SIP phones, DN and Pool concepts

leam2 · ‎05-23-2018

Hello.

Please consider two SIP phones (model Cisco CP_8821) in a CUCME system (v. 10.5).

For now, I just want one IP phone (phone_1) to be able to call the other IP phone (phone_2) and the other way around inside the LAN.

Just to be clear:
- on phone_1, I would compose a number corresponding to phone_2 and phone_2 would ring and I would be able to place the call and speak,
- on phone_2, I would compose another number corresponding to phone_1 and phone_1 would ring and I would be able to place the call and speak.

Please consider the chunk of configuration below:

(config)# voice register pool 1
(config-register-pool)# id 413A.F352.67B1
(config-register-pool)# type 8821
(config-register-pool)# number 1 dn 1

(config)# voice register pool 2
(config-register-pool)# id 413A.F352.43AC
(config-register-pool)# type 8821
(config-register-pool)# number 1 dn 2

(config)# voice register dn 1
(config-register-dn)# name phone_1
(config-register-dn)# number 101

(config)# voice register dn 2
(config-register-dn)# name phone_2
(config-register-dn)# number 102

With the configuration above, on phone_1, can I directly compose the sequence 102, to have phone_2 ring and then start a conversation?
With the configuration above, on phone_2, can I directly compose the sequence 101, to have phone_1 ring and then start a conversation?

I do not really understand the concept of a directory number (dn) and a pool.
Why pool? "pool" sounds like a "group" but we associate a phone MAC address to it, which is unique, definitely not a group.
Also, if a directory number is a voice channel, when we have two phones communicating with each other, isn't there only one voice channel or two?
In (config-register-dn) mode, is "number" what I described above that is the number of a phone that has to be composed to have this phone ring and start a conversation with it?

Additional question:

Later I'll have to place calls outside from the LAN with only one phone line.
Do I have to plan something to permit:
- phone_1 to call phone_2,
- phone_2 to call phone_1,
- phone_1 to call a phone number like 0112541321 or 33112541321 which is an external number.
What I mean is that should I plan something like:
- press key 9 and then the number of a local phone (101, 102 in my case),
- press key 8 and then the number of a remote phone (0112541321 or 33112541321)?

Would it be the right way to do things?
What would be the corresponding config.?

Thank you for your help.

Maren Mahoney · ‎05-23-2018

The phone configuration portion you posted of your SIP CME looks fine, but as you did not post the complete config I can't say whether the two phones would be able to call each other. For instance, the following is required. (There is more, but this is an example):

voice service voip
  allow-connections sip to sip
  sip
    registrar server

As for dn and pool:

DN: Directory number. In SIP CME these are created with the 'voice register dn' command.

Pool: This represents device(s) in SIP CME. There are situations (like SIP SRST) where a voice register pool does represent a group of devices, rather than a single device. This is how SIP CME is configured, so just go with it for now.

DNs and Pools are created separately, but will be associated with each other. This gives you flexibility with regards to which numbers show up on which phones on which phone buttons. The voice register dn creates the number itself and allows you to assign parameters to the dn. Then, under voice register pool, you assign a dn to a line appearance. In your config, "number 1 dn 2" puts 'voice register dn 2' onto the first button on the physical phone.

So, in your example, if someone dials the digits 102 then the first button on the phone that is 'voice register pool 2' would ring.

You are right that a single phone call is a pair of RTP streams, but the two streams together take a single 'channel' with regards to the phone. The channel represents a single (full) phone call.

Your remaining questions are not something that can be answered quickly, but will take some reading on your part. There is an ocean of documentation, but here are a couple of resources that might be useful. Please read up on SIP CME and on IOS-based dialplans, and let us know if you have questions.

Maren

Cisco Unified Communications Manager Express: SIP Implementation Guide

Cisco CME Administration Guide - Dial Plans

gmgarrian · ‎05-23-2018

Great info Maren!

The only thing I would add is that for outbound calls to the PSTN you will need to configuration dial peers:

https://www.cisco.com/c/en/us/td/docs/ios/12_2/voice/configuration/guide/fvvfax_c/vvfpeers.html

These would dictate whether you would need to press 9 or 8 prior to the desired number.