Keven Vachon Kiptel
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Posts posted by Keven Vachon Kiptel
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Testimonial?
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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
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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 ).
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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.
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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.
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I agree with all comments here , need docs seriously , this is why I'm back to big boys like panasonic .
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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.
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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 )
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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$.
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After over 1 year of debugging and testing with VOIP I decided that this technology is really not ready for the prime time , even if people like pbxnsip are having one of the best VOIP pbx out there I think that the SIP Protocol itself is not ready for enterprise grade like Nortel or Panasonic or NEC or CISCO , this is probably why CISCO developped their protocol . During the next week I will complete remove my installation to replace it with brand new panasonic Hybrid-IP one . I think that another 3 to 5 years is a necessity to reach the enterprise grade , pbxnsip did a great job with their pbx but every combination of phone and gateway that I tried with it can't handle the high quality demand of my customers ( 500 to 600 calls every day ) , each day I receive complaint about some glitch like , lost RTP , some echo , could not answer a call , semi-working sla , exclusive hold inexsistant , and alot more . After some research I discovered that this happened only on 10 to 15 calls of their 500 to 600 incoming one but its already to much , I can buy a complete brand new pbx from panasonic with 12 phones , voicemail , 8 CO lines and 24 extension for only 2500$ 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!
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No - we're using the Windows version.
We've upgraded to 2.0.2.1676, but we're still having the problem. I don't think it's the CPU - it's a fast dual opteron server with no other applications running on it.
Something weird here , could you provide us more info about your setup ( switch model and brand ) ,
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Has anyone setup SLA ( Shared Line Appearance ).... I'm looking at purchasing , but need to know that SLA works without any problems.
Example of SLA would be the following....
Phones used are Snom 360
A call comes in and they want to talk to a person in another area.... A page is made for that person to answer line 1 which was placed on hold.... The Person is near a phone and should see the blinking light on line 1.... The person picks up the phone and pushes the blinking line 1 to talk to the caller.
First of all I would like to know if this can be done and if so can it be done on analog lines as well as VOIP.
Yes SLA work with SNOM phone(only)but there is one limitation that you need to know , outgoing calls can't be pick up more than one time with SLA
Scenario :
Phone 1 placed a call and put it on hold
Phone 2 pick up the line currently on hold and all phones lost "lines" status
This is only for outgoing calls , every incoming calls can be put on hold as much as you want
Pbxnsip are aware of this limitation
The SLA work with Trunk but I could not get it working with VOIP Trunk.
If you need help just PM me
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i will try that. funny thing is that after getting dial tone if you place the handset in the cradle the dialtone still continues and at some point gets louder. most phones do have the option to give you dial tone after hold but only until you have the hadsret off-hook. thanks.
I looked at my 360 an this what you search
under Advanced
turn off "Dialtone during Hold:"
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latest v2 release. we may have deleted a trunk group while there were co lines and maybe thats why it is doing that. our license states 200 and i know we are not near that even with all accounts added up for all of the domains.
i tried using different names for the co lines like "st101201 st101202" etcetera and it still does not work. but we can make and receive calls. i just dont know how many calls the current config can handle since we cant see the co lines.
Ok , I tried something on mine
First stop pbxnsip , go to your pbxnsip directory into "colines" directory and delete everything , restart pbxnsip and try to recreate it like this
co1 co2 co2 co4 co5
( dont forget the space between each declaration )
Do you have a demo or official license?(I think that you need license to use colines now,let me check this with Kevin or Yori)
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Hi , I would like to find somenone to help us developping a solution to integrate PBXnSIP with SugarCRM , currently we are looking to develop a solution with SugarCRM for one of your customer but the only plugin that exist for VOIP is VoiceRD for Asterisk , please send private message if you're interest
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using the latest code on pbxnsip and snom300 phones when a call is placed on hold by pressing the line key the call is held and the outside party hears music but the phone starts to generate a secondary tone and does not go away until the call is taken off hold. so essentially when you place a call onhold you have to listen to a fairly loud dial tone while you go about what ever you need to do. does anyone know of a fix for this ?
