Now that they say this will be available around this Friday does anyone know where I could download a copy of the install manuals if they are out anywhere yet. I've also gotten the admin software from the ftp site, however I may have missed something but where can I find the username and password to get into this software to play around a little bit. Any info would be great thanks.

This product will be discussed at our conference in Las Vegas on Saturday. We'll be sure to share any information that we gather at that time.

Unfortunately, Vertical was unable to attend, so any information that we obtain will be provided through distributors who will be in attendance.

We will be attending a road show for this product on 3/4 locally, so again, if there's any information that is gained, we will let you know.

Good to know thanks.

The manuals have now been posted on the Vertical CCC site.

-Larry

Larry, they're not on the Comical side of the site. I'd like a peek at what's to come if you can send me the pdf to put in the commode area of the office. Thanks markrts @ comcast. net

Mark

Mark..I just found them. CCC then tech manuals.

Damn I looked there, tech support / tech docs - I'll have to go look again.

Found it all - got it all - It under product overview, not tech support.
Mark

Anyone install one of these yet? One of our techs took the certs on Vertical U and attended the training/recruitment class last week, now we just have to get one.. probably going to order the 4001-20 kit and bang on it in the shop to see what shakes loose.

matt

I know that Rather Be Fishing, a member on this forum has received one and was planning on tinkering with it last week.

Ours came in yesterday. So far I like it.

Does anyone know if there are any restrictions on the number of voip phones,other than maximum 32 users. In other word can we set up a 4 port voip card with the four channels, plug in an ethernet switch and go with say eight voip phones. I have a customer that wants to dump his Telenium system.

16 max IP sets with 8 working at the same time. The 16 IP sets do not take away from the 32 TDM ports.

Thanks for the reply, RATHER BE FISHING. I saw the info. in the manual after your reply. I am still a little concerned about the number plan. It looks as though we can just assign the voip extensions to 132 thru 147. On page 1-2 it talks about the usable station range only being 100-131, hence the confusion. Any thoughts would be appreciated.
tk

I've got the VoIP card installed in a 3x8 system. My Nomad SP came up as x108 and my VoIP phone came up as x109. I have not tried to change the extensions.

So far I like the system, but the voice-mail is quite limited. I'm sure that will change with the new units coming out.

Thanks for the reply. I'd like to know what happens when you change extension numbers. If "RBF" is right then we have a system that can theoretically handle 46 extensions.
tk

If I remember correctly from the webinar the voicemail will be getting some added ommph on next quarters release.

unified messaging in the 4th qtr., I can't wait that long to try it out though.

Anyone here know the exacts on the SIP trunking limitations on the 320? For example if I create a trunk to another site, how many concurrent calls can be sent over that trunk. Or does the VoIP card(s) only allow the max of 4 or 8(dependant on the additional card) concurrent VoIP Connections over that trunk?

tia

crees

A trunk is a trunk, as in one call at a time, no matter what type it is and which system is being used.

The number of simultaneous calls over the VoIP card is also dictated by the number of circuits that are equipped.

In general, it is a 1:1 ratio no matter how you cut it.

Thanks for the clarification.

Not according to Vertical. They are saying that you can have numerous calls on one VoIP channel if you're using SIP trunks. They have specifically said that you'll need one VoIP channel for a SIP provider and it will be able to support numerous calls.

At this point in my research I cannot say that this is true. They have not come out with any cut sheets for SIP providers to test with.

Not sure if the same holds true for system networking.

NFC just to expand on what you said, Vertical says you can have multiple calls on one sip trunk however networking and IP phones are strictly 1:1 ratio.

That is my understanding but like everyone else I cannot verify this as I have not had one out to test yet.

Freddyg, that is correct. The SIP trunk would have a trunk with Multiple DID's on it. and the phones and netwrking are 1:1 ratio.

I got the system last week and it seems pretty good. The voicemail needs some help though. I haven't tried the voip but I've got a phone and card on order.

