How to prevent call routing loop between CUCM6 and pbx

[markjiang]

Hi,

Is there a service parameter to prevent call routing loop between CUCM6 and pbx.

There are CUCM6 and PBX in my company.

CUCM6--GW--E1--PBX

DNs of IP and analog Phone are in the range of [1-8]xxx.We have config a route patter [1-8]xxx pointed to PBX.When users of IP phone dial a DN which isn't a DN of IP Phone,CUCM will sent this call to PBX.When users of IP phone dial a DN which isn't a DN of analog Phone,PBX will sent this call to CUCM.

So when user dial a unsigned number,call routing loop occur.

I know we can config a proper Inbound CSS of GW to prevent the loop.However,Is there maybe a service parameter like "forward maximum hop hount" to prevent call routing loop between CUCM6 and pbx?

Best regards

Mark

[paolo bevilacqua]

There is no maxiumhop count, because there is no easy way for a router to tell that calls are the same and looping.

You should configure so that either CMor GW knows which numbers are oneside which on the other. If numbers are intermixed,it will be very difficult to avoid loops.

[markjiang]

The numbers are intermixed.We config each unsigned number into DN of a dead phone,but it is hard to maintain.

[paolo bevilacqua]

You're suffering the consequences of a poor design, now either reassign numbers or keep the difficult maintenance.

[Ronald Spencer]

Hi mark,

This is common in the mixed environment scenario. Probably one of the easiest things to do would be a site prefix.

You would config CM to add a digit to the DNs it passes to PBX. On the PBX, you would configure it to strip the prefix (in order to match the local phones). You would do the same for the PBX (configure to prefix digit when sending calls to Call Manager) and then Call Manager to strip digit when receiving call from PBX.

Another way (as mentioned earier) would be to modify the dial-plan, still using site prefix. the way these 2 differ is that you would not strip the digit (instead, you are adding it to the DN and using the mask to ensure that the proper ANI is sent).

Of the 2, the second is probably easier (depending on the flexibility you have with your legacy PBX), from a config stand-point, but #1 would be easier from a user experience standpoint.

[paolo bevilacqua]

The thing (as you implied) is that when doing #1 you need major help from PBX technicians, but often time they are not so cooperative, also often customer gives you too short maintenance window.

[Ronald Spencer]

Can you give me a sample of you dial plan, 4-digit, 5-digit? Are there individual DNs that are duplicate?, or is it only the ranges that overlap?

[markjiang]

Hi t00832112,

Thx for your advice.

The dial plan is 4-digit and there are no individual DNs that are duplicate.

Using site prefix makes user experience bad and maintain more complex.My boss won't allow me to do like this.

Best regards,

Mark

[Ronald Spencer]

Mark,

As I started to type out this long-winded response, it occured to me...Can we use translation patterns to poison routes (in order to prevent the call routing loop)?

Or how about this scenario,

If the PBX is tied to Call Manager, make the inbound on that connection restricted (so that it does not have access to get back to itself).

I am trying to decide how best to accomplish either, but I don't have firm resolution now.

If any other NetPro'er can lend their expertise, I am sure that Mark would appreciate it (as would I). If I can put together the actual config for either scenario, I will post it here.

-Ronnie

[markjiang]

Hi Ronnie,

Thx for your patient reply.

Maintenance of PBX isn't the primary task of the IT staff.So he isn't familiar with tech of PBX.

And there are few designs of dial plan like this.

Therefore maybe configing a restricted inbound CSS of GW is the proper way of preventing the loop.

Mark

[Ronald Spencer · Accepted solution]

Hi Mark,

It just came to me. To accomplish this you need a few things.

1) the PBX TieLine needs to be in a different partition than ONNet.

2) Inbound calls (to call manager from PBX) will be assigned the restricted CSS (ONNet only), which will give them access to the DNs and not the trunk.

3) outbound calls (from call manager) will be placed in a CSS that has access to that ICT (for the purpose of routing the call to PBX -when necessary).

I think that you can accomplish all of this without digit manipulation.

Probably would look like this:

Trunk's inbound CSS would have access to ONNet only

the CSS of the Call Manager DNs would have access to the partion associated with the route Pattern you created [1-8]XXX.

there are 2 things to consider when applying this config.

1) if the PBX is not directly tied to the PSTN, your inbound CSS on the ICT should have access to the partitions that will grant this access. The easiest way to accomplish this is probably to copy the config of the Inbound CSS on the ICT and remove the ONNet partition).

2) Without being able to update the config on PBX, it will be prossible for a call (originating on call manager) to be returned for routing. It will not, however, loop, as when the PBX routes it back, call manager will reject it (no matching patterns).

If you need further detail, or help configuring this, I will be glad to assist.

[markjiang]

Hi Ronnie,

Thanks a lot.What you said is what I need.

Mark

[Ronald Spencer]

No problem, I am glad I could help.

ONe thing though...I noticed (when re-reading my post) that I incorrectly said that the CSS of the ICT should not have ONNET (from #1 at the end). What it should have read is no access to the ICT (as mentioned previously). If the CSS applied to the inbound ICT does not have ONNet, no calls from PBX to CCM phones would complete.

[markjiang]

Hi,

A New problem appears.

Topology:

CUCM--GW--T1--PBX

When the protocol of GW is MGCP,we can config a restricted inbound CSS of GW to prevent loop.But when the protocol of GW is h323 ,the loop continues.

configuration like this:

dial-peer voice 1000 voip

destination-pattern [1-8]...

session target ipv4:192.168.100.248

dtmf-relay h245-alphanumeric

no vad

The call comes from PBX will be sent back to PBX immediately.

Best regards

Mark

[lwai · Accepted solution]

I had the same issue, the Corlist help me solved the issue.

Define the following:

!

dial-peer cor custom

name to_cm

name to_pbx

!

!

dial-peer cor list TO-CM

member to_cm

!

dial-peer cor list FROM-CM

member to_pbx

!

under each dial-peer voice, add the corlist:

dial-peer voice 2000 voip

corlist incoming FROM-CM

corlist outgoing TO-CM

preference 1

destination-pattern 2...

progress_ind setup enable 3

progress_ind connect enable 1

progress_ind disconnect enable 8

session target ipv4:172.16.1.204

dtmf-relay h245-alphanumeric

codec g711ulaw

fax-relay ecm disable

fax rate 14400

fax nsf 000000

ip qos dscp cs5 media

no vad

!

dial-peer voice 2001 voip

corlist incoming FROM-CM

corlist outgoing TO-CM

preference 2

destination-pattern 2...

progress_ind setup enable 3

progress_ind connect enable 1

progress_ind disconnect enable 8

session target ipv4:172.16.1.201

dtmf-relay h245-alphanumeric

codec g711ulaw

fax-relay ecm disable

fax rate 14400

fax nsf 000000

ip qos dscp cs5 media

!

dial-peer voice 2002 voip

corlist incoming FROM-CM

corlist outgoing TO-CM

huntstop

preference 3

destination-pattern 2...

progress_ind setup enable 3

progress_ind connect enable 1

progress_ind disconnect enable 8

session target ipv4:172.16.1.202

dtmf-relay h245-alphanumeric

codec g711ulaw

fax-relay ecm disable

fax rate 14400

fax nsf 000000

ip qos dscp cs5 media

no vad

!

Remark: under the highest preference no, add

the huntstop command.

After I applied above commands, the call loops betweem CM and PBX caused by end user dialing any un-assigned DN solved.

Regards,

Michael.