andrewgroup

Is a Manual State Change accessing the WEB PAGE as the domain administrator? Or is a Manual State Change dialing the FLAG and are restrictions placed on which extension can dial the flag? The way I see it dialing the FLAG resets it, but not means other than WEB or SCHEDULE time to set the flag at this revision level?

Younger than 1 month? Simply download and upgrade the latest firmware, not just one that begins with 3.0... Can you explain how the service flag now supports dynamically being changed? Can the business owner call into the system remotely and flip the night flag and can the recieptionist dial the service flag or a special ext and flig the night flag so they can go home early? Thanks

Is this feature in the 3.0 CS410 releases currently available? I checked the WIKI and see no references.

We've tested and used SIP aware routers in single IP based installations. Intertex - Somewhat tough to configure - Advanced settings best done with IPTABLES experience, Support is OK based on Europe Xyzel - All WEB interfaces - A bit more $$ Support is great and accessible based on US 5 Polycom Phones, A Small Business Server hosting Exchange, 7 WEB surfers, 4 on the phone connected to hosted PBX, mail coming and going, plus a remote user using GOTO MY PC (crappy protocol from Citrix) This all resulted in some but not much jitter on calls. This wasn't bad at all considering. But if you want to control every packet, then consider Microtik. It's QOS queuing is likely to be the best. http://www.mikrotik.com/testdocs/ros/3.0/qos/queue.php

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.)

Opening Ports VS. Directing Ports VIA forwards may be the problem. Many Routers have limits on how much SPI data they can track. Forcing the UDP range of ports directly to the PBX is a more reliable means of port redirection. 1 to 1 natting on an extra IP address is even better. Logging SIP and TRUNK events when you dial back in prior to making that call out would be helpful. No log data might indicate the ITSP doesn't know the trunk is alive until it recieves a call. (A wireshark capture goes a long way when you create the service ticket with the ITSP.) Is Teliax configured as a GATEWAY or PROXY?

With any PBX that interfaces with Analog PSTN pots lines there will come a day when you need to diagnose and troubleshoot problems. There is an old saying that says, "He who has the numbers, Wins" and being able to provide telecom providers with facts and figures about their lines will go a long way to satisfying customers. A great source for PSTN - ANALOG line tools is http://www.sandman.com/

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.

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)

(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 -

Any Chance we might see a 3.x feature enhancement list or Beta soon?

So many times when dealing with routed networks we find default gateways, default routes and added routes are often times wrong or not present. I'd begin by putting a 15 day free trial of a softphone on a PC and install Etherreal and get some real traces of the actual registration process. Maybe a good forum to start would be SIP PACKET TRACING DIAGNOSTICS 101 and how various settings with PBXnSIP affect these traces. Doing that as soon as I solve world peace. Cheers,

In topic http://forum.pbxnsip.com/index.php?showtopic=665 a long discussion occurred about a more flexible method for night mode. Let me add that at 8:00am on this Saturday morning the owner of a business opened his business for service outside of the schedule in PBXnSIP and knocked on my home residence to have me turn on the phones. While another employee had these instructions, they were no where to be found. I was half asleep when the door was knocked on the request floored me and made me realize the lack of being able to dynamically set a night service flag is $%^##@^% crazy. Let's get this high on the feature list. (Please)

This all points to more line related issues. Having a high empendance line recorder is often the first line of defense. http://www.999.co.jp/us/telecoderan8us/index.html http://www.elyssacorp.com/PDF_Cutsheets/Vo...ogCut_05_04.pdf PLus google searching will find many more - The betters ones can detect / decode DTMF tones while dialing and analyze wav forms Despite the fact that PSTN is over 100 years old, it has troubles and installing new technology onto know crap makes you the goat. Having the most and the best diagnostic evidence allows you to stand clear of the debris' and hopefully to be paid for your expertise.... Add clauses to your purchase agreements that you invoice for all time - including time diagnosing pre-existing conditions with Telecom Providers. Cheers.

