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4 digit dialing between 6 uc560 sites

James Lyons
Level 1
Level 1

Hello I have a customer with 6 different sites.

We have installed a UC560 plus a ESW switch at each site.

We are replacing an older VoIP system. The existing system allows the customer to do 4 digit dialing between sites.

I know this is possible but my question is how do I do this?

I have searched and have been told its possible to do this via the CCA as well as via the CLI.

My next question is which way is best and can someone point me in a direction that gives me the instructions of how to accomplish this.

Thank You (my first post).

2 Accepted Solutions

Accepted Solutions

Thanks for pointing that out, but that was intended as a reply to the maximum calls James could have b/n sites over the T1. The limit makes no sense for calls over IP, i agree. But doesnt the calculation and maximum calls stand valid for the Inter Site calls over the T1

It does and I haven't commented it.

My recommendation however is for pervasive g.711u for reasons of simplicity and robustness.

View solution in original post

Hi Paolo,

Yes i agree G.711 would be a lot less hassle free. But i guess it would be up to James to decide based on the maximum number of calls he wants and the desired quality.

James I hope this helps you to decide - http://www.cisco.com/en/US/tech/tk1077/technologies_tech_note09186a00800b6710.shtml

Regards

Thomas Dominic

View solution in original post

13 Replies 13

dksingh
Cisco Employee
Cisco Employee

HI there,

All u need is to determine codec u want to use

between the two sites (depending on BW) and

then configure h.323 or sip dialpeers for your

dialplan...

A couple of notes on dialplan to keep it simple:

- try avoiding overlapping extn/DN between sites

- try using a unique site code for each site...

Let me know if u hv further ques and I'll try to find

u a doc that has an example for setting up simple

voip calls between two sites...

Thx.

DK

Take a look at following which covers dialpeers and dialplan concepts on IOS:

http://www.cisco.com/en/US/docs/ios/12_2/voice/configuration/guide/vvfpeers.html

And, I would not worry abt. using CCA for this....I find CLI much easier.

Thanks for the answer. I'm looking at the doc you posted for instructions.

as far as bandwidth is concerned there will be T1 between sites.

With that said what is a good codec to choose?

Sure...

You need to figure out the max number of calls that could be active

between the two sites and if you need BW for other data as well.

Use G711 if you have enough voice BW for the max active calls as that

would eliminate the need for any transcoding; If BW is limiting, go

with g729r8 (default) codec but make sure to configure some transcoders

on each side as there would be calls forwarding to CUE/VM.

Roughly G711 will consume 82K per call compared to 26K for g729.

http://www.cisco.com/en/US/tech/tk652/tk698/technologies_tech_note09186a0080094ae2.shtml

Hi James,

I infer you are using UC560-T1E1-K9 (UC 560 system with 4 FXO, 4 FXS, 1 T1/E1)

Reference guide for the same –

http://www.cisco.com/en/US/prod/collateral/voicesw/ps6788/vcallcon/ps7293/reference_guide_c07-566560.pdf

Now the UC 560 has 8 DSP’s so you can maximum of :

                                   

                                    64 channels with G.711

                                    32 channels with G.729

Since u have T1 b/n the sites with T1 with b/w of 1.54 Mbps u can have maximum of

                                    17 calls with G.711

                                    49 calls with G.729

http://www.cisco.com/en/US/tech/tk652/tk698/technologies_tech_note09186a0080094ae2.shtml#topic1

So since u have sufficient DSP’s and insufficient b/w I suggest you go for G.729 Codec and you can always go for more DSP’s if you find the need for more calls.

Regards

Thomas

P.S. Please rate helpful answers

tdominic wrote:


Now the UC 560 has 8 DSP’s so you can maximum of :

                                   

                                    64 channels with G.711

                                    32 channels with G.729

Please note that DSPs are only used for calls using voice interfaces, like fxs, fxo, bri, or pri.

Call between phones do not use any DSP resources. Consequently, for sites connected via IP, there is no need for DSPs.

Hi Paolo,

Thanks for pointing that out, but that was intended as a reply to the maximum calls James could have b/n sites over the T1. The limit makes no sense for calls over IP, i agree. But doesnt the calculation and maximum calls stand valid for the Inter Site calls over the T1

Thanks and Regards

Thomas

Thanks for pointing that out, but that was intended as a reply to the maximum calls James could have b/n sites over the T1. The limit makes no sense for calls over IP, i agree. But doesnt the calculation and maximum calls stand valid for the Inter Site calls over the T1

It does and I haven't commented it.

My recommendation however is for pervasive g.711u for reasons of simplicity and robustness.

Hi Paolo,

Yes i agree G.711 would be a lot less hassle free. But i guess it would be up to James to decide based on the maximum number of calls he wants and the desired quality.

James I hope this helps you to decide - http://www.cisco.com/en/US/tech/tk1077/technologies_tech_note09186a00800b6710.shtml

Regards

Thomas Dominic

James Lyons
Level 1
Level 1

Hey guys thanks for all the insight.

I've read over the documents supplied and I'm going to try to configure things between two sites for testing.

Any other insight is welcome as I've just learned so much in a few hours.

James Lyons
Level 1
Level 1

Hey guys I just tried this to test and I wasnt successful.

Here are the configs I added at each site:


dial-peer voice 4333 voip
description Site1 to Site2
destination-pattern 16[0-2].
b2bua
session protocol sipv2
session target ipv4:192.168.4.10
dtmf-relay sip-notify
codec g711ulaw
no vad

***Site 2 has extensions 1600-1620***


dial-peer voice 4333 voip
description Site2 to Site1
destination-pattern 20[0-2].
b2bua
session protocol sipv2
session target ipv4:192.168.4.10
dtmf-relay sip-notify
codec g711ulaw
no vad

***Site1 has extensions 2000-2020

I tested this and it didnt work. What am I missing?

I just get a fast busy when trying to call from site to site.

Hi James,

Your dialpeers need little tweaking...both are pointing to the same IP!

Pl. try this

- add incoming called-number 20.. to Site 1 dialpeer
- modify session target ip on site 1 dialpeer to point to site 2 IP
- add incoming called-number 16.. to Site 2 dialpeer
- modify session target ip on site 2 dialpeer to point to site 1 IP
- get rid of b2bua from dialpeer

shd also add following to both sites....

voice service voip
allow-conn sip to sip

HTH,

DK

James Lyons
Level 1
Level 1

Hello guys I was able to get this working using the following configuration (this is the headend UC560 but its very similar on the other UC's):

!
dial-peer voice 5555 voip
description Intersite to Site1
preference 1
destination-pattern 20..
session target ipv4:172.16.1.10
codec g711ulaw
no vad
!
dial-peer voice 5556 voip

description incoming for all sites
incoming called-number .
codec g711ulaw
no vad
!
dial-peer voice 5552 voip
description Intersite to Site2
preference 1
destination-pattern 23..
session target ipv4:192.168.2.10
codec g711ulaw
no vad
!

Thanks to everyone for your help.

If anyone has insight or comments on the above config, please let me know.

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