Multisite - Same Extension Question

[inewman]

Hello,

We have two sites each with a UC520 – HQ and Branch. Can they be set up so the HQ has SIP to our Telco provider for PSTN access and the branch uses the SIP connection from the HQ over IPSec?

Also our CEO needs a phone at the branch with his HQ extension. Ideally we would like them to ring simultaneously, this way no matter where he is the phone can be answered without log in’s. Can this be done?

Thank you!

[pierrescotland]

There's no problem with using the SIP provider at the HQ from the branch, we do this a lot.  A simple dial-peer pointing to IP address of the HQ UC520 will suffice, you may have to do number-translation depending on the existing dial-plan.

Easiest way to accomplish your second goal is to run another extsion directly off the HQ system, just assign a button to his number.  (but then you will lose all the local brach facilities (Page/Park/etc).

Alternatively you can use Single Number Reach to ring any other phone.

https://supportforums.cisco.com/docs/DOC-15822

hope this helps.

[johschaf]

Hello,

It is possible to configure TEHO(Tail End Hope Off)/LCR(Least Cost Routing) on the UC500s, but it is only possible through CLI. In addition, since your PSTN connection is SIP that adds a bit of complexity to the configuration.

After doing the initial multisite configuration, you will need to modify the multisite dial peers to accept more digits. You will also probably need to strip the codec settings off of the dial peers.

As for the CEO phone, there is a few ways I can think we could do this, but it might not be exactly what you are looking for. Since each site will have its own UC, there is really no way to share the primary extension at HQ. A few things you could do would include putting in a Single Number Reach configuration to ring the branch phone if not answered at HQ, you could create a blast group at HQ that rings both the HQ and Branch extensions, or you could create a second phone at HQ that connects back from the Branch over the SSLVPN. With the SSLVPN you could use the same extension as at HQ, but this only works with a SPA525.

Hope this helps. Let me know what questions you have regarding this.

Thanks,

-john

[inewman]

Hi John,

I am not clear on the difference between using TEHO/LCR or using a dial-peer at the HQ pointing to the branch?

I like the idea of the blast group at the HQ but this poses some problems with voicemail and outbound calls:

Is there a way to make the branch extension VM light illuminate when the HQ phone receives a VM?

Also when he places out bound calls the receiving party will not see his DID.

The CEO will have a SPA525G2 at home as well.

Is there a better Cisco product to accomplish this design?

Thanks everyone for your help!

[johschaf]

Hello,

The TEHO/LCR will use a dial-peer to achieve that functionality.

There is not a way to configure the remote site to receive a voicemail notification on the phone. You can configure voicemail to email or voicemail notification to the remote phone DID. You could mask the remote caller ID with the HQ caller ID, but that configuration will be dependent on your service provider allowing you to mask the caller ID with another number that is not assigned to the remote site.

This is definitely possible on some of the other VOIP PBX solutions like Call Manager Express on a ISR router or full blown Call Manager.

Hope this helps. Let me know what other questions you have regarding this.

Thanks,

-john

[inewman]

Hi John,

I have site to site extension dialing working. Next I would like to configure the branch to place all calls through the HQ site.

Site to Site config:

HQ:

voice translation-rule 1112

rule 1 /^9/ //

!

voice translation-rule 1114

rule 1 /^01\(....\)/ /\1/

!

voice translation-profile Multisite_Internal

translate called 1114

!

voice translation-profile OUTGOING_TRANSLATION_PROFILE

translate called 1112

!

dial-peer voice 3012 voip

translation-profile incoming Multisite_Internal

destination-pattern 31..

session target ipv4:10.2.102.20

incoming called-number 31..

dtmf-relay h245-alphanumeric

codec g711ulaw

fax protocol cisco

no vad

!

