I pulled the plug on my VOIP installation

[Keven Vachon Kiptel]

After over 1 year of debugging and testing with VOIP I decided that this technology is really not ready for the prime time , even if people like pbxnsip are having one of the best VOIP pbx out there I think that the SIP Protocol itself is not ready for enterprise grade like Nortel or Panasonic or NEC or CISCO , this is probably why CISCO developped their protocol . During the next week I will complete remove my installation to replace it with brand new panasonic Hybrid-IP one . I think that another 3 to 5 years is a necessity to reach the enterprise grade , pbxnsip did a great job with their pbx but every combination of phone and gateway that I tried with it can't handle the high quality demand of my customers ( 500 to 600 calls every day ) , each day I receive complaint about some glitch like , lost RTP , some echo , could not answer a call , semi-working sla , exclusive hold inexsistant , and alot more . After some research I discovered that this happened only on 10 to 15 calls of their 500 to 600 incoming one but its already to much , I can buy a complete brand new pbx from panasonic with 12 phones , voicemail , 8 CO lines and 24 extension for only 2500$ everything included so why wasting my time with custom solution?

 

So everyone at pbxnsip keep your good work! I think that you have one of the best voip pbx out there but for me is enough , if someone is interest I will sell my pbxnsip license 1.5 25 Users , 11 snom 320 phone , mediatrix 1204 , 11 Aastra 480i , Sangoma PCI 4 channels with HW Echo canceller , if someone want it functionnal with the server and all hardware I can sell the complete package also ( 5 months old )

 

IBM XSeries 100 , Cel 2.53Ghz , 80GB Raid1 , Debian 3.1 , pbxnsip 1.5.2.5 , 11 snom 320 , 1 mediatrix 1204 , 1 spa3000 setup for wireless fxs ,

Dell SC420 , 40GB Raid1 , Debian 3.1 , Asterisk 1.2.14 , 11 Aastra 480i , 1 spa 3102 setup for wireless , Sangoma A200 with 4 fxo and HW echo cancel.

 

also 2 switch POE Linksys SRW224P ( 24 ports POE 10/100 and 2 ports 1000/fiber for server .

 

 

Thanks and have a great day!

[hosted]

After over 1 year of debugging and testing with VOIP I decided that this technology is really not ready for the prime time , even if people like pbxnsip are having one of the best VOIP pbx out there I think that the SIP Protocol itself is not ready for enterprise grade like Nortel or Panasonic or NEC or CISCO , this is probably why CISCO developped their protocol .

 

Need a quote for a new PBX? haha just kidding.

 

I disagree with you. VoIP is only as stong as the money you invest in it. I would however agree with one fact SIP is featureless. you step back about 4-5 years in technology feature sets.

 

People get into asterisk because of cost. pbxnsip is in my opinion the best key system emulation of any SIP pbx. Built properly i think pbxnsip has 80% of the features of the traditional PBX's at a fraction of the cost. Look at bigger PBX systems who dont operate as a key system i would say the feature set it closer to 90%.

 

Any loss in quality is not a properly build PBX.

[Keven Vachon Kiptel]

Need a quote for a new PBX? haha just kidding.

 

I disagree with you. VoIP is only as stong as the money you invest in it. I would however agree with one fact SIP is featureless. you step back about 4-5 years in technology feature sets.

 

People get into asterisk because of cost. pbxnsip is in my opinion the best key system emulation of any SIP pbx. Built properly i think pbxnsip has 80% of the features of the traditional PBX's at a fraction of the cost. Look at bigger PBX systems who dont operate as a key system i would say the feature set it closer to 90%.

 

Any loss in quality is not a properly build PBX.

 

 

I agree for big pbx(50-100 or more) but the SME are still better with a good key system like panasonic with USB port to program it. I just bought a complete panasonic ta-824 with 12 phones,voicemail that can handle 8 co lines and 24 extensions and this just for 2300$ brand new. I can also buy panalog to have a complete CDR report software just for 200$.

[hosted]

I guess it depends on what you want from a PBX. I would not give up my IP system for a Panasonic.

[Keven Vachon Kiptel]

I guess it depends on what you want from a PBX. I would not give up my IP system for a Panasonic.

 

 

Panasonic installed this weekend , exclusive hold , share line appearance , multi color led for BLF,SLA , Can program all phones from one click and this for EVERY function, from USB port , can connect it to VPN for remote programming and CDR , all I want and this for less money. I really hope that VOIP will catch up with time but for SME they are seriously out of the game in term of functionnality and stability.

