CS410 Issues

[Bradley_M]

Ok, about once a day, or perhaps every other day . . it's totally random, a person is on a call and the system will just drop them. That's issue#1. I don't know if this is related also, but apparently the system will forget FX0 ports and leave them hanging open. I went in and tried to call and got a busy signal (main line with extra line roll-over). I had the person at facility look and all FX ports were lit up . . . so I looked at the onscreen call status . . . it only showed two calls. Hmm. I ended up rebooting the appliance to get the calls to drop. I've attempted to contact my VAR - Digital Communications out of Bakersfield, CA and so far have not received any calls back. I sent an email to Kevin Moroz and his suggestion was contacting abptech and they said they hadn't hear of any thing like I was describing, but offered a $120/hour support contract. I replaced a Asterisk system with this CS410 and I'm really beginning to regret that decision . . . they were having operational issues with the Asterisk system, but at least calls weren't being dropped.

 

What can I go to resolve these issues? I'm in the middle of evaluating this product for another identical office, as well as our corporate system, but unless I have support and a rock solid system I can't proceed. Any assistance would be greatly appreciated. Here's our stats:

 

Snom 360 Phones - All on version 7.1.30

CS-410 (Black) Version: 3.0.0.2905 (Linux)

 

Four POTS lines in with rollover from main DID.

[Vodia PBX]

Calls dropping are usually a problem with the hangup detection. Unfortunately, FXO is not very clear about when to hang up a call. In an extreme case, an operator might play the message "The other side has disconnected the call, now it is time for you to hang up" and wait until the PBX disconnects the line.

 

There are three ways in the CS410 to detect call disconnect:

 

Detect Busy Tone: The PBX tries to detect a busy tone on the line. If the other party plays a busy tone (or something that is similar), well then the PBX things oh that's my time to disconect the call. If your operator does not play busy tone, then shut this off.

 

Detect Dial Tone: Similar, but with dial tone. In doubt, turn this off.

 

Detect Polarity Change: That's another way. The idea is to change the polarity of the analog signal when the call gets connected and when the call gets disconnected. This can be dangerous if you have a long line and the detection is on the edge.

 

If you know how the operator indicates the call disconnect, then you should use only that method.

 

BTW the latest and greatest is http://www.pbxnsip.com/cs410/[color=&qu...update-2914.tgz[/color]

[Bradley_M]

I just upgraded everything about an hour ago . . . DBT is off, DDT is off, DPC is on. Should I change polarity and see what that does? My biggest issue is when facility calls in and is getting busy signal first. I definitely need those lines to be reflected properly. Why do the lines show hot via the LED's but not on the web interface?

 

 

Calls dropping are usually a problem with the hangup detection. Unfortunately, FXO is not very clear about when to hang up a call. In an extreme case, an operator might play the message "The other side has disconnected the call, now it is time for you to hang up" and wait until the PBX disconnects the line.

 

There are three ways in the CS410 to detect call disconnect:

 

Detect Busy Tone: The PBX tries to detect a busy tone on the line. If the other party plays a busy tone (or something that is similar), well then the PBX things oh that's my time to disconect the call. If your operator does not play busy tone, then shut this off.

 

Detect Dial Tone: Similar, but with dial tone. In doubt, turn this off.

 

Detect Polarity Change: That's another way. The idea is to change the polarity of the analog signal when the call gets connected and when the call gets disconnected. This can be dangerous if you have a long line and the detection is on the edge.

 

If you know how the operator indicates the call disconnect, then you should use only that method.

 

BTW the latest and greatest is http://www.pbxnsip.com/cs410/[color=&am...update-2914.tgz[/color]

[Vodia PBX]

Why do the lines show hot via the LED's but not on the web interface?

 

Well, the "CO lines" on the web are far away from the real hardware. If the lines are still on, the FXO gateway obviously believes that the call is still on. Anything in the log with PSTN? You can turn logging for PSTN events on (requires a reboot, unfortunately).

[Bradley_M]

Well, the "CO lines" on the web are far away from the real hardware. If the lines are still on, the FXO gateway obviously believes that the call is still on. Anything in the log with PSTN? You can turn logging for PSTN events on (requires a reboot, unfortunately).