Branch:

!

voice translation-rule 1113

rule 1 /^01\(...\)/ /\1/

!

voice translation-rule 1114

rule 1 /\(^....$\)/ /01\1/

!

voice translation-profile multisiteInbound

translate called 1113

!

voice translation-profile multisiteOutbound

translate calling 1114

!

dial-peer voice 2100 voip

corlist incoming call-internal

description **CCA*INTERSITE inbound call to 181

translation-profile incoming multisiteInbound

destination-pattern 31..

session target ipv4:10.1.102.20

incoming called-number 31..

dtmf-relay h245-alphanumeric

codec g711ulaw

fax protocol cisco

no vad

!

dial-peer voice 2101 voip

corlist incoming call-internal

description **CCA*INTERSITE outbound calls to 30

translation-profile outgoing multisiteOutbound

destination-pattern 02....

session target ipv4:10.1.102.20

dtmf-relay h245-alphanumeric

fax protocol cisco

no vad

!

Think I need to make a dial-peer for 7 digit numbers next. Am I on the right track with the following configuration at the branch?

dial-peer voice 2102 voip

corlist outgoing call-local

description outbound call to local through 30

translation-profile incoming multisiteInbound

destination-pattern 9[2-9]......

session target ipv4:10.1.102.20

dtmf-relay h245-alphanumeric

codec g711ulaw

fax protocol cisco

no vad

Thank you!

[johschaf · Accepted solution]

Hello,

You are on the right track, but there is some issues with the dial peer 2102 configuration at the remote site. Generally speaking, you do not want to use the same steering digit on the multisite dial peers as the main steering digit, but maybe your scenario is different. Since the dial peer currently references the translation-profile incoming multisiteInbound, you will need to correct that. If you are planning on keep the current steering digit remove the translation rule completely. This way the 9+7 digit local will be passed to the HQ UC, which should then match an outgoing dial peer with destination pattern of: destination-pattern 9[2-9]......, translate the number removing the 9, and then send the call out of your SIP trunk.

Also, pay particular attention to the codecs being used on the dial peers. Since you are using SIP depending on the codec being used by your SIP provider you might have to do some transcoding or hard code the codecs throughout.

Let me know what questions you have regarding this.

Thanks,

-john

[inewman]

I changed the remote site dial peer to the following:

dial-peer voice 2102 voip

corlist outgoing call-local

description outbound call to local through 30

destination-pattern 9[2-9]......

session target ipv4:10.1.102.20

dtmf-relay h245-alphanumeric

codec g711ulaw

fax protocol cisco

no vad

When I enable the following debug on each end I dont see any debugs on the HQ when dialing a 7 dig number with 9 from the branch. The branch phone does not ring, I only get silence.

H.225:

H.225 ASN1 Messages debugging is on

H.245:

H.245 ASN1 Messages debugging is on

CCAPI:

debug voip ccapi inout is ON (filter is OFF)

Any suggestions?

[johschaf]

Hello,

You need to do the following debug: debug ccsip messages

This should show why the call fails.

Thanks,

-john

[paolo bevilacqua]

You need to do the following debug: debug ccsip messages
This should show why the call fails.

It won't, because OP is using H.323, not SIP.

[inewman]

It works now, thanks John!

Branch:

dial-peer voice 2102 voip

description outbound call to local through HQ

destination-pattern 91[2-9]..[2-9]......

session target ipv4:10.1.102.20

dtmf-relay h245-alphanumeric

codec g711ulaw

fax protocol cisco

no vad

HQ:

dial-peer voice 3012 voip

translation-profile incoming Multisite_Internal

destination-pattern 31..

session target ipv4:10.2.102.20

incoming called-number 31..

dtmf-relay h245-alphanumeric

codec g711ulaw

fax protocol cisco

no vad

[bkwon]

regarding Multisite deployment with UC500,

- have to use interworking mode in H323.

voice class h323 1

call start interwork

- transcoding is not supported in H323 interowk mode, which means you have to use G711ulaw in multisite and SIP trunk as well, since CUE uses G711ulaw only.