&/
 rule 3 /^1[2-9]..[2-9]......$/ /9&/
 rule 4 /.*/ /9011&/
!
voice translation-rule 3
 rule 1 /^\(3...\)$/ /6123333\1/
!
voice translation-rule 4
 rule 1 /^\(5...\)$/ /1212555\1/
!
voice translation-profile PSTN-IN
 translate calling 2
 translate called 1
!
voice translation-profile PSTN-OUT
 translate calling 4
 translate called 3
!
voice-port 0/0/0:23
 translation-profile incoming PSTN-IN
 translation-profile outgoing PSTN-OUT

I need to configure the same on the other side where CUCM and MGCP gateway are set up. As MGCP gateway is dependent on CUCM settings then I need to configure called and calling transformation on CUCM.

Could you please explain in which section of CUCM settings I can do that?

[Aseem Anand]

Hi,

To fix the calling number you can use ext mask set on the phone under line settings and automatically each phone's ext mask will be send out to the gateway specified in the route pattern.

call routing >>. Route hunt >>> Route pattern >>> Use the checkbox "Use Calling Party's External Phone Number Mask" to take care of the calling number.

For called number you can use the prefix at translation pattern or on route pattern.

Aseem

(Please rate if useful)

[Nightwolf_82]

Hi Aseem,

Thank you for your reply. This (pls see screenshot) helped me. I am not sure if it's correct solution.

BTW, I got another question. On CME I use this config to switch from SIP trunk to PSTN in case WAN link is failed.

dial-peer voice 3000 voip
 description **OUGOING CALLS TO SYDNEY OVER WAN**
 destination-pattern 3...
 session protocol sipv2
 session target ipv4:192.168.10.5
!
dial-peer voice 3001 pots
 description **OUGOING CALLS TO SYDNEY OVER PSTN**
 translation-profile outgoing PSTN-OUT
 preference 1
 destination-pattern 3...
 port 0/0/0:23

How to configure the same on CUCM?

[Aseem Anand]

Hi,

Yes, the screenshot you attached would work as well. You can modify the calling party either on translation pattern or on route pattern/route list.

You can achieve the redundancy using two route groups. Basically create two route groups RG1 and RG2 and associate then with a route list in order RG1 and then RG2.

Create a SIP trunk on your call manager to CME and add it in route group 1 and you can add a Voice gateway (h323/MGCP) with the T1/E1 controller/FXO on CUCM and add it in route group 1.

So, call calls would first take RG1 and then RG2.

http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/8_6_1/ccmcfg/bccm-861-cm/b03rtgrp.html

Aseem

(Please rate if useful)

[Nightwolf_82]

1. Created two Route Groups, one for WAN with SIP Trunk and MGCP Gateway and one for PSTN with MGCP Gateway.

2. Created one Route List and added both WAN and PSTN Route Groups.

3. Created Route Pattern 5XXX and specified Route List.

In Route List I added 000111212555 as Prefix Digits (Outgoing Call) in Calling Party Transformations (0 - trunk code, 0011 - international code, 1 - USA code, 212 - New York code, 555 - first three digits of NY branch DID). However, redundancy still not working.

What has been done wrong or is there anything else that should be configured?

[Aseem Anand]

Assuming you have only two devices to call out, one SIP trunk for WAN and second an MGCP gateway with a PRI, you should create two RG with first RG containing only the trunk and second containing PSTN gateway.

As per the screenshot you shared, you have added both trunk and an MGCP gateway in the same RG whereas the RG for trunk should have only RG.

Do you have two MGCP gateways and a Trunk to call out?

Aseem

[Nightwolf_82]

Hi Aseem,

I just followed the video by Jeremy Cioara about understanding route lists and route groups where he explains how to configure redundancy in CUCM. In his scenario users who dial extension number should be switched to PSTN in case WAN link is down. So, he put both WAN and PSTN links in the list of devices of the same same WAN Route Group with WAN on top and PSTN right under the WAN.

I have only one MGCP gateway and only one SIP trunk.

Thank you.

[Aseem Anand]

Hi,

Can you simply create two RG's, one for SIP trunk and other one for MGCP gateway to test redundancy? Also make sure the following are set to false in CUCM service parameters:

System >>> Service parameters

Stop Routing on Out of Bandwidth Flag:	This parameter determines routing behavior for calls through trunks when not enough bandwidth exists. Valid values specify True or False. When the parameter is set to True and a call that is being routed to a remote Cisco cluster through a route list is released by a remote Cisco CallManager because of the insufficient bandwidth of a destination device at the remote cluster, a local Cisco CallManager will stop routing the call to the next device in the route list. When the parameter is set to False, the local Cisco CallManager will route the call to the next device. An associated Location specifies the bandwidth of the device.
	This is a required field.
	Default:  False
Stop Routing on Unallocated Number Flag:	This parameter determines routing behavior for trunk calls to an unallocated number. An unallocated number represents a dialed directory number that does not exist in a Cisco cluster. Valid values specify True or False. When the parameter is set to True and a call that is being routed to a remote Cisco cluster through a route list is released by a remote Cisco CallManager because of the unallocated number, a local Cisco CallManager will stop routing the call to a next device in the route list. When the parameter is set to False, the local Cisco CallManager will route the call to the next device.
	This is a required field.
	Default:  True
Stop Routing on User Busy Flag:	This parameter determines routing behavior for trunk calls to a busy phone at a remote Cisco cluster. When the parameter is set to True and a call that is being routed to a remote Cisco cluster through a route list is released by a remote Cisco CallManager because a remote user (phone) is busy, a local Cisco CallManager will stop routing the call to the next device in the route list. When the parameter is set to False, the local Cisco CallManager will route the call to the next device.
	This is a required field.
	Default:  True

Aseem

[Aseem Anand]

Hi,

Please accept it as solution if i have answered your query.

Thanks

Aseem