Enable SIP features on Nortel Communication Server 1000 and Signalling Server

[sarahtanembaum]

Dear Nortel VoIP guru,

1) How do you enable SIP on Nortel CS1000 and Nortel Signalling Server? If required license, how much it cost?
2) Currently we are using Nortel IP Phone 1140e running firmware 0625C7J. Is it possible to convert 1140e to be SIP-enabled phone? Can it run both SIP and UNISTIM? If upgraded to SIP, is it possible to revert back?
3) Can Nortel IPSoftPhone 2050 run on iPhone, Windows Mobile, and/or Android devices? If it can't, is there 3rd party Nortel IPSoftphone 2050 client for the Mobile Devices?

Thanks

[sarahtanembaum]

I also forgot, is it possible to interface asterisk with Nortel Communication Server 1000? Where do I find the info as how to do it?

Thanks.

[Michael McNamara]

1) How do you enable SIP on Nortel CS1000 and Nortel Signalling Server? If required license, how much it cost?

You really need to consult a voice reseller... you need to be running at least 5.5 or later (highly suggested to be running 7.0 or later) and you need to purchase SIP trunk and line licensing.

2) Currently we are using Nortel IP Phone 1140e running firmware 0625C7J. Is it possible to convert 1140e to be SIP-enabled phone? Can it run both SIP and UNISTIM? If upgraded to SIP, is it possible to revert back?

Call Server release 5.5 or later can run both UNIStim and SIP IP trunks at the same time... again you need the licensing to support it.

3) Can Nortel IPSoftPhone 2050 run on iPhone, Windows Mobile, and/or Android devices? If it can't, is there 3rd party Nortel IPSoftphone 2050 client for the Mobile Devices?

In theory if you upgrade to Call Server 7.5 and deploy Avaya Session Manager you should be able to use any third-party SIP client.

Unfortunately the fact is that SIP is not really the great standard it was supposed to be. You have Nortel SIP, Avaya SIP, Cisco SIP, etc, all of which are not compatible with each other. Now enter Avaya Session Manager or Cisco Session Manager, these two offer abstraction layers to help merge the different SIP protocols.

Again you really need to consult a voice reseller, it's very complicated stuff and there are no manuals.

Hopefully I've answered some of your questions.

Good Luck!

[sarahtanembaum]

Michael, thank you.

I am having this dilemma whereas we have data and voice department whereas the voice guys are not too keen on making the VoIP to work. What I'd like to do is to do proof-of-concept with minimal disruption on the voice side(e.g. without asking to get bri or pri line),  though there is a plan to merge both data/voice as one department.

I've came across an article a while ago(but lost them) on how to use Nortel Extension as the pots line to the Asterisk box. That is, using the Nortel extension line, asterisk box will receive/sending voice call. Or, perhaps ask for minimum 2 analog line to the Asterisk box for testing and proof-of-concept.

If anyone has it, please share it in this forum.

Thanks  again.

[buckman]

I also forgot, is it possible to interface asterisk with Nortel Communication Server 1000? Where do I find the info as how to do it?

hi sarah. yes its possible, you need a sip trunk. thats simple to setup, we run 5.5 here with sip trunking.

all my pure sip client (android, ipad, linksys SIP ATA and sip deskphones) runs off the asterisk, goes to the CS1000 and outside lines via a sip trunk. i do have a few audiocodes running of the sip trunk too, but they are few. we plan on going to 7.5 soon (just after christmas) and learned the my old quintum running h323 wont work anymore. so here again, the little asterisk to the rescue. we had a go at the AG2000 too, which  was an intercom solution. bought that, just to be f***ed by avaya a couples of months better. here again, the little asterisk runs intercom and paging, along with call recording. look into it, its worth every penny.