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Need some help.  I have a requirement to forward a PSTN call from PBX on left to PBXs on the right at multiple site locations over an IP WAN network.

PBX---channelized t1----ROUTER(hub)--------1st ds0--IP WAN---ROUTER(SITE 1)-----PBX
                                                         --------2nd ds0--IP WAN---ROUTER(SITE 2)-----PBX
                                                         --------3rd ds0--IP WAN---ROUTER(SITE 3)-----PBX
For example:
(a) a PSTN call comes into hub router from a channelized t1 on 1st ds0, i would like to forward that call to SITE 1's PBX;

(b) a PSTN call comes into hub router from a channelized t1 on 2nd ds0, i would like to forward that call to SITE 2's PBX....etcs.

Is it possbile and how to do it?  a sample configuration or a link will help.

Thanks in advance.

I think it won't be tough. Let's start with ds0-group as you have channelized T1.

!

controller t1 0/0/0

 ds0-group 0 timeslots 1 type r2-digital r2-compelled ani

 ds0-group 1 timeslots 2 type r2-digital r2-compelled ani

 ds0-group 2 timeslots 3 type r2-digital r2-compelled ani

! 

(The signaling type would be set here by consulting the PBX signaling type.)

These three ds0-group will create three seperate voice-ports like below:

!

!

voice-port 1/0:0

 bearer-cap Speech

!

voice-port 1/0:1

 bearer-cap Speech

!

voice-port 1/0:2

 bearer-cap Speech

!

Now create three seperate incoming dial-peers with these voice-ports assigned:

!

dial-peer voice 101 pots

incoming called-number <PhoneNumberPrefix>

port 1/0:0

!

This is important: If you have seperate site codes for each PBX site then define the site codes into expression at 'incoming called-number'. For example, your first ds0-group would be destined to PBX-Site-1 and PBX-Site-1 DNs are started with 1 and DN pattern is 1... or 1T then the expression should be:

!

incoming called-number 1T

!

So the dial-peer should be:

!

dial-peer voice 101 pots

incoming called-number 1T

port 1/0:0

!

dial-peer voice 102 pots

incoming called-number 2T

port 1/0:1

!

dial-peer voice 103 pots

incoming called-number 3T

port 1/0:3

!

If you site dial-plan is not uniform, i.e. not started with site code/not distinguishable then you have to apply translation-rules to seperate and identify each PBX type calls. For example here:

 

!

dial-peer voice 101 pots

incoming called-number .T

port 1/0:0

translation-profile incoming FirstChannel

!

dial-peer voice 102 pots

incoming called-number .T

port 1/0:1

translation-profile incoming SecondChannel

!

dial-peer voice 103 pots

incoming called-number .T

port 1/0:3

translation-profile incoming ThirdChannel

!

You can also apply these translation-profile into voice-ports, not the dial-peers, it makes more understandable right? Create your translation-rules your own. :) An example here:

!

voice translation-rule 1

rule 1 // /11/

!

voice translation-profile FirstChannel

translate calling 1

!

For example you have added prefix 11 infront all of your calls towards PBX-Site-1, so the number pattern would be 11T, for PBX-Site-2 the number pattern is 22T and for PBX-Site-3 the pattern is 33T.

Now create three seperate voip dial-peers towards three routers like here:

!

dial-peer voice 201 voip

destination-pattern 11T

session-target ipv4:<h323 Router-01 IP>

!

dial-peer voice 202 voip

destination-patter 22T

session-target ipv4:<h323 Router-02 IP>

!

dial-peer voice 203 voip

destination-pattern 33T

session-target ipv4:<h323 Router-03 IP>

!

Thus the calls would be routed towards different destination based on the ds0-group channel.

Some of the commands are just typed as I can remember, some of them are copied from routers, please follow the idea not the CLI commands, also excuse any typo and brevity.

HTH!

Thanks MijanurRahma!!!  I got the idea.  

I have few more questions:

- I see the voice channel configurations, do we care for the "control channel" for signalling? or it doesn't need in this case.

- on hub router, should I use a VWIC-T1 to termnate the channelized T1?

- on remote router, since there is only one ds0 for each remote site, should I use FXS port in connecting to PBX?

Thanks again

jchan:

- I see the voice channel configurations, do we care for the "control channel" for signalling? or it doesn't need in this case.

It will depend on the signaling pattern your PBX supports, Cisco router supports variety of variants of R2 and CAS signaling. Your PBX can use in-band or out-of-band signaling, so the HUB router should use the same pattern as PBX is doing. For in-band you dont need seperate signaling channel but for out-of-band you need a seperate channel (16 for E1, 23 for T1).