 

 

Ok -- I turned on PSTN logging and will see what that turns up. What level logging should I be set to? I'm at level 5 right now.

[kellenw]

Ok -- I turned on PSTN logging and will see what that turns up. What level logging should I be set to? I'm at level 5 right now.

 

Bradley_M works for my company, so I thought I'd update where we are at with this issue. We are still having this problem, and it looks like we are going to have to pull the plug on trying to use pbxnsip/cs410. It's just not "working", and it is costing us huge amounts of money every day in lost business, frustrated customers and travel and lodging expenses for our IT people when they have to go to this branch office to try to "fix" the cs410. The VAR support has been non-existent, bottom line. They won't return calls or emails. When we ask for help on the technical issues of this problem from pbxnsip directly, we are told to talk to the var, pay for support or post to the forums. The forums have yielded zero help on this issue (here we are a week since Bradley_M's last post in this thread asking for the correct logging level and no reply), the var doesn't want to support the sale, and quite frankly, we aren't going to pay for support for a problem that from my perspective appears to be product related. We already paid for the product, and it hasn't worked correctly from day one. Incidentally, we had SBC/ATT do a line check, and they claim all lines check out fine.

 

This is not free open source software. We paid for it. We expect it to be supported. We expect it to work.

 

We were sold on this product, and we were planning on rolling it out enterprise wide. That would have equated to a lot more money for PBXNSIP. We are a two time Inc 500 company (top 500 fastest growing privately held companies in the US). We have plans for continued expansion. That would have meant MORE money for PBXNSIP.

 

As your own sales literature says.... "I want a telephone system that just works".

 

Will you please help us? I'm sincerely sorry if I came off sounding a bit harsh, but this has been an incredibly frustrating experience so far, and I would LOVE for it to get worked out, but I need that to happen right now. I just can't afford to wait any longer.

[andrewgroup]

(I didn't see any subsequent posts with the PSTN logging)

 

Why is it the manufacturer of a system gets all the grief? try to call Microsoft and make the claims you're allowed to make in a public forum about a VoIP product. CS410's, Shoretel, Comdail, Asterisks Source or distros, all require a level of skill and experience that isn't common. PSTN POTS lines have the most variables and getting ANY Analog Gateway properly configured may be difficult. However, not knowing the health and condition of the lines themselves add a layer of difficulty that only compounds the troubles.

 

Anyone attempting to get into the VoIP telecom equipment business really - really needs to invest in the correct tools to properly diagnose POTS lines. Knowing test-tone tone lines on switches, pair lengths are all common knowledge of traditional telecom technicians.

 

Problems on the PSTN side - Problems on the VoIP side can all contribute to troubles.

 

We have several remote sites +150 miles, as a courtesy I just did a status snapshot and the report is below.

 

Version: 2.1.6.2448 (Linux)

License Status: Appliance Key

License Duration: Permanent

Additional license information: Extensions: 10/32 Accounts: 22/40

Working Directory: /pbx

IP Addresses: eth2 192.168.1.99 192.168.1.0 255.255.255.0

eth1 1.1.1.1 1.1.1.0 255.255.255.0

lo 127.0.0.1 127.0.0.0 255.0.0.0

default 192.168.1.99

 

MAC Addresses: 00191568404A 00191568404B

Calls: 74/263 (CDR: 77) 0/0 Calls

SIP packet statistics: Tx: 375882 Rx: 375935

Emails: Successful sent: 101 Unsuccessful attempts: 0

Uptime: 47 14:15:11 (4876 5302720-0) WAV cache: 0

Media CPU Usage: 100%

 

We worked long and to learn the CS410 and PBXnSIP in general before we started installing them. Attended PBXnSIP training, paid for support on issues we couldn't resolve and invested in tools and technology to assure our installations go as planned.

 

For those of use that do likewise, the opportunities are only going to increase as many older / TDM telecom interconnects will never - never make the transition to IP based systems. The owners will never hire the right staffers to make this a go.

 

I know I didn't give you the free answered that you feel are owed.

