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View Full Version : Testing the GXP2200



uf_shane
11-04-2013, 03:18 PM
So we received a Grandstream GXP2200 today, and it works great. One thing I noticed is that you have to put your user id in both the "SIP USER ID" and "
SIP Authentication ID" fields or it will not register.

JacobsLive
11-05-2013, 11:37 AM
Yes, I have been using this phone with VOIPO for couple of months now. It is really a nice phone with lot of customization options. Yes for some reason, it need both fields to be filled to successfully authenticate the account.

Another thing I noticed that you need to remove all other codecs and keep only PCMU in order to work smoothly with VOIPO. If you guys have any other tips, please share them :)

uf_shane
11-05-2013, 01:24 PM
I actually did not have to make any other adjustments other than having both fields filled out

chevyman
02-01-2014, 03:17 PM
I have been looking at this phone, what things have you guys found out?

Does this work good with the voipo PBX system?

Thanks!

uf_shane
02-03-2014, 07:35 AM
Yes its working great, I have a PBX connected to it along with a couple of direct lines. Works like a charm.

chevyman
02-14-2014, 10:23 PM
Yes its working great, I have a PBX connected to it along with a couple of direct lines. Works like a charm.

So do you have this running on VOIPo's PBX? and if so how would you set it up with 5 lines, having line#1 as main in-out line and lines 2-5 incoming only. all going to IVR. But have a way to know when a call comes in, you can tell which of the 5 lines it came in on.

Thanks

uf_shane
02-14-2014, 10:40 PM
Well you could monitor 6 extensions without the expansion, but as for lines, you would not be able to monitor the actual lines.

chevyman
02-15-2014, 04:46 AM
Well you could monitor 6 extensions without the expansion, but as for lines, you would not be able to monitor the actual lines.

I'm getting the expansion unit. so the only way would be in the call logs.

chevyman
02-15-2014, 04:53 PM
uf_shane

do know what to program if a line is setup as a regular sip with voipo number, to get the * codes to go through. If i dial like *21 i get a busy signal but right after that if i push the green tel button it calls and i get a "thank You".

Thanks