Sorry, I'm a little late to the party, had a delay with the SIP trunks. Right now, I have it configured purely SIP with NGT trunks, no FXO, no FXS, here are the things I've run into so far. If you need me to, I can split them all up into multiple threads:
RIght out of the box, I started through the wizard, changed the WAN IP to static and submitted changes. After it came back up, it retained the subnet mask and both DNS servers, but the IP addres and default gateway stayed the same as when they were DHCP. Per a previous post, I updated firmware (although I think it was the same image) and performed the factory reset. I went through and set the WAN IP again, this time the settings stuck. May have been a one time thing.
In the wireless configuration, it will not accept special characters for WPA2 key.
I agree with a previous post that PoE should be standard on the box, but as I think about it, I'd really like to keep the unit as inexpensive as possible, so PoE is a big plus, but not if it adds much at all to the cost. Even another $100 is a substantial percentage of the as-is anticipated price. PoE would be great for 1-4 phone deployments, but even then, the power adapters are pretty affordable.
When on a call on the handset, I place the call on hold and then hang up the handset. I then press the RESUME softkey (or flashing line key, whichever is applicable) while the handset is still in the cradle and it picks the call back up, but it uses the handset for audio, not the speakerphone, I actually have to press the speakerphone button to turn it back on even though the call is already live again.
Hold music (from the internal source) is working in internal calls but not when it's across the SIP trunk. Inbound and outbound.
Someone already mentioned it, reboots take about 5 minutes, that's a long time. What I don't think was mentioned is that it seems to have to reboot for any minor changes. I would think there are many features that could just be changed on the fly.
When the internal music source is off, you get tones while you're on hold. That's right, but it only makes the tone three times about 5-10 seconds apart, then it's total silence. Again, that was for internal calls, I don't think there was any sound at all for calls over the SIP trunk.
Dial plan. When a user dials a phone number, he has to wait for the phone to process it, I think it's called the interdigit timeout? Some of us know you can hit the "dial" softkey or the "#" key to tell the phone you're done, but that's not common knowledge at least among end users. For past systems, we've altered the dial plan to speed up the process. It'd be nice to have those options in the GUI much like you see in CCA. To speak in CLI terms, we set the local 10-digit dial to [2-9]..[2-9]......, LD to 1[2-9]..[2-9]......, international to 011.T, which except for the international, speeds things up as opposed to the typically default .T behavior. I tried to go into the phone's GUI directly to edit the dial plan, and I could see where to do it, but it looks like you have to be in the admin page and you get a big "FORBIDDEN!" if you click on the "admin" page. It would be nice to have some OOB capability if it's a partner that's willing and able to tweak it a little like that wants to.
VM. Two new voicemails in the box. I dial in, press "1" to listen. It starts playing the first message, I already heard it once or I just don't want to hear it all the way through but I want to get to "8" for advanced options. No can do. Pressing "8" seems to pause the playback and no other digit worked to speed along the process so I could get to the "8" advanced options. So you have to wait for it to finish. Fine if it's a short one, but if it's a message from my mom, I really don't want to have to wait (and listen) for 4 minutes till the end. I know the regular options are available mid-message, but not the advanced options.
Now, once in the advanced options, I just listened to an internal message (I didn't test it on external calls here) and pressed 8 for advanced and then I press "2" to return the call. It prompts me with a confirmation that I want to call extension 101, I press "1" for yes and then it's dead silence for about 30 seconds, then a fast busy. Didn't work.
During my time inside the VM system, I noticed fairly frequent intermittent skips in audio, not stutters, more like the audio just cut out as in with a bad Internet connection, except this was internally and the phones are connected directly to the UC320.
And last, for now, I didn't see any way to manipulate the outbound caller ID. On the contrary, I found where you set the caller ID globally per SIP trunk group. The scenario is a site may have 10 users with 10 DIDs going directly to each user and they'd want their direct extensions to be sent out in the caller ID. Not always going to be the case, but certainly possible. All the SIP providers I know of accept this manipulation of the calling number, so long as it really is a phone number that's part of your trunk group. Well, I haven't figured out to make it work with NGT, but the others like CBeyond allow it.
As I get some time, I'll try to push the system a little and see what else I can find. Most of the basic and frequently used functionality seems to be inthe GUI already.