 

If somenone need phones, gateway PM me ( mediatrix,snom,aastra,sangoma )

[hosted]

Panasonic installed this weekend , exclusive hold , share line appearance , multi color led for BLF,SLA , Can program all phones from one click and this for EVERY function, from USB port , can connect it to VPN for remote programming and CDR , all I want and this for less money. I really hope that VOIP will catch up with time but for SME they are seriously out of the game in term of functionnality and stability.

 

Functionality yea i suppose. VoIP is more like Centrex service. Todays digital systems are superior as feature sets go. I think the pbxnsip/Snom combo is pretty good.

[Daniel Floeckinger]

I think the pbxnsip/Snom combo is pretty good.

 

Sorry, did i understand that correctly? PBXnSIP + SNOM works fine??

 

Let me tell you what we try to do.....

We try to get SNOM and PBXnSIP WORK....

 

And... here is our list of problems.

 

1. SNOM TLS ... Is not supported with PBXnSIP. Phones no moe register after a couple of hours. Need to be restarted.

2. SNOM SRTP ... After 7-10 mins, there is white noise on the line. SRTP with snom does not work.

3. DND function ... If you switch DND on and off a couple of times in a short period of time, the phone does not register anymore

4. SNOM 370 ... not supported for PNP

5. Installation with PNP does not work at all. Several phones register with the same extension, no matter if the MAC address is set or not.

etc....

 

It does not work at all!!

[hosted]

Sorry, did i understand that correctly? PBXnSIP + SNOM works fine??

 

Let me tell you what we try to do.....

We try to get SNOM and PBXnSIP WORK....

 

And... here is our list of problems.

 

1. SNOM TLS ... Is not supported with PBXnSIP. Phones no moe register after a couple of hours. Need to be restarted.

2. SNOM SRTP ... After 7-10 mins, there is white noise on the line. SRTP with snom does not work.

3. DND function ... If you switch DND on and off a couple of times in a short period of time, the phone does not register anymore

4. SNOM 370 ... not supported for PNP

5. Installation with PNP does not work at all. Several phones register with the same extension, no matter if the MAC address is set or not.

etc....

 

It does not work at all!!

 

I havnt tried the 370 but i havnt had these issues with 2.0 version of pbxnsip

[Keven Vachon Kiptel]

Just to add some news about my new life without VOIP , I'm happy!! . Seriously if you want to make money and sleep at night you're better to install VoIP only in your business or not at all.

[lance@YSL]

I am glad that you are happy with your change - I would do the same if I had had your experiences and hassle... I have a few comments...

 

VoIP years before its ready... Hmm... Perhaps in certain aspects it may well be but as a general comment about VoIP I would have to disagree.

 

The key here is that the requirements of a site/customer need to be correctly mapped out and the solution chosen to meet those requirements - if you had requirements that VoIP could not meet, then I would have gladly steered you away from VoIP from the very start and recommended a traditional or hybrid solution from one of the other providers.

 

A well designed well spec'd VoIP solution can work very well... It is important NOT to put force a square plug into a round hole.

 

Seen as though you make a general comment about VoIP - here is a general comment of a live client - We have a client running on a VoIP solution (nonPBXnSIP) with digital ISDN connections and a ITSP connection for least cost calling... They are about 80 people and a 24x7 call centre.... Their call volumes (excluding internal calls) are over 30 000 per month (similar to your call volume) and in that period they have maybe 0.001% call problems of a nature similar to yours (not quantity)... I am sure other readers will have some similar success stories to mention.....

 

The key here is that the system chosen and designed was taken with all its limitations and benefits and it was not compared directly to another traditional PBX... i.e. the client wanted a phone...Tick, the solution did that, the client wanted cheaper calls, Tick, the solution did that.... the client wanted billing, Tick, the solution did that... It did what it was intended to do AND more importantly it did not have to be reworked or patched or linked in a hundred little ways to meet the client requirements.... It was spec'd to handle a specific call rate.. which it does... etc etc. etc...

 

Just looking at your mix and match hardware and software solution and system spec makes me cringe - even if I did not read your problems, I would have put money on the fact that you would have had problems from day one....

 

We have now moved to PBXnSIP and are using the same approach... If the client wants something we cannot professionally supply, then we simply 1) tell them the function is not available and 2) a patch job to make it work will undermine the solution to you... Everything is properly spec'd and hybrid workarounds to make an old system (like the accountants paper fax machine) work are dealt with and immediately addressed...

 

We have a large amount of faith in VoIP and PBXnSIP and are putting a huge amount of money (read tens of thousands of US$) into setting up a hosted solution for 1000's of users....

 

VoIP is here to stay.... Anyway... just my two cents on a friday afternoon....

