CME Remote Phone Line

[alexanderbird]

Hi,

I have a 2811 running CME 8.6. In it is a VIC2-2FXO. This connects to a POTS line. On the CME are a few 7940, 7960, 7970, 7912, 7905, 7906. (About 10 phones total) (Site A)

We have a remote office (Site B) which has a phone line in with a different phone number. I would like to have that phone line as a selectable line on one of the IP phones at Site A. (I accept the 7912 are only single line phones)

How do I get the phone line at Site B to "appear" on the CME / Ip Phones at site A. Do I need another CME at site B and some form of interconnecting link (There is already a VPN to that site, and a VPLS link in testing)

Or am I better to extend the phone line over IP, some sort of FXO to FXS over ip and then connect to the spare FXO port of the CME?

Any help much appreciated 

Thanks

Alexander

[Vivek Batra]

Do you mean you don't have CME at siteB. Otherwise, where PSTN lines in siteB are bieng terminated.

I assume you have CME at siteB.

If I understand your requirement correctly, what you want is SiteA user can access the PSTN line of SiteB to make PSTN calls. I don't know if there is straight way to achieve this but you can try following as well.

In SiteA phone, create speed dial (or let user manually dials this) for number say 99 and point this number over SIP Trunk to SiteB gateway. Let this call hit the incoming dial peer in SiteB. Create outbound dialpeer in SiteB gateway configured with desired fxo/analog trunk port. Also configured destination-patter 99 under this outbound dial peer.

Doing so, when site A user presses BLF key (or dial 99), this call will go over SIP to site B gateway and hit the incoming dial peer. Will select the respective outbound dial peer and off-hook the fxo port. By default, 99 digit will be stripped and you will straightaway gets dial tone from the provider.

[alexanderbird]

Hi,

Thanks for your help,

Currently there is no CME at site B - the PSTN terminates on a single analog phone (A very small Site B setup).
From the reply it looks as if we will have to invest in another CME.

From a budget point of view (and space / noise as it is little more than a porter cabin & Cisco routers are not quiet or cheap :) ) would I be able to use some other PSTN to sip server and then create a SIP trunk between the two sites or does it have to be a CME.

Thanks

Alexander

[Mohammed al Baqari]

Hi,

You can use any type of voice gateway or server which can interface with PRI and has SIP capabilities. You can look for asterisk server.

[Vivek Batra]

I don't recommend to use but still there are various analog ATAs available in the market where you can terminate analog trunk in ATA at SiteB and extend the SIP trunk to CME at SiteA. 

[Mohammed al Baqari]

Hi,

If my understanding is correct, your are getting an analog line @ site B and would to extend this line to site A ip phone for inbound/outbound calls.

In this case, you need a CME @ siteB and you need to have a SIP trunk between SiteA CME and SiteB CME.

Now, you can use speed dial, overrelay dn, etc to extend the number to a phone on siteA.

[troussedetoilette]

Oh thank you I had the same problem !