 

TIP #1 The only allowable drop for POTS line is the DMARC directly to the CS410. I can't tell you how many DMARC's have multiple drops in a building. (It's like antenna's going everywhere creating transient noise on lines)

 

Having a high empendance line audio recorder should be in every technician tool box. These are the analog equivalent of Etherreal Packet Sniffers. Just gotta have-em or you spend a lot of time guessing and cussing.

 

Hope you get these resolved -

[Bradley_M]

Anyone attempting to get into the VoIP telecom equipment business really - really needs to invest in the correct tools to properly diagnose POTS lines. Knowing test-tone tone lines on switches, pair lengths are all common knowledge of traditional telecom technicians.

 

First off -- this is a device a RESELLER is selling and is being touted as a plug and play (nor plug and pray) device. Show me any info on the wiki that or product info that details out this stuff and I'll be happy to read and research. In the meantime, products such as this without proper support and without proper VAR channels are a risk. Someone is letting them out in the field and not calling the shots on what is being sold. Regardless of the finger pointing episode here, I'm more than anxious to get our install working. We're the end user . . . and the appeal of this product is the ease of use to install.

 

Problems on the PSTN side - Problems on the VoIP side can all contribute to troubles.

I know I didn't give you the free answered that you feel are owed.

 

SBC did line voltage checks, impedance, the whole gamut of checks . . . so nothing to be found there . . . ???

 

TIP #1 The only allowable drop for POTS line is the DMARC directly to the CS410. I can't tell you how many DMARC's have multiple drops in a building. (It's like antenna's going everywhere creating transient noise on lines)

 

The DMARC is coming in on a fresh install in a new office and goes right from the punch down straight into a RJ11 and into the phone system. It doesn't go any further.

 

Having a high empendance line audio recorder should be in every technician tool box. These are the analog equivalent of Etherreal Packet Sniffers. Just gotta have-em or you spend a lot of time guessing and cussing.

 

Which again, would be a nice tech support/wiki/etc... addition, but from reading other forum posts, and getting info directly FROM pbxnsip -- they don't have this info. Don't stand and defend a product that the manufacturer is openly saying "yea, we've got improvements to make on our end".

 

 

Hope you get these resolved -

 

One way or another they will be solved, whether at the expense of installing another phone system, or doing whatever it takes -- but the attitude check should be left at the door. I've had quite enough experiences with Microsoft products and their attitudes to lead me to my Mac and my Linux based laptop. If your wanting to lump this product in the same vein as MS go right ahead.

[andrewgroup]

Let me see if I got this right, you are the end user, a reseller sold you a CS410 for your application making claims reminiscent of USB Plug and Play, Right?

 

If so, after 35 years in the IT support industry, (IBM 360 Mainframe and PIG Iron Card Punch) through today, I've yet to experience a Plug and Play anything on any operating system that worked or would work under any circumstances. It simply doesn't happen out of the box like that. Plug and PLay only works, and it does, when the installer knows how to setup the environment to allow it to work. This includes specified information on DHCP, VLAN's if USED, Button Settings on phones for SLA's, and more. If you begin your search in the WIKI for how-tos of how all of this works, I believe you will find the helpful friendly advice that you seek. But to come to a forum after somethings been installed and is failing and expecting unlimited post installation diagnostics and troubleshooting regardless of begin an end user or the reseller here or in any Vendor supported Forum you are living on another planet.

 

I do not speak for PBXnSIP, but I searched the WIKI for all of Bradley_M's posts and all posts are of the nature of help me fix my problems vs. HOW do I do this?

 

I've managed hundreds of Support technicians in the last 30 years, and the most frustrating personality trait most (NOT ALL) is a gun slinger mentality. They go marching off into the woods and anything that looks like a problem they start fixin stuff. The really great ones have a plan - A plan that is a straight line between point A and point B allowing concise and precise implementations and any deviation is no more than 1 step in a tangent direction allowing quick recovery.

 

Our clients during the last 5 years of VoIP business think their phone system is a SNOM system, only because they've never asked. We sold and supported a telephone system, not a product. We could replace PBXnSIP with any backend VoIP system, just as we replaced Asterisk with PBXnSIP. The minute the End-User becomes aware of or concerned with the Product name in Utility based services like Telephones, A/C Power.

 

 

There is hardly anything in the world that some man cannot make a little worse and sell a little cheaper. People who consider price only are this man’s lawful prey.

