sundance-communications.com/forum Forums General General Telephone & Systems Information VONAGE LINES

I think this was stated before, but I can't seen to find where. Does Vonage have loop supervision disconnect. If not, what voip company does? I have an NEC DS1000 and whenever an automated recording calls in it always busies out the port. It sucks that we always have to restart the system to clear it up.

------------------
| Telephone Systems | Voice & Data Cabling

I asked the same question a while back, Most said it will work. I called Vonage and they told me that it would not work. Couldn’t get an exact answer as to why. I assumed the disconnect signal was the reason. But they did say that they are coming out with a module that was specifically designed for KSU and PBX type equipment. Groundstart claims that it works on DS1000/2000 and also states this on his website. Although I would not advertise the DS as VOIP capable! Mitch, no offense. I would like to try myself.

Well I do know on the avaya voice mails it works BUT, at the end of a message you get a busy before it actually hangs up which is very annoying.

Just as a side note, I’ve had a couple of Intramail’s lock up with regular POTS lines. The NEC tech tip 65 about this really does not help (supposed work around). A new revision level will soon be available on the website. I have a version 3.21.06 that will fix the Intramail lockup. Tech support sent some techs 3.21.02, which is really screwy. Another thing that may help is disabling the NCRMB option so the AA doesn’t just Loop! And make the timeout go to mailbox without NCRMB so that the AA will hang up. I assume it works fine with your POTS lines, just not Vonage. Something to keep in mind.

One more thing while I'm on a roll. Instead of resetting the system by means of power down. Use the forced trunk disconnect!!

The only line that gets hung is the vonage line. There is no way I would recommend vonage to any of my phone system clients.
I do have the latest version 3.21.06. I just can't beleive that vonage wouldn't send a disconnect signal. How can they offer to businesses and not have everything working properly.

------------------
| Telephone Systems | Voice & Data Cabling

I think they expect you to use their voicemail not yours.

Even for businesses?

------------------
| Telephone Systems | Voice & Data Cabling

The problem you are having is that by default the Linksys adapters do not send a CPC disconnect signal.

CPC is a reduction of voltage to less than 1 volt for a specified duration after the remote party hangs up. It can also be a polarity reversal instead. The US standard is a 200ms voltage drop. The reason for the signal is to notify hold buttons and PBX's that the remote party has hung up.

On the Linksys boxes, that are of course actually Sipura boxes, the default settings for CPC are a 2 second delay and a 0 second duration. It's that 0 second duration that is causing your problem.

This can be changed, but only by Vonage on the Linksys box since you are locked out of the admin functions. The problem is that no one you talk to is going to have a clue what you are talking about, so you will need to be very specific and very persistent.

First of all, you might avoid using the term "PBX" since that does give the tech support person a good reason not to help as they don't support PBX's. I would suggest that you explain that the hold buttons on your phones will keep the line off-hook after the remote party disconnects. That, by the way, is a true statement so long as you have phones with hold buttons.

Next explain that you absolutely know how to fix this. Tell them that there is a parameter in the Linksys called "CPC Duration". Tell them you would like them to set this to .2 (that's a decimal then "2" for 2 10th's of a second or 200ms). With that level of specificity, anyone at tech support should be able to do it for you. I would also ask them to note what they did in your account and that it solved your problem so that when you add more adapters in the future, you can just have the next tech read the notes.

I've done this on 3 linksys adpaters successfully so you should have no problem. By the way, I did use the "PBX" word and it was almost a deal killer at the start. Fortunately I was having issues with hold buttons at another location, I knew exactly what the problem was and how to fix it, and I got a very helpful tech who was willing to read the articles I pointed him to so he did the deed. Once he had the notes in the account, getting the next ones done was a breeze.


[This message has been edited by gkar (edited March 22, 2005).]

I have a customer using a Sipura SPA2100. We set the CPC setting up to work on a Norstar, and normal CO calls Disconnect. I can monitor lines and see disconect signal.

The problem we have is with a trunk to trunk conference (unsupervised conference) not releasing call after either party hanks up. If you monitor lines you get reorder tone from both lines, then goto silence. You can unplug one line and both drop. trying to figure out the callee/caller polarity setting to force disconnect. Or is there a timer parameter to disconnect line if they get reorder tone, or dead silence for a period of time.

