Backup link with ISDN PRI - Page 2

sagar.sarker · ‎01-29-2011

Hi,

We have an ISDN PRI E1 line as the primary link. Cisco 2821 router is being used as voice gateway and MCS 7825 as Call Manager Server. Now we want to use an IP trunk line as secondary link. Can anyone give me some idea on the configuration regarding this?

Best regards,

Sagar

sagar.sarker · ‎02-10-2011

Hi All,

Thanks to all for your valuable feedback.Can you please make me clear on below issues-

1) I didn't have any route group configuired in my system. One PRI line was connected to one gateay and the other PRI connected to the other system. And they were working fine without any route group or route list defined. If I press 7 the call goes through one PRI line and if 8 is pressed, the call goes through the other line.

2) @Senthil, the process you mentioned is very straightforward. I will definitely try it. But can you confirm is there any function of route group in your scenerio?

3) @John, I have two voice gateway and two SIP trunk configured in my system. One SIP trunk is for unified presence and the other one created lately as per your suggestion (using remote SIP server IP address as the destination address). But while creating route group, I don't see any gateway device in the list of available devices (screen shot attached). Please help me to configure the route group with the PRI connection.

I will update you on this on next Sunday.

Regards,

Sagar

Senthil Kumar Sankar · ‎02-10-2011

Hi

Yes In my design, you will be having 2 Routepatterns as below

   Route-Patterns :
     7XXXXXXX
     8XXXXXXX
   Route-List :
     MGW1-RL
     MGW2-RL
   Route-Group :
     PRI-TERM-RG----->Which contains the MGW1 as a Route-Member which has a PRI to PSTN
     SIP-TERM-RG----->Which contains the MGW2 as a Route-Member which has SIP trunk to ITSP

The structure would be like this

7XXXXXXX

-------->MGW1-RL

---------->PRI-TERM-RG

8XXXXXXX

---------->MGW2-RL

------------>SIP-TERM-RG

But, this can be used if you have the SIP Trunk terminating to the Voice Gateway (MGW2).

But in the screenshot you have attached, are these devices are SIP-Trunks which you added in CUCM ???

sagar.sarker · ‎02-12-2011

Hello Senthil,

Thanks once again for your suggestion.

1) As i said, i don't find gateway devices in my avaiable device list, only the SIP trunk devices are there.Please find attached the screenshots for my gateway devices and let me know if there is any prolem in the configuration. Why i'm not getting MGWs in available devices list?

2) Quote "are these devices are SIP-Trunks which you added in CUCM ???"

Yes, these are SIP trunks added in CUCM, one for Unified Presence and the other for IPTSP SIP service provider.

3) What will be the destination address field in the SIP trunk used for IPTSP? Will it be the address of the real IP provided the service provider or the MGW2 address for the system?

Please help me to find the answers.

Regards,

Sagar

dijohn · ‎02-13-2011

Been away for a while.!

Do you have Route Patterns directly pointing to those Gateways. If yes, thats why some of your gateways don't list under Route Groups.!

Remove them...Add to RG and try again.

Senthil Kumar Sankar · ‎02-13-2011

Hi

1) What will be the destination address field in the SIP trunk used for IPTSP? Will it be the address of the real IP provided the service provider or the MGW2 address for the system?

Yes you should be having the IP address of ITSP, If you are providing the CUCM IP as a signallling IP to the ITSP.

CUCM IP and ITSP IP will talk SIP Signalling for Sessions,

2) As dijohn advised.. you might be having Routepatterns's pointing to the Gateway IP directly.. Thats why you are unable to view that in Route group.

sagar.sarker · ‎02-14-2011

Hi Senthil & John,

Thanks for your co-operation regarding this. I really appreciate your assitance.

I am not so clear if I have been able to let you understand about my requirement (attached is the network diagram, if it helps) and the steps you are refering to solve it. I need more of your help in this regard.

1. As of now if I press 7, calls are outgoing through MGW1 without any route group or route list configured. If I pressed 8 earlier, it went through MGW2. Now I want to pass call the through MGW2. the provider gave us a public IP address as the gateway address for those calls. What shall I do to accomplish this.

2. I had configured CUBE as per Senthil, calls are coming through the GW. But I cant hear any IVR prompt (attached is the debug log).

3. Shall I still need to configure SIP even though CUBE is configured. If so what shall be the destination address. As per Senthil, this should be the public IP of ITSP, but the line is terminated to MGW2.CUBE configuration is as below-

dial-peer voice 1000 voip
destination-pattern .T
session target ipv4:172.20.1.101 --------------> CUCM publisher IP address
dtmf-relay h245-alphanumeric
codec g711ulaw
no vad
!
dial-peer voice 500 voip
destination-pattern 8T
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
!
dial-peer voice 2000 voip
service ivr
incoming called-number 09xxxxxxxx1
!
!
sip-ua
sip-server ipv4:x.x.x.x -------------->Public IP address for ITSP

4. Please state something about the route pattern, partition and calling search space required for this. As i said above, we had a route patter 8.! earlier through which calls went through MGW2. Now I want to use the same pattern for ITSP, but I cant hear any outgoing dial tone if 8 is pressed. What can be the problem here.