Does anyone know how to automatically turn off the message light after listening to the message. It seems the message light stays on until you delete the message. I guess I can have the customer turn off the light using the feature code but that's kind of mickey mouse.

The other strange thing I've found is a setting to turn on or off the "enblock". I had no idea what that was so I turned it on. Then I found the live keyboard stopped working. Turned enblock off and the live keyboard worked again. It needs a better name like hot keys or live keys, not enblock.

I haven't tried to access messages while out of the office, that may be a challenge. Anyone know how?

well, our SBX should be here in the next couple days and the Wave shortly thereafter. unfortunately it'll probably be a couple weeks before i get a chance to give them much attention.

as far as the voicemail on the SBX goes, they told us at training that they knew it needed improvement (and more recording time!) and that new boards would be available before too long, but probably at a slightly higher cost.

As I am reviewing the installation manual, I noticed that it appears that digital stations require 2 pair. Is there someone that can confirm this? I am expecting my unit next week and was just trying to plan ahead.

For some reason, I thought it was explained to us that there would be one pair for analog, and one pair for digital, but that was at least a year ago.

It is a 2 pair system. All ports are hybrid except port 1...similar to the Partner system.

That is what I was afraid of. Thank you for your help.

The phones only take one pair. BUT, the digital phone takes the 2nd position on the jack and the analog takes the first. Works like an Intertel system.

Which means we may have to rewire some jacks that were only wired for one pair...

But don't the phones have an aux. port for the analog extension like Panasonic XDP's do?

Basically, it's just a matter of cross-connecting the second pair through to the digital station and using the paired analog station for whatever you feel like, correct?

Pointing out the rewiring of jacks is a good point Larry. As much as I hate it when people do that, it goes on all the time.

No. The SBX phones don't have the ADP jack. While I have not tried it, I believe that you cannot use the analog and digital of a port at the same time.

The analog phone works fine but shuts off as soon as you plug in the "paired" digital phone.

If the analog phone is talking to another station, as soon as you plug in the digital phone the call is put on hold. When you hang up the analog phone it rings back with the call. The digital phone doesn't sync up and it stays as an analog phone until you unplug and plug in the digital phone again.

It would be nice if they added paired ports to the software n the future.

Important:

When setting up the Custom Calling Tables you NEED to turn on DISA (on/off) in program 140 or the system won't accept any digits.

This seems to be only if you send the call directly to the CCR tree. If you send it to a phone and have the DISA ticked the CCR picks up right away. If I untick the DISA then the call rings to the station and forwards but I get the station 100 message. I'm going to try to figure out how to get the CCR greeting.

that's good to know! thanks..

Ok new question, I must be doing something simple but wrong. I have been trying to connect to the sbx with the admin and it will connect then shortly after tell me system disconnected abnormal disconnect. So what am I doing wrong. I've been going by the settings in the pdf files.

I just installed my second one! Had no problems connecting admin with IP. I'm still figuring out the programming!

Serial or IP. Hooked up my third today and had no issues with a crossover ethernet cable.

Man that programming is a learning curve.

Yep. Very comparable to the Telenium programming. Not WAY off from original Vodavi programming, but far enough to where I am already hearing resistance from the technicians.

We put in one (and have two others on contract), it works great and the customer loves it, but now we are being told that they are on back order for three weeks...is anyone else getting that?

Steve

Yep.

Aha got my connection problem fixed of course it was user error. I was going with serial and one of the settings I didn't see in the manual for connecting so I guessed and was wrong.

I have not had a chance to play with the programming much but do like the interface, of course when I did my vodavi training years ago they didn't even teach us flash it was all windba. I can see there will be a learning curve on this but one thing I don't like is the tone you get when programming from the station the confirmation tone and the error tone. Seems they could have toned that down a little bit.

Reply to billyho's question regarding the message waiting lights:

After you listen to a message, if you hang up the callback light will continue to blink - message saved as new. If after listening to a message you press next, it will save the message (as an old message) and play the next one - the light will go out when you have no more new messages. The key is to press next after listening to the message.