 

It is unwise to pay too much, but it is even worse to pay too little. If you pay too much, you lose some money, that is all. If you pay too little, however, you will sometimes lose everything, as the thing you bought cannot do the intended job.

 

The law of economy forbids to obtain something of high value for little money. If you accept the lowest bid, you must add something for the risk taken by you. And if you do so, you have enough money to pay for something of higher value.

 

John Ruskin (1819-1900)

[Bradley_M]

From the latest eBay sale of the pbxnsip system:

 

"Item Description

 

PBXNSIP CS410. Appliance Single-Domain, 10 extensions and 10 accounts (Choose 10 of any of the following to meet your needs: Auto Atendant, Conference, Hunt Group, Agent Group, Calling Card, Paging, Service Flag, and IVR Node).

Fully SIP compliant, built-in 4 port FXO gateway for PSTN connectivity.

Now with additional WAN port.

 

It is a perfect solution for small to medium sized business’s (SMB) that want to take advantage of VoIP’s benefits and have the option to keep their existing phone lines. The pbxnsip CS 410 contains 4 integrated FXO ports to connect to existing PSTN trunks and the system also supports IP trunks to connect to Internet Telephony Service Providers that support SIP. The system includes all available features from the other pbxnsip PBX editions like voicemail, auto attendant or conferencing, but also advanced features like NAT Transversal, call barge in or cell phone integration.

 

Key features of the CS 410 include:

 

* Built-in 4 port FXO PSTN gateway and SIP trunking

* Paging and "Music on Hold" audio jacks

* Full PBX features including Auto Attendant, Conference Server, and Voicemail

* Unified Messaging integration with Microsoft Exchange 2007 UM

* Plug and Play phone support and full security support with TLS and SRTP

* Quiet, compact hardware design utilizing a Mindspeed Comcerto VoIP processor

* Easily upgradable to 25 extensions

 

The pbxnsip CS 410 is opening new markets for Value Added Resellers and Systems Integrators in the US and abroad. "Our company delivers high end VoIP solutions to Internet Telephony Service Providers in the UK and US", said Jonathan Greenwood, founder and CEO of Solutions11 Limited located in Aspect Court, Leeds, UK. "With the introduction of the CS 410, we now have an IP phone system that we can deliver to offices of 10 to 25 employees. We can integrate the system with IP phones from leading vendors like Polycom, snom, and Cisco and provide a plug and play solution. This opens up a brand new market segment for us."

 

The pbxnsip CS 410 is shipping now and is available through pbxnsip channel partners in the US and throughout the world."

 

Let me re-iterate, this says "plug and play solution" and I could really care less what your years of outdated experience indicate, today more and more solutions are based on very easy to integrate products that should be easy to diagnose and work with. If I fill my car with 91 octane fuel, or 87 octane, the engine nowadays is brilliant enough to listen for knock and retard the timing based on what it knows is going on. Ten+ years ago, that was around too, but didn't work right all the time . . . so nobody relied on that for consumers. This is 2008, the telephone system has been a known quantity for at least a few years.

 

You want questions? Let's ask a few questions:

 

1) How can a device show FX0 ports as active, yet the web interface shows no calls? How do you resolve such a condition? Please don't tell me "reboot the device" in a production environment. The last thing I need is phones ringing open while the system comes back up, or the calls get routed to a voice mail box that will never be checked because the phones are still registering.

 

2) How can you find adequate logging information on this system? -- one of the questions (still unanswered Mr. Gunslinger) is what level of logging is required.

 

3) If this product isn't for general consumption -- why do some resellers have product to push on eBay? Most products with exclusivity to "installers only" have provisions in their sale to say only VAR installations are support in terms of warranty and support. My product(s) came with a couple of sheets of 8x11 pages printed from the wiki for installation . . and that was it. And that's what was provided by pbxnsip!

[andrewgroup]

No Disrespect, but I feel like I'm spoon feeding my children, but anyone buying anything based upon Marketing Material and that alone is a.......fill in the blanks

 

 

You want questions? Let's ask a few questions:

 

1) How can a device show FX0 ports as active, yet the web interface shows no calls?