Regards,

Sagar

Senthil Kumar Sankar · ‎02-15-2011

Hi Sagar

By looking into your Network Diagram, it shows your SIP link is terminatiing to MGW2,

So here the SIP Signalling happens between MGW2 and ITSP-Sip-Server. I believe you dont want to add SIP-TRUNK in CUCM,

If this is the case you can follow the CUBE architecture i have given before.

Then your MGW2 will talk H323 Signalling to CUCM as given below.

Phones----8T---->CUCM-----H323----->MGW2-----SIP------>SIP Server.

Steps :

1) You have to add two H323 Gateways in CUCM, one for MGW1 and one for MGW2.

2) Create the route pattern 8.!, point to the device MGW2, ( if you directly point the deice in routepattern, no need of Routegroup & Routelist here)

3) Crea the route pattern 7.!, point to the device MGW1, ( if you directly point the deice in routepattern, no need of Routegroup & Routelist here)

When you create the RoutePattern 8.!, there will be a option like "Provide Outside Dial Tone"

Please check that box, if you want to hear the dialtone as soon as you press 8

Regards,

Senthil

sagar.sarker · ‎02-27-2011

Hi Senthil,

Sorry I was kind of busy with some other projects. I already have configured the call manager as per your suggestion. Calls are coming and going through voice gateway, but I cant hear anything. Also I am not getting any dial tone if 8 is pressed, though "Provide Outside Dial Tone" is checked.

Attached is the debug log while I was making incoming & outgoing calls through that voice gateway. Please help me on this issue.

Regards,

Sagar

sagar.sarker · ‎03-06-2011

Hi Senthil,

Have you got any chance to check the debug logs. Please share your observation regarding this.

Waiting for your response regarding this.

Regards,

Sagar

Senthil Kumar Sankar · ‎03-07-2011

Hi Sagar,

In the above scenario, The MGW2 device acts as CUBE device for your network.

Here we are doing performing Interworking between H323 and SIP, which required MTP.

I would suggest you to review the below section in the followinf URL

http://www.cisco.com/en/US/products/sw/voicesw/ps5640/products_configuration_example09186a00808ead0f.shtml

H.323 Trunk to the Cisco Unified Border Element

Regards

Senthil

sagar.sarker · ‎03-09-2011

Hi Senthil,

Thanks for your reply and sorry for bothering you again.

refer to your discussion, let me follow the steps required for this-

as per configuration segment-

1. I need to allow the communication between h323 gateway & SIP.

2. Configure incoming & outgoing dial-peers. Here I can see two different IP address. Can you please let me know which IP addresses should be there, one I can guess is CUCM server IP, what about the other one. Should it be the IPT service provider IP?

dial-peer voice 1 voip
session target ipv4:10.13.8.150
incoming called-number 8...
dtmf-relay h245-alphanumeric
codec g711ulaw
!
dial-peer voice 2 voip
destination-pattern 8...
session protocol sipv2
session target ipv4:10.13.8.16
dtmf-relay rtp-nte
codec g711ulaw

3. In SIP User Agent configuration, what will be the ip address-

sip-ua
registrar ipv4:10.1.1.10

4. in the segment you refered

H.323 Trunk to the Cisco Unified Border Element

there are other two dial-peers configured. Please help me to clear this also. I already have configured MGW2 in the CUCM server as per document.

BTW, did you find anything in the debug file I sent earlier?

Regards,

Sagar

Senthil Kumar Sankar · ‎03-16-2011

Hi Sagar

There will be two dial-peers, one for Incoming call from CUCM and another Outgoing to your ITSP-Provider.

Incoming dial-peer is H323 and Outgoing would be SIP.

I think this is your CUCM IP 10.13.8.150 so that is INCOMING DIAL-PEER, where the session target command is really not required.

In the OUTGOING DIAL-PEER you should have ITSP's IP as session Target with respective Dest-Pattern and the same ip should be there under "sip-ua"

INCOMING-DIAL-PEER :

dial-peer voice 1 voip
session target ipv4:10.13.8.150  /////Which is really not required/////
incoming called-number 8...
dtmf-relay h245-alphanumeric
codec g711ulaw

OUTGOING-DIAL-PEER
dial-peer voice 2 voip
destination-pattern 8...
session protocol sipv2
session target sip-server    ////sip-server command/////
dtmf-relay rtp-nte
codec g711ulaw

3. In SIP User Agent configuration, what will be the ip address-


sip-ua

sip-server ipv4:X.X.X.X:5060 ////ITSP-SIP-IP should be there/////

Regards
Senthil

sagar.sarker · ‎01-07-2012

Hi Senthil,

Sorry for bringing this old discussion in action once again. I need a suggestion from you regarding CUBE architecture you refered earlier. Do I need some kind of licnese to be activated in the system for CUBE configuration or it is FOC.

BR\\

Sagar

+8801714164719

Gajanan Pande · ‎01-08-2012

If your question is " Do you need any licenses for enabling CUBE functionality ", the answer is YES.

Pls rate the post if it helps.

GP.