Can you monitor a line with a linemans Butt set and listen to the line when the hangup occurs? Until you can validate the LEC is providing call disconnect you are likely spinning your wheels. Call disconnect on lines is a switch setting in the LEC switch and was not always a default settting. The 410 has FXO settings to assistn but they are no substitute for no call disconnect. We have also seen the following trouble. The carrier is providing FXS ports from a CISCO IAD 2431 and when the call is disconnected the CS410 see's these tones and immediately goes off hook and must time out in order to accept another call on that channel. The CS410 FXO gateway needs a timer setting for a wait timer before it answers the next incoming call, this would prevent this problem. I can duplicate this trouble if PBXnSIP might be interested in this issue. I'm sure it can happen with any IAD CISCO 2431 device.

We previously support many Asterisk Solutions with a full time Asterisk Developer. We were able to plug-n-play PBXnSIP into the Asterisk clients without missing a beat and our support time fell and we exclusively used Digium Hardware. Our largest installatiion was and still is 45,000 call minutes a month across a full PRI. We have numerous sites with Audio Code Analog gateways, CS 410's, SPA2002 gateways and soon to be another brand. Your references to channels hanging, wacky rollover issues, duplicates of troubles experienced with Asterisk, all indicate to me with no personal knowledge of your installation that an external event may be related. Successful interconnect companies for years have had to deal with these issues and adding the complexities of VoIP only add to the possibilities. Line Quality is always an issue - and previously dropped calls that were once tolerated will now be blamed on the new system. Fact O Life.

Jumping through hoops is the result of not having a plan. This technology isn't easy because it's IP and without a good IP foundation and experience you cannot make and execute a good plan. All you are left to do is GunSling hoping to find the magic bullet. Building and testing a plan prior to rolling out this or any other VoIP solution is always the best plan. If you are coming from the TDM telecom world, Congrates on your successes.

I'd like to review and create a consolidated checklist prior to upgrading 75 ext site using 2.1.5.2357 (Win32) from experience and posts, doing an inplace upgrade will result is an unknown loss of database dependancies that will require a resave of settings. Has anyone compiled from our joint experiences what specific config pages must be resaved to assure a seemless upgrade. We'd like to get it right first crack. Also what present release is the best choice? Cheers.

Not sure I saw that you were running Linux, but based on practical experience, I would prefer to see you move the files from the PBX onto another platform and allow it to email or archive the files. In the M$ world numerous tools exist that can be schedules to move files via SMB, FTP, or others. It would be nice to see some Meta Code for Linux to do some file moving using FTP to a known FTP site during Low CPU cycle times or to be CRONNED to occur hourly or after hours.

We've used this extensively with a client with 8 remote salespeople. Adding a 1 to the 10 digit telephone number redirection field will prevent the special IVR out dialing capabilities. Where this is very affective is adding a CELL PHONE Gateway that support SIP to allow remote cell phone users to place cell to cell calls and access to the the PSTN without using minutes. Also many cell providers have Circle of Friend (special numbers) and adding backdoor dialouts is an affective way to open the PSTN and reduce cell minutes. Beware, having the phone number in the cell redirection will always result in the IVR prompt and using the STOP caller ID *67 is the best way to prevent and get normal AA service. Cheers.

You best simply use the audio jack on the rear. If the previous explanation was to much, I recommend that you not attempt the process above until you are more familiar with basic linux commands. Please don't be offended, that isn't the intent, but these forums are not the best place to get training on the basics of an OS. Cheers

We regularly see where LECS don't have call disconnect enabled local switches like Northern Telecom, Siemens, and AT&T. This become obvious when a business begins using Autoattendents and Voice Mails as before they simply hung the phone up. PBXnSIP will time out unless the LEC provides a call disconnect option on each line. (polarity reversal, short, open) and sometimes the only way to get this enabled is to tell the LEC support that you want to verify that Call Disconnect is on each Line and not stopping until you talk to a real switch tech to get it done. Having a linemans phoneset and monitoring a call and have the remote caller hangup, the disconnect will be obvious when it's working right. Silence shouldn't be heard.

We've deployed several of the new CS410's since late January and they have been very reliable. Limit your logging unless you experience a problem. we upgraded them all to the latest 3.x 20 days ago and all continued to be stable just as before. All 4 ports are used on each.

I also believe placing a 1 in front of the 10 digit cell phone prevents this feature, but still allows redirections.