Keven Vachon Kiptel

Why is the snom phone forum don't work anymore from the snom one forum categories , when you click on it its writed "Unnavailable"

Testimonial?

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

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 ).

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.

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.

I agree with all comments here , need docs seriously , this is why I'm back to big boys like panasonic .

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.

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 )

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$.

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!

Something weird here , could you provide us more info about your setup ( switch model and brand ) ,

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

I looked at my 360 an this what you search under Advanced turn off "Dialtone during Hold:"

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)

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

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

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)

Linux Version?

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

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

I used sipp to mesure some highload on pbxnsip 1.5. http://sipp.sourceforge.net/

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

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 )

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