It would have been a lot more intuitive if they had labelled it save instead of next.

New Question

If an extension is busy and you dial it from the vm greeting, it plays an announcement that says "that extension is busy & to try another extension." I should note this is the experience in night mode (with DISA on). In day mode (phone calls delay ring to vm - DISA off), when you dial a busy extension you hear a busy signal and are disconnected.

I should note that Auto Fwd VMIB is on. When the phone rings and is not busy, the call will go to their mailbox.

Using the forward button on the phone, I can get a workaround. But then the light blinks on the phone and you lose the ability to put the phone in handsfree mode.

Any suggestions?

I've installed 2 so far and they seemed to go in fairly well. Single pair to a digital set is great, much better then 2 pair like with a Avaya Partner or Panasonic.

Tech support isn't up to speed on the system yet but one tech sent me all the cheat sheets etc, which was great.

I wish 2 way recording was turned on by default and the flash time seems to be off a bit with the default setup.

If there was a simple way to change COS (local, LD, internal) it would work well for a small hotel/motel system.

We've had a system in our office for the last couple of days and for the life of me, we can't get the IP phone to see the MFIM. I can ping all the devices fine, but the phone never sees the MFIM. Any ideas?

Thanks in advance for any replies.

I got the system working on the local network fairly easily. The tough part was getting a remote connection.

The system needs a one to one NAT router, not port forwarding, or putting it in the DMZ. It actually needs an extra IP address from the service provider in order to work. Most cheap routers can't grab two IP's so I had to improvise.

The Cable company here doesn't do static IP's unless you pay for it and I needed a connection in a hurry so I changed the MAC address on the customers router to the same MAC address as the VOIB card and got a DHCP address from the cable company.

Then I put a cheap switch in front of the customers router, and plugged the VOIB card into one port, the router in another, and the feed from the cable modem in another.

I changed the router back to its regular MAC so it got another IP address, and then programmed 340/341 to the IP address, gateway, subnet, and DNS info I had gotten by using the VOIB's MAC address.

Then I was able to get a Nomad soft phone working from anywhere by going into the setup section and putting in the public IP address of the VOIB card and setting the Network adapter section to "Windows Default Adapter". I couldn't make it work externally until I changed the network adapter setting.

I had some echo problems but I turned on H245 tunneling and that seemed to help.

I would have liked it if the system used IP port forwarding like most systems instead of having to have a second IP address but it does work.

The strangest thing with the Nomad soft phone is there doesn't seem to be a way of deleting messages after listening to them. There's no delete key and pressing 3 doesn't work.

Anybody have any suggestions?

HI all... I have run into some documentation flaws for the sbx ip320 mostly on the voicemail setup. it doesnt really give you anything on how to set passwords for mail boxes or default settings or even how admin boxes. i have checked all of the booklets and searched the ccc site for any other additional information on this. my main beef is when you set up password access for the vm it doesnt tell you the default passwords. can anyone direct me to a alternative source for feature programing?

Has anyone installed the new voicemail upgrade (8 port) to the SBX 320 yet? If so will I lose greetings and current messages? Or does that save elsewhere? Thanks in advance for any help...

And ez Phone??

We did a Beta on the bigger voicemail. The voicemail greeting and mailbox greetings are stored in the database programming so as long as you do a db backup you should be fine. There is a VMIB prompt file that you will have to load for the new voicemail.

And we have used the EZ phone and it is pretty Slick.

Just a side note for you all if you activate demo mode it will run out in 60 days but if you set the date and time back to when you started Demo mode it will still work in that 60 day period!!!! Found this out when we were to demo EZ phone for a client and the Demo mode had expired so I took a shot in the dark and set the date and time way back and it worked.

I have had an issue where I am randomly losing packets on the VOIP card. Wont accept calls for brief periods of time. Also, if I pick up a phone it will dial. very strange.

If you go onto www.sbxip320.com there is now a new manual under 2.0 and it is much better. Documentation was weak due to quick release of product in the US since it was in Eurpoe already.