 

THE FXO GATEWAY is a stand alone device and the call disconnected and the LEC didn't send disconnect tones.

Why? Pigeons on the line is the old guess, but proving the failure is the result of the POTS line requires tools and you don't own them. Or adjustments in the FXO and you don't know how to do that either.

 

How do you resolve such a condition?

Place a Call to SBC and get a service ticket issued, and don't stop until you speak to teir2 and get the status call back number and speak to the tech center, be on site with your tools and assist the technian to prove the line quality with some testing.

 

Please don't tell me "reboot the device" in a production environment. (Reboots aren't the answer, strong troubleshooting skills will always prevail)

 

 

The last thing I need is phones ringing open while the system comes back up, or the calls get routed to a voice mail box that will never be checked because the phones are still registering.

 

The LEC dropped the call without sending call disconnect sequences. Buy a tool to record these sequences

 

2) How can you find adequate logging information on this system? -- one of the questions (still unanswered Mr. Gunslinger) is what level of logging is required.

Pick one and post the results ZERO may not be enough and 9 may be too much, so do something to save yourself and learn.

 

 

3) If this product isn't for general consumption -- why do some resellers have product to push on eBay?

Who says this or any other VoIP telephone system is for general consumption. For the last 100 years in the PSTN telephone business I know no business that bought and installed their own phone system.

 

Most products with exclusivity to "installers only" have provisions in their sale to say only VAR installations are support in terms of warranty and support. My product(s) came with a couple of sheets of 8x11 pages printed from the wiki for installation . . and that was it. And that's what was provided by pbxnsip!

 

The Sherman Anti-Trust acts would prevent any vendor from being able to make and enforce any such claims. If I can buy it through any channel, I can resell it an make any claims I wish. Those claims are not passed to the manufacturer, so any EBAY seller is free to copy - distort anything.

 

If you bought a Business Telephone system from EBAY - My guess is you are getting what you paid for and those of us that contribute to this forum, including the makers of the product owe you little or nothing.

 

THAT's The reality and you can slam the makers of something you bought off ebay all day but that isn't helping your situation.

 

At this point, I'd repost the CS410 back on EBAY and cut your losses and hire a local Interconnect to come install you a phone system with a warranty and committment to make it all work.

 

Done..

So - Microsoft Provides a CDROM to install Windows Server and that's it. Go read the EULA from Microsoft, The ONLY agreement they are stand behind is the distribution MEDIA is good.

[Bradley_M]

Alright, I've figured out what I need to do here is just keep asking the right questions and ignore the people not being beneficial on actual support issues.

 

I have someone from pbxnsip that I'm talking to now and working to see if we can replicate problems. I did discover a new little gem in the system . . . if you are keying touch tones from a phone and the other side of the call disconnects, the channel will hang open. Th call never terminates, and no timeout ever occurs on the line either. I don't know what that problem is . . .but it's been handed off to pbxnsip to see if they can replicate.

 

Another question I have is how the system decides if all channels are full? If the PSTN side never releases, I haven't figured out yet if the system will allow someone to dial out again, or if it will give service unavailable message. If it gives service unavailable message, then why wouldn't the CS410 be able to determine "hmm -- I show no calls online over here on the web side . . . what's up?"

[andrewgroup]

Alright, I've figured out what I need to do here is just keep asking the right questions and ignore the people not being beneficial on actual support issues.

 

Agreed, I to spoke to folks at PBXnSIP about this on your behalf. PBXnSIP certainly doesn't want an EBAY'er overselling or overpromising. However, in a free market, people can do mostly as they please.

 

I'm happy to contribute our experiences as it strengthens or knowledge. The reality is Analog Lines are not all created equal, Not now and never have been.

 

Tools like http://www.vconsole.com/4-Port-Analog-Phon...(FXS)-p-19.html are a GodSend to professional installation companies.

 

Other good arrows to have in your quiver are;

url=http://www.sandman.com/loop.html#LongLoopAdapter

http://www.sandman.com/loop.html#LongLoopAdapter

 

Again, while the goal is to fix a problem, it's far better to fully understand the problem and to know in the future how to prevent things that will directly impact paying customers. (You have to charge a lot too, but that's better than a refund - and we have all done this.)