POTS dial-peer voice config

[dec]

Hello,

we have a CCM4.1 with 100 Phones and a POTS with 400 Phones connected over E1-PRI on Cisco-3660-Router.

The telephone numbers are on both systems without uniformity distributed.

Thus I have on the router for each number extra an entry in this form:

to pots

dial-peer voice xyz pots

destination-pattern xyz

direct-inward-dial

port 1/0:15

prefix xyz

to callamanger

dial-peer voice zyx voip

destination-pattern zyx

session target ipv4:callmanager_x

codec g711alaw

ip qos dscp cs5 media

That makes the configuration however very large and unmanageable.

Are there ideas, how one can make it better?

mfg dec

[vasu_si]

The only way to get large installations manageable is to get the addressing right. The same rule that applies for IP address routing applies here.

[dweiner]

First, a best practice item - in CCM assign the gateway a CSS that does NOT have the partition assigned to the route pattern leading to the PBX. This avoids loops. For example, I would have a partition Internal_IP_PT and assign that to all the DNs in CallManager, and a partition of PBXPhone_PT and assign that to the route pattern leading to PBX extensions. The Gateway CSS would have Internal_IP_PT but NOT PBXPhone_PT. All IP phone CSSs that can call the PBX would, of course, have both. This prevents incoming calls from the PBX from being routed back out to the PBX and creating a loop.

Second, In CallManager insert a digit in the Route List -> Route Group Details for all outbound station-to-station calls to the PBX. Use something out of the dial plan, like *. Now you can separate calls TO CCM and calls FROM CCM. Your dial-peers could look like:

dial-peer voice 123 pots

destination-pattern *T

direct-inward-dial

port 1/0:15

no prefix

dial-peer voice 321 voip

destination-pattern [2-8]....

session target ipv4:callmanager_x

and all the rest (though I'd recommend using a codec class that included g.729 as well as g.711 in case you get into using regions in CCM).

The end users will never know that the extra digit is inserted, and it makes your dial-peers very easy.

Don

[dec]

Thank you for the tip! I will try it.

mfg dec

[dec]

Hallo, and a further question

For better coupling with a Siemens HICOM we want to change gateway from H.323 to MGCP.

How should one configure that?

MfG

dec

[dweiner]

hostname xxxxxx

ip domain-name yyyyyy

mgcp

ccm-manager mgcp

ccm-manager config server zzz.zzz.zzz.zzz

ccm-manager config

x = whatever name you want for the gateway.

y = optional domain name

z = ip address of CCM tftp server (comma separate for multiple tftp servers)

That sets up the router for download from CCM for the config (like with IP phones). hostname is required, ip domain-name is not. You want to make sure you administer in CCM the gateway name the same as the router is configured. The easiest way to find out what the full name is = 'show ccm' in the gateway. The first line you get back is what you need in CCMAdmin for the gateway 'Domain Name' field. After that CallManager will configure the specifics in the router after you configure via web pages. Note that 'show ccm' will show when the gw registers with CCM, but CCM web pages won't show the gateway as registered until the trunk comes up (if T1/E1).

Don

[dec]

Hallo !

... and as say I to the callmanager that "dial-peer voice pots" and "dial-peer voice voip", as in the upper example with H.323 ?

Thanks

dec

[dweiner]

All you have to do are the aforementioned commands. CallManager will create necessary dial-peers, E1 parameters, etc.

If you want to administer SRST parameters do so after CCM has downloaded the MGCP configuration. For SRST you do create dial-peers as the gateway 'falls back' into H.323 mode if it can't connect to CCM.

Don