Routing to Polycom RMX 2000

[TimPNRHA]

Good Morning Everyone,

I do not know if anyone has done this before or has an idea where to begin.

Currently our we are 5.0 and our video conferencing solution is Polycom with CMA 4000 and RMX 2000. What I am looking to accomplish is to have a 800 number on our PRI and have it routed to the RMX 2000 to enable audio participants into conference video/audo calls or audio only conferences but I cannot find any documentation on how to accomplish this.

[LoopyLou]

Can I assume your 5.0 means a 3300 controller running MCD 5 software. Not familiar with the Polycom so what is the method of connection to the 3300? Would assume SIP trunks perhaps? How would I access the polycom if I was a user on the Mitel ( how would I dial it from my phone )? What I think you need to do is create a speed dial on the Mitel to call the polycom. So if I dial 3000 on my Mitel phone to join a conference then I would create a speed dial that dials 3000. The 1-800 will be delivered to the Mitel via the PRI as a number of digits. You create a speed dial matching those digits. So if you dial 1-800-555-1212 and the carrier sends 4000 down the PRI you create a speed dial 4000 that dials your polycom access 3000. It should then work.

[ralph]

Can the Mitel pass video now?

Ralph

[LoopyLou]

Polycom is a stand alone.

[TimPNRHA]

Can I assume your 5.0 means a 3300 controller running MCD 5 software. Not familiar with the Polycom so what is the method of connection to the 3300? Would assume SIP trunks perhaps? How would I access the polycom if I was a user on the Mitel ( how would I dial it from my phone )? What I think you need to do is create a speed dial on the Mitel to call the polycom. So if I dial 3000 on my Mitel phone to join a conference then I would create a speed dial that dials 3000. The 1-800 will be delivered to the Mitel via the PRI as a number of digits. You create a speed dial matching those digits. So if you dial 1-800-555-1212 and the carrier sends 4000 down the PRI you create a speed dial 4000 that dials your polycom access 3000. It should then work.

That is exactly what I am trying to accomplish and I am thinking that I am not doing the sip trunking correctly because I am not using a sip provider but trying to utilize the MCD 3300 (which is version 5.0 sorry was exhausted that day) as a sip register. Currently I am utilizing the try before you buy option with 5 Sip Trunk Licenses. This is so far what I have completed which I might have screwed up something along the way:

1. Created a Network Element Labelled Polycom selected other and pointed to my Polycom RMX (MCU)
2. Created a SIP Peer Profile with no Authentication Required
3. Created Route for my SIP Peer to go through the Trunk Group connected to the PRI
4. Currently just a single DID programmed with a System speed call of 59800 which is the entry que number on my Polycom RMX to the IVR

Is that about right or am I missing something?

[Dutch]

I might be interpreting this wrong as to what you have done already or how you want it to work, but if the Polycom is connected to a SIP trunk and if the link is UP and functional (SIP LINK STATE ALL will show probably) and reacts on the 59800 number would you not:

- Program a route in "ARS Routes" to the SIP trunk (Leave settings default as in no restriction and no digit mod, which is probably 1 in both cases)

- Program the 59800 number in the ARS Digits dialed to that route number you just programmed pointing to your SIP trunk with digits to follow 0?

- Make a DID in the speeddial pointing to the ARS entry corresponding to the number you dial internally and which corresponds to the external number, e.g. Speeddial = 3000 -> Actual number = 59800

- You can then dial the speedcall or ARS entry and you should end up in the Polycom receiving a call through the SIP trunk. You can then also program it in your Teldir but you can not make the number private (just saying)

As said I might be reading this wrong, but this is how I am used to do this, as it works the same with PRI -> PRI conference systems.

Make sure the "trunk attributes" you programmed for the SIP trunk mention "Dial In trunks Incoming Digit Mod = 0" for the SIP trunk as well, as you don't want any modifications in the most cases.

As said I might be interpreting this wrong but this is how I would do it roughly.

Dutch