Thanks,
Jeff Jackman (JJ)

Does this only an issue with the Linksys router?

How about the Cisco or Motorola routers?

I just ordered vonage and they have listed to send a linksys adapter. How do you get them to send you one of the other adapter?

I don't know how they choose which adapter they are sending you?

Is this an issue that mainly effect voicemail or does this effect using the lines on a phone system completely?

Does anyone have reference to the article needed to convince vonage to change the disconnect settings?

Is the CPC changed on the actual physical adapter on your side or some piece of equipment on vonage's side?

Thanks.


[This message has been edited by Fisher (edited April 18, 2005).]

I called them and did what gkar suggested and got hem to change it, all went well.

gkar, Good work! I had Vonage with a linksys router that I couldn't get a CPC signal from. I spent some time with Vonage tech support educating them on CPC control, why it's needed and other names it goes by. I unfortunatly used the words PBX and Voice mail which promptly stopped all thinking on their part. I was told that they simply did not support this feature. I ended up canceling the service and have not suggested it to any customers as an alternative for fax's or P-lines.

I know they are having issues with the 911 service as far as the PSAP is concerned if you get an NPANXX out of your area. And I’m still not willing to recommend it as a primary service. But, I think I’ll give it another shot and change one of my Verizon lines since Verizon kind of sucks any way.

Why can't Vonage just have the right CPC setting as a default?

It sounds like so many problems would be solved if they changed that duration time from a "0" to a ".2" right?

Does anyone know exactly how to explain where to look for this feature or know how to get into the router settings yourself?

Does this issue lock the trunk up for ALL phone calls or just when dealing with VOICEMAIL?


Thanks.

Fisher, Fisher, Fisher, to do it right the first time, would go against all common sense.

gkar,
Do you know the model linksys? I saw a couple of them, a little silver one that has phone ports only and a normal looking blue linksys router type.

Sorry for being so technical!

Hey Guys.

Is the disconnect signal only an issue with Voicemail or all phone calls?

For example: If I were to place a regular phone call on a phone system to a person and then I hung up the phone.

Would the phone system(not voicemail) be able to recognize the line as being disconnected?

I understand that the voicemail need to be able to see the disconnect signal to drop off the call. However on a regular phone call, both parties hang up the phone.

Does anyone here know or does it depend on the system?

Thanks.

They should make a adaptor that will turn an data T1 into 17 analog trunk for VOIP.

Why buy an expensive fancy VOIP phone system instead of using something on the analog input cards of your system?



[This message has been edited by Fisher (edited April 24, 2005).]

Fisher--if YOU hang up, the system will know for sure. The issue is when the other person hangs up. If you place a call and then the caller hangs up, the system will not drop the call. Instead, the line will hang until you take the call off hold (and noybody will be on the other end. With voicemail, the system will not know to stop recording, so you will get very long periods of scilence in each message.

Fisher,

They do, it's called a media gateway and they are available in any number of port configurations, 2,4,8,16,24 they are available as either FXS or FXO ports or both so depending on your needs you can do whatever you want.

------------------
One stop Voice and Data Solutions VoIP

Thank you. This is one of the best explanations I've heard. I think I finally understand the issue. So its mainly the problem when the OTHER party hangs up in situations when:

1. Inbound calls to a voicemail system.
2. Inbound/outbound calls placed in hold for extended period of time after the person hangs up.

So for a majority of outbound regular calls to actual people, it seem like it would be ok and actually the few problems it does have can be corrected by changing 1 little settings in the adaptor from 0 to .2

Is this something use to transfer call from an outside extension in a branch office?

I was thinking more of of a product that maybe a phone company could offer that you would pay like a flat fee of like $1,000 and they provide you a T1 using VOIP and provide you with unlimited inbound/outbound calls.

IF a data T1 has 1.54 Mbs of data and a good quality VOIP takes up 90K. Then 1 T1 could provide you with 17 trunks.