First I think that you will need to read some Snom doc about their phone because its powerfull but you need to understand every option if you want that everythings goes well , but for now if you want to solve your problem go to your web admin interface on this Snom and disable the option "Dialtone on hold" and this is suppose to stop the second dialtone when you placed someone on hold.
Note:Upgrade your firmware to 6.5.2 if its not already done
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when we try to add co lines everything we entered disappears and the co line field goes back to blank. however we can make calls for that domain even though there are no co lines defined in the trunks section. anyone seen this before?
I have the same problem with mine ,I think there is a bug with 'co' lines and license , not forget that 'co' lines count in the license.
The way I found to solve this is stop pbxnsip,start pbxnsip and recreate "co" lines under your trunk , also never delete a trunk with co lines attached to it without deleting co lines before or you will not be able to reuse the co1,co2... if you recreate new trunk.You can also delete XML files about co lines and you will be able to recreate it ( always start and stop pbxnsip after this move)
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Hi - We have several Snom 360's (v6.2.3) connecting to PBXnSIP (v2.0.0.1611). Some of the phones ring when an incoming call is received (using a hunt group).
If anyone is in a call when an incoming call is received, the sound drops out for about half a second each time the phones ring. It's not just the phones in the hunt group that are dropping out - all the phones are doing it.
This problem started when we upgraded to v2.
Linux Version?
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Has anyone been able to get BLF to work with Aastra?
Wep , but Aastra are having very serious problem ( hang ) with their phone so I recommend you to stay away for now , I have 25 Aastra phones into production if you have questions
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Hi , after some testing with patton 4114 I can tell you that this product work well expect some little things . I'm currently speaking with Patton to resolve it.
The fxs port does not transfer the status to SIP side ( hold )
The fxs port has the "Transfer" function but its appear that you can't use it ( no docs for it )
Sorry for the delay have been busy this last days , well the Patton is good but not for use with FXS port , if you want to use Patton buy only the model with FXO port and buy their MATA for your FXS needs.
So far I tested this Gateway with PBXnSIP
Mediatrix 1204 = Good and work well with PBXnSIP but some echo problems that I can't solve , carrier voice quality
Patton 4114 = Good and work well with PBXnSIP , echo cancel is really better than Mediatrix but not completetly inexistant , carrier voice quality
SPA3000 = Inexpensive and impresive echo cancel for the price ( if you have the right settings )
SPA3102 = Same as SPA3000 but a little confusing for setup because of wan side , my home line is currently on a SPA3102 with PBXnSIP 1.5.2.5 and wireless phone on FXS port , with right setup not echo at all , voice is a little on cell phone side
Vegastream 50 6x4 = Never get it working with PBXnSIP , some problems with dialog between PBXnSIP and Vega
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I start this topic because it is a frequent topic. Please feel free to comment.
I used sipp to mesure some highload on pbxnsip 1.5.
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If you plan to use Mediatrix 1204 with "CO" lines on 1.5 you will need to disable SRTP ( Secure RTP ) or your 1204 will randomly stop doing RTP connect correctly for no reason
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Hi , after some testing with patton 4114 I can tell you that this product work well expect some little things . I'm currently speaking with Patton to resolve it.
The fxs port does not transfer the status to SIP side ( hold )
The fxs port has the "Transfer" function but its appear that you can't use it ( no docs for it )
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the 2.1 version has the overhauled recording, per trunk or user automatically ... no buttons necessary and keeps them available on the file system for easy retrieval and archiving ..
But, alas the overhauled price as well ..
yori
One word Orkaudio
http://oreka.sourceforge.net/download/windows
Free RTP audio sniffer with complete web based database , awesome I used it on Cisco and Pbxnsip system and everything working fine for over 6 months , don't forget to enable port mirroring on port that you want to record
Snom phone froum link broken
in snom Phones
Posted
Why is the snom phone forum don't work anymore from the snom one forum categories , when you click on it its writed "Unnavailable"