[Keven Vachon Kiptel]

The next time that I will touch VOIP my bet will be on complete manage system like Cisco,Avaya,Nortel,Panasonic,Nec , I'm sure that they do their homework correctly , you plug a phone and everything work like its suppose to. I'm agree with you that my hardware choice was a mistake

 

Aastra have bug over bug with their phone and they don't even acknowledge it ( look at trixbox forum and their releases notes documents )

Sangoma push production products with beta drivers

Asterisk is simply a big marketing joke and the digium hardware is like an old 56k modem with jumper settings in a 80486

Snom is good but their phone are not durable like Polycom or Cisco

 

The problem is that pbxnsip is a great product but its lack some polish around it ( from SME point of view ) , I want to buy an analog gateway ( echo free please ) with some phones and get it up and running in the next hours , this is the point of big player like panasonic or nortel , its work , you pay but its works.

[Vodia PBX]

Check out the latest Cisco financials and how much they made on VoIP. They were smart enough to come up with a solution that is pure IP, and they got it working. Doing the same thing in SIP instead of Skinny is more complex, but the result will be even better than a closed solution from one vendor. Then check out the latest sales report about the WiFi-based phones. Maybe Cisco buys Nokia to keep their solution closed!

[mustardman]

Excuse me but I gotta take exception with your comments about Sangoma and Aastra.

 

Aastra is trying to satisfy absolutely everyone with a mind numbing list of features that work on everything from PBXnSIP on Windows to Asterisk on Linux. When they first came out they were buggy and feature incomplete just like everything else that just came out including Snom, Cisco call manager, Panasonic Hybrid etc. As of now the Aastra phones work great.

 

I don't know what your reading on the forums but the Sangoma cards just work! If you are using Beta drivers and open source software echo cancellation that's a whole other story.

 

I have production installs using Sangoma and Aastra phones running for close to a year now. One customer is so happy with it they want me to expand it to cover their entire organization linking several offices across the US. I wouldn't be doing this if I didn't have complete faith in Sangoma and Aastr products but you gotta know what your doing. If you don't have a good hardware/software foundation to begin with then it does not matter whose products you use.

 

Yea, you could throw a bunch of money at it and get an integrated (and closed) solution from Cisco that just works usually. The difference with PBXnSIP/Asterisk etc. is that YOU are doing the integration. Again, you gotta know what your doing, test the solution etc. etc.

 

Digium X100, TDM400, T110 cards are absolute garbage. Their newer cards are much better now that they are using a better PCI interface! I still much prefer Sangoma but things are changing rapidly.

[hosted]

VoIP will eventually takeover TDM thats a given look at the market. Its just logical that IP applications and Voice will merge its called increased productivity and people pay for it.

 

I dont see ANY advantage with polished providers vs pbxnsip or asterisk, except they are different applications. IP is IP if you deploy wrong it will suck even it the brand is Avaya.

 

Asterisk is getting better, for a lot of companies its stable but it doesnt scale really good. Asterisk is in the dark ages of open source telephony.. as was Linux at one point. I have installed well over 100 asterisk solutions quite honestly i wish i would have installed pbxnsip but i didnt know about it back then. Plus im damn good with asterisk :/ I have so far installed 4 pbxnsip (just taking it slowly for recaution). Overall I really like it and am going to continue with pbxnsip over asterisk.

 

I have a couple hundred Aastra phones out there. Not an issue with one of them. I have a config (and firmware) that works, no side tone/echo at all. They are priced good. I just dont install them in call centers because of the way they handle multiple incoming calls (unless that was fixed).

 

Sangoma is the best. They are the technology leader and rock solid. Again if you run bleeding edge firmware you will have issues. Use one that works and leave it alone.

 

If your going to sell this low end put it together PBX stuff you better as hell just find out what works stick with a configuration that works solid and dont change things. staying bleeding edge causes bleeding ulsers.

 

Lastly, if you sell asterisk as OpenSource. you better explain the Open Source model ie software is free however the maintenance/knowledge is not. Bottom line- selling asterisk as a cheap solution is WRONG. its a cheap initial investment but it is anything from cheap. There are FAR to many asterisk people deploying and getting good cash for the installation but walk away from customers because its a service nightmare. Then i come in. Sell them stable, sell them great service. If pbxnsip had a 'HUD' (GUI operator panel) there would be no stopping me. Its hard to sell them into not being able to see the calls like that when they use it. Right now im working on a small deployement of FreeSwitch that talks nativly to wanpipe (sangoma) to be the SIP gateway. So I can then re-use asterisk box's including the T1 cards.