Here is a beautiful list of specs you need to check from the PBX vendor:

http://www.cisco.com/en/US/docs/ios/12_0/dial/configuration/guide/dcchant.html#wp4669

jchan:

- on hub router, should I use a VWIC-T1 to termnate the channelized T1?

You can use VWIC2-1MFT-T1/E1 or VWIC2-1MFT-T1 OR even NM-HDV2-1T1/E1 or NM-HDV2-1T1

jchan:

- on remote router, since there is only one ds0 for each remote site, should I use FXS port in connecting to PBX?

It's not FXS, it's FXO you need at router end to connect with PBX FSX port. I'm not sure if there is 1-port FXO module, I have never seen but 2 port/4 port is available, part# should be VIC2-2FXO or VIC2-4FXO.

HTH!

MijanurRahman:

For in-band you dont need seperate signaling channel but for out-of-band you need a seperate channel (16 for E1, 23 for T1).

Thanks MijanurRahman!!! I just wanted to clarify that the signaling is local in between hub pbx to hub router, we don't need a seperate signaling channel for each dso and extend the signaling to each remote site.  Is that correct?

Yep, you right, my mistake, it is a FXO port on remote router.

So, the end-to-end topology will look like this:

PBX---channelized t1----(VWIC2-1MFT-T1/E1)ROUTER(hub)---1st ds0----IP WAN----ROUTER(SITE 1)(FXO)-----(FXS)PBX

BTW, do you recommend using 2911 router on hub and 2811 on remote site?

Thanks

jchan:

I just wanted to clarify that the signaling is local in between hub pbx to hub router, we don't need a seperate signaling channel for each dso and extend the signaling to each remote site.  Is that correct?

Yes, if you configure ds0-group, no seperate D-channel/Serial Port is created on the router it means it's an in-band signaling and no seperate signaling channel is needed.

The topology seems correct to me.

Regarding the router model: it's upto you and your budget, these configurations won't vary on ISRs.

One small piece of advice, don't forget to test these configurations on lab first before you go for implementation as I haven't tested the exact scenario you need.

Yes, I plan to test in lab.

Do I need PVDM2-32 on hub and remote router to support the VWIC for 2 voice calls?

Not actually, if you active only three channels at hub router the PVDM2-8 is enough, same for remotes.

Thanks. How to calculate the PVDM?

let's say, I have 20 channels (ds0) on hub router and each remote router will have one channel (ds0)

It's simple! PVDM2-32 can cover 32 channels, PVDM2-64 can cover 64 channel, as simple as it is!

On your example, for hub router you can add PVDM2-16+PVDM2-8 but I will prefer PVDM2-32 at a once. There is a DSP calculator at Cisco you can try (which I really don't like): http://www.cisco.com/cgi-bin/Support/DSP/cisco_prodsel.pl

For Remote routers, if you are planning to put FXO lines there you dont need PVDM, these FXO cards are build with dsp.

HTH!

I also prefer the PDVDM2-32.  For the budget-conscious student, though, it's a pricey endeavor.  

Quick look on eBay:

• PVDM2-16 (used): $65
• PVDM2-32 (used): $110

Thanks again for all your inpurts and comments.  I appreciated.

MijanurRahman / ciscovoiceguru:

some new ideas, what you think?

I just found out that Service provider can support SIP over mpls, so using SIP trunk may work better than my original thinking...at least from the cost prespective.

same requirement: forward pstn call to multiple site location

Here is the new topology:
PBX---channelized t1----ROUTER(hub)-----SIP trunk----->Service Provider SIP Gateway over MPLS-------PSTN--------SITE 1's PBX

Note - in this setup, PBX on hub side can not interface with SIP, we use a router to interface with PBX and Service Provider SIP GW.

same setup & operation goes to site 2, 3...etcs

Good thing about this, Service Provider does all the call routing to remote sites and we save hardware and telco cost on each remote site location.  In talking with Service Provider, they claimed that this is a very standard implementation.

Will this works?  Any comments are welcomed.

Thanks!

Hi,

MijanurRahman:

Who is going to integrate the service provider SIP gateway to PSTN? The service provider itself? If YES, then you have good chance to save one router with FXO ports on each site.

Yes, the Service Provider

MijanurRahman:

what type of TDM connectivity the PSTN is going to offer here? You need only one ds0 channel, so it should be only one analog line, not the PRI.

On remote site, customer has PBX installed and connected to PSTN (i guess via PRI), So, I am thinking SIP gateway can forward call to customer's PRI since this piece of connection is already existied on remote site.  Does it make sense?  Or, should I order an analog line in connecting to customer's PBX?  Yep, you are corrected, only one ds0 channel is needed to remote site, a PRI may be over kill.  What you think?

Thanks!