If this is the case, why spend all the money on IP expensive IP phones when you can convert your regular business phone system into a VOIP system at the source(trunk input)

Fisher,

The headroom and required bandwidth depends on the codecs being used, If you chose to you could use the equivalent of all 24 channels,the gateway would be the handoff to your PBX, VoIP trunks are virtual lines so depending on your provider you could use one number and have it pointed to all the FXS ports and from there connect to your PBX.

------------------
One stop Voice and Data Solutions VoIP

If you had 24 trunks on a T1. That would only leave about 64 kpbs per line. That would be very low quality call won't it?

Are pretty much all VOIP products delivered in the form of analog trunks to a phone system.

I am not sure which system you use but in the Samsung line, if you use a TEPRI(T1 card) you can have the T1 go directly into your phone system and assign virtual (DID) numbers to transfer calls to different extensions instead of pointing the trunk.

This mean that you can have 1 T1 line and 100 different routing location as long as there are no more then 24 simetaneous calls.

I wonder if there is any such routing abilities with a VOIP product with either the ability to:

1. Be delivered in the form of a T1 with DID numbers which are virtual numbers to route the call to each extension.

2. Able to transfer a call to an extention based on the phone number(instead of permanately assigning a number to a trunk, and the trunk to the extension)

Thanks.

gkar,

I got vonage to change the settings to ".2" on all 4 of my vonage lines. I am still having the same problem. When someone calls into my voicemail and then hangs up, I still see an indicator on my system and then it transfers the call to the operator and when the operator picks the phone up they here a fast busy signal.

If its after hours it will just fill up my voicemail boxes.

Any suggestions?

NEC, you may want to review a thred located at https://www.sundance-communications.com/cgi-bin/ultimatebb.cgi?ubb=get_topic;f=4;t=000599

In that situation we were dealing with an Inter-Tel Axxess that was behind an Adtran router. Same type situation as Vonage lines in that the line comes in across broadband and then breaks out into analog ports. In the Adtran there was a setting that allowed us to add come capacitance across the line to change false readings. They were having a problem with the system knowing when it was off hook. It sounds like yours is having a problem knowing when it is on hook.

Please keep us posted. These issues are getting more and more common.

MacGyver,

Sorry I forgot to mention my phone system is the NEC Aspire S.

I was just thinking that regardless of the type system, it could be the settings on the router used by Vonage may be your issue. If it is not interpreting the on and off hook correctly it could perhaps cause your issue. That is what was happening with the Inter-Tel system.

Just a thought.

I was just thinking that regardless of the type system, it could be the settings on the router used by Vonage may be your issue. If it is not interpreting the on and off hook correctly it could perhaps cause your issue. That is what was happening with the Inter-Tel system.

Just a thought.

In the US all T1's are 24 channels x 64k = 1.54 Meg. When integrated circuits are put up here, ie NuVox's FlexLink product www.nuvox.com, they bring in an Adtran to the size of the channels used for Voice. So if 6 voice and the rest data they bring in a Adtran 606, etc.

With VoIP circuits it's different entirely, since the codec usually brings the overhead down at about 20 - 30k per voice call. In this case NuVox has a dynamic allocation circuit product they call VoxIP and unless the circuit needs to be terminated to an analog 66/110 block they use a Cisco router. If it needs to be terminated to an analog punch down they use an Adtran device again.

And no I don't work for them, but have been talking to them recently about their offerings for a couple of clients.

As always, other than the fixed channel Integrated T1 in the first example, ie FlexLink product; mileage will vary depending upon the circuit provider.

As to the rest of your questions, likely. ;-).

Again implementations are specific to the circuit vender and their circuit offerings.

cef1000

From my personal experience with Vonage, I wouldn't be so sure they actually did make the change, or that who you talked to wasn't just "placating" you. These are very undertrained, "wrote" tech support folks, as you can tell by the other info here.

I've had 3 of them tell me my WRTP54G can't have the firmware updated if the inside IP address isn't set to the factory default address (crimeny pete). Worse their the only ones that can update the firmware and it has to be remotely, since they don't want to give you the user login to get to the flash process or let you get at the latest image file (assuming you can get someone not in New Deli).

cef1000

Gee, why not get CPC disconnect boxes from Mike Sandman, at least as an emergency measure?

www.sandman.com 630-980-7710