[Keven Vachon Kiptel]

I will tell you my story about Aastra , first Aastra is locate in Ontario, Canada and I'm locate in Quebec,Canada so when I came to VoIP first time I used Aastra because of proximity and also because I like to buy in Canada, the phone work correctly but only if you use the basic fonctionnality , multiple incoming call is a nightmare and heavy use of BLF hang the phone , the problem about incoming calls taking over the call your're currently dialing was report by me last year and at first they tell me that this is a functionnality that some customers want . After some reading I recently discover that they introduce a new function in version 1.4.2 to solve this but 1 year later!!!! , you can look at Google groups Aastra-480i users and you will see that alot of people get the hang problem with heavy BLF use , also if you look at trixbox forum you will see that even their new phone's series get this problem of hanging with BLF , after that I used snom with pbxnsip 1.5 but like one people on this post I lost RTP for no reason and sometime I get whitenoise and I need to completely reboot the server ( SRTP problem here to ) , I asked pbxnsip to get a free upgrade to 2.0 ( only 2 months after 1.5 order ) but they tell me that I need to pay for the upgrade. I'm using VoIP for my business and I'm looking from time to time on various project forum to see evolution of various project ( Callweaver,Asterisk,pbxnsip,trixbox and SIPx ) , its look like various part of this world doesn't have the same defenition of quality , today quality mean nothing and money mean everything( at least in America ) , marketing take decision and f**k the rest. Over months only the SIPx project as give me some feeling of quality and transparent bug aknowledgement and man their wiki is so much great! The Jira Issue Tracker they used is so great , you can see very clearly if a bug can block your deployment and their plug and play phone system can upgrade phone firmware, setup gateway and setup phones config files.

[SGJ Solutions]

Well this is not my experiance, I have many installs out there in version 1.5 and version 2, some with itsp only connections and some with isdn connections.

Some are stand alone some are intergrated with existing traditional pbx's

 

I've been doing voip and sip for over 10 years now! originally with Snom 4s and then moving to PBXNSIP.

All my systems are fine, work correctly and dont' fall over.

 

2 Points (1) use a decent gateway Vega is my choice (2) build on linux more stable than windows.

 

Takes me one day to build and deploy a system of up to 20 handsets including gateway work.

 

I work with some of the main pbx manufactures as well and they're voip stuff isn't even close and not open.

I was a SIP convert back in 1999 so maybe I just remember how bad things were when we started

 

Steve

[Keven Vachon Kiptel]

Well this is not my experiance, I have many installs out there in version 1.5 and version 2, some with itsp only connections and some with isdn connections.

Some are stand alone some are intergrated with existing traditional pbx's

 

I've been doing voip and sip for over 10 years now! originally with Snom 4s and then moving to PBXNSIP.

All my systems are fine, work correctly and dont' fall over.

 

2 Points (1) use a decent gateway Vega is my choice (2) build on linux more stable than windows.

 

Takes me one day to build and deploy a system of up to 20 handsets including gateway work.

 

I work with some of the main pbx manufactures as well and they're voip stuff isn't even close and not open.

I was a SIP convert back in 1999 so maybe I just remember how bad things were when we started

 

Steve

 

 

 

First my installation was with linux and second how much VoIP system did you install with SLA ( Shared line appearance) that work correctly or even work? ( except big brothers like avaya,cisco,nortel and panasonic I never saw any wannabe voip sip system that can do SLA correctly except the small one name allworx with their in house phone ).

[thomasv]

I guess it depends on what you want from a PBX. I would not give up my IP system for a Panasonic.

 

Same here

[Keven Vachon Kiptel]

Finally , I get contract over contract because of things I learned during my VOIP story. People get VOIP from big telco providers but their employees are not able to support it correctly so they call me to change settings, setup new phones. I thinking about trying pbxnsip 2

[Kristan]

I can't really recommend PBXnSIP highly enough. The PBX itself is great, the support is first class, and for the price I think it can't be beaten.

 

It takes me no time at all these days to knock out systems, especially using either snoms or polycoms as the autoprovisioning makes everything a breeze. Users like it because the web interface is easy to use and I like it because it's so flexible.

 

Yes, there is the odd bug that appears but generally it's stable and works well, and if you do find something, the PBX support guys are onto it straight away.

[gotvoip]

Hi Kristan, Can we use this as a testimonial on the pbxnsip web site? Thanks!

[Kristan]

As long as you leave the bit about the odd bug in

[Keven Vachon Kiptel]

Testimonial?

[andrewgroup]

Done bashing? PBXnSIP has proven itself in far to many installations and like anything, is dependent on skilled installers.

[Patel]

Done bashing? PBXnSIP has proven itself in far to many installations and like anything, is dependent on skilled installers.

 

Yup but I'm not a skilled installer, yet.