Backup link with ISDN PRI

[sagar.sarker]

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

[dijohn]

You can try preference on the dialpeer. Pref 0 on POTS and Pref 1 on the voip dial-peer. pref 0 is better than 1.

dial-peer hunt ?
<0-7> Dial-peer hunting choices, listed in hunting order within each choice:
0 – Longest match in phone number, explicit preference, random selection. Default
1 – Longest match in phone number, explicit preference, least recent use.
2 – Explicit preference, longest match in phone number, random selection.
3 – Explicit preference, longest match in phone number, least recent use.
4 – Least recent use, longest match in phone number, explicit preference.
5 – Least recent use, explicit preference, longest match in phone number.
6 – Random selection.
7 – Least recent use.

hth

ps: rate useful posts.

[sagar.sarker]

Hi John,

Thanks for your reply. Can you please ellaborate the issue a little bit. We have a voip dial-peer which is directed to the call manager server. Where to accomodate the other voip peer. For your convenience, I'm attaching the current configuration of the voice gateway. Please help me to have the sample configuration for the way out you suggested.

Do let me know if you have any query regarding this.

Best regards,

Sagar

[dijohn]

Sagar,

Could you please copy paste the dialpeer towards the T1/E1(pots) and the one towards the Call Manager(voip).

HTH

[dijohn]

What's the call flow like?

Whoz calling whom?.

Config depends on that

[sagar.sarker]

Hi John,

Call flow is like this -

Users press 7 + the destination number and the call goes through E1 link connected to the voice gateway.

For internal calls, i think internal calls use voip dial peer and connect to the call manager server to reach the destination.

Now the scenerio should be like this, if customer press 7 the call will go throuhg the E1 line and if they press 8, the call will go through the IP line that is connected to the voice gateway. Can SIP trunk be a work around for this?

Thanks in advance for your effort in resolving this.

Regards,

Sagar

[dijohn]

Hi Sagar,

I suppose all your phones are on the CUCM.

So, If the User on the phone dials 7T then It should go to the T1/E1. right?.

I have no idea why you want the user to dial 8T and route it to which other gateway?.

If the T1/E1 is down, where should the call go through?.

IP Phone ---- CUCM --H323- -----GW ---T1/E1 PSTN

I do understand this?.

But why would you loop the call back to CUCM incase the T1/E1 is down?.

I will definitely help you out. But i need to know the exact need of this particular conifg.!

HTH

PS:Rate useful posts.!

[sagar.sarker]

Hi John,

I really appreciate your assistance regarding this.

My concern is to use the backup link in case the primary is down. All the phones are registered with CUCM. Not sure how voip dial peer is working. I think if we have the primary line only, voip dial peer is not required. Please correct me if I am wrong.

In brief, my voice gateway is connected to the E1 line and all the calls are coming and going through the E1 line. But in case if the primary link is down, we should have some other way to pass the calls. Thats why we are trying to connect an IP trunk, so that all the calls pass through the SIP gateway on the other end. How can I accomplish this within our infrastructure.

Please let me know if you have any other query regarding this.

Regards,

Sagar

[dijohn]

Okay.

I got your problem now.

IP Phone -

---

> CUCM --->H323 --- T1/E1

If the T1/E1 fails what happens?.

Right you can do the config on the CUCM itself to route it out another

GW or a SIP Trunk if you wish to do that.

Now the problem with H323 is that once the call is send to the GW, The

CUCM will not by default switch to the backup path. So.. you will have

to disable in enterprise/service parameters.

Stop Routing on Unallocated Number Flag:

This parameter determines routing behavior for intercluster 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.

Once you have done that...create a route-list with 2 route-groups

Put the H323 GW with t1/e1 on the top of the list in the RL. and the

SIP/BackUp GW towards the end. Essentially you don't need two different

access codes "8" and "7" you can manually do the number manipulation in

case of each path

This will reroute the call in case of T1/E1 failure on the H323 GW.

Hope that helps. Shoot a post..if this doesn't help

HTH

PS RATE useful posts!

[MARTIN STREULE]

If you configure the command  "no dial-peer outbound status-check pots" on the H323 gateway, it does not signal a "unallocated number" if the PRI is down but a "temporary failure". The CUCM will try an alternative route - and you don't have to change CUCM service parameters.

hth

Martin

[dijohn]

Martin,

Nice info :-)

[sagar.sarker]

Hi John & Martin,

Thanks for your valuable comments on this issue. I am a bit confused in understanding what you are referring to. Let me clear my demand. We had two voice gateways earlier that had one ISDN PRI connected to each. We configured the system so that if we press 7 the call would go through one PSTN and if 8 was pressed, the call went through the other PSTN line. Now we want to remove one PSTN line from the system and want to connect an IP line from a provider which has SIP gateway at their end. The gateway has a real IP address and our task will be to pass the voice traffic upto that SIP gateway. How this can be achived from the CUCM or from the VG?

Please do let me know if you have further query regarding this.

Regards,

Sagar

[Guru Murthy A]

If i can take that question,

Setup a SIP trunk from the CUCM, Device > Trunk > SIP Trunk.

Configure the provider's SIP gateway address under 'Destination Address' on the SIP trunk config.

Add the SIP trunk to the Route Group in place of the 'removed' PSTN endpoint.

--

Guru

[Senthil Kumar Sankar]

Hi

If you have SIP trunk terminating to Second Mediagateway(MGW),you can go for the below proposed way.

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

So when users dials the number with prefix 8 CUCM will choose the 8T Routepattern which has the MGW2 as H323Gateway

VoiceGateway 2 Configuration :

(config)#sip-ua
(config-sip-ua)#sip-server ipv4:X.X.X.(Sip Signalling IP of the Provider)

dial-peer voice 200 voip
destination-pattern 8T
Session target sip-server
session protocol Sipv2

You would still have the 7T routepattern terminating to First Media Gateway

Phones----7T---->CUCM-----H323/MGCP----->MGW1------>PSTN

So when users dials the number with prefix 7 CUCM will choose the 7T Routepattern which has the MGW1 as H323 Gateway under routelist

dial-peer voice 100 pots
preference 1
destination-pattern 7T
port 0/3/0:15

Regards

Senthil Sankar

[dijohn]

Or you could just get away with the "7" or "8" access code. Make the routing of the call transparent to the end user with Access code "9"

Have a RP 9.! ---> RL

                                   -------> RG1 -----> PRI

                                   -------> RG2------> SIP Trunk

Do digit manipulations RL level/Xformation(Trunk/Gateway)  according to the desired ANI/DNIS on the path out . Senthil's CUBE config looks good.