pcunite

Thank you. Looking at this link does agree, however, unless I do certain types of Transcoding. Is there a way to avoid those situations? When calls stay inside the PBX (user to user) I like to have G.722, for the outside world it is fine for the PBX to choose whatever codec it thinks it needs. Most calls will come into the PBX from people on PSTN devices.

I need to build two 2U rack mount Linux boxes. One will have Snom ONE PBX installed and the other box will have some router/firewall package (IPCop, ClearOS, etc.) installed. Prefer the same hardware for both to keep things simple. Usage: The PBX will have at most 50 simultaneous calls using G.722 and G.711u codec. The router will have at most 50 PC's doing simple web browsing. Naturally calls and surfing will overlap at times. The router will perform QoS/Shaping for the VoIP connections. Gigabit everywhere. I would appreciate your opinion on the hardware I've chosen. Is the CPU powerful enough? Overkill is fine. Hardware $500 budget per box: Case - iStarUSA $80 MB - ASRock H77M $80 CPU - Intel G2020 $70 RAM - Crucial 4GB $50 PSU - Corsair CX430 $50 SSD - Crucial M4 64GB $70 NIC - Intel PRO/1000 $30

Awesome, now please make an iPhone app as well. I always have my cell phone on me, and walking through our office I could decide if I wanted to call out over that. I know... kids are never satisfied!

Update! Phone call to support helped me. The default user/pass for an auto provisioned phone is "admin/password". Please I recommend you disable auto provisioning on new installs. I just plugged in a Snom 370 and the phone was totally taken over by PBXnSIP, it rebooted, updated the firmware and changed the password. This is was very alarming for a new user/noobie such as myself. Thank you for the help.

I need help, this is an emergency, I am testing your software, which has automatically taken over my phone. I can't log into the local web server of a Snom 370 and I need to go out of town with this phone today. Please advise me how to log into the local webserver of a Snom 370 that was auto provisioned by PBXnSIP ver 4.0.1.3499. Please also considering disabling this by default.

Testing PBXnSIP ver 4.0.1.3499 and my snom 370 auto updated, rebooted, and password all controlled by PBXnSIP. Please don't make this the default. What is the local username and password for the phone now?

IE 8 from a Windows 7 x64 machine. PBXnSIP is running on Windows XP x64 and listening on port 81 (something else is running on 80). In fact it does it just about every time on initial login in which the http://server:81/dom_index.htm page is rendered.

Something else is also up with it not showing all the images upon logging in, little red X's here and there.

There is a specification in the hardware that must support Wideband, HD, or TIA-920-B. All three terms mean the same thing which I believe means FM radio quality. Via software the phone must support the G.722 codec and so must your VoIP provider, but the call for the entire route must use G.722 or quality is compromised. If your familiar with Snom, their older 3xx series can be upgraded very easily with this, the new phones are already enabled for Wideband: http://www.snom.com/en/products/accessorie...rvoice-handset/ For Polycom look for "HD" support in the phones description.

Okay cool, don't worry about posting, I might buy the book someday. Still testing around with various ip pbx software offerings.

Has this ever been addressed? I have a 370 on order and look forward to turning the analog clock off.

Oops, something else was listening on port 80. Maybe PBXnSIP could write to the log this error. That would have clued me in. Working fine now.

On Windows XP x64 starting the service terminates instantly.

FutureQuest.net or Pair.com

Testing 4.0.1.3452 (Win32) today. We used: * 1 VoIP trunk * 1 Auto Attendant * 2 service flags (during and after hours) * 1 Hunt Group (extension 1000 and 555-5555) * 1 extension (1000) Everything on our list worked with the following exceptions: * There was no music on hold, only ringing * An agent group was not tested as suggested. Is this not priced higher than the PRO version and not on the sipJack? * The final stage of the Hunt Group, extension 1000, played a prompt before our customized greeting. Does not seem to be an option to only play our sound file only. BUGS: * We were able to get pbxctrl.exe to enter an endless loop based on attempting to use our current browser session after a restart of the service. That machines IP address now can never connect as pbxctrl.exe will enter an endless loop. * Other instabilities with this version made it so we could not properly test various handling of calls, often times a busy signal was generated for the caller. A restart of the services corrects this until it happens again. The bugs may be the result of testing inside Virtual PC 2007 using Windows XP SP3.

Yes, I just edited my post, please re-read it. But yeah, ringing I think it was. I will test again tomorrow.

Thank you for helping nail this description down. I should mention that an extension (it does not matter if local or remote PBX) does NOT need the "press 1 to answer" feature. A PBX extension user knows what is going on. Only external numbers, that is to say numbers not controled by this PBX. Basically just calling a regular home/office line and cell phones need the "press 1" option. My apologies for that. I will try version 4, two days ago I was using 4 and it did not have constant uninterrupted music after the first auto attendant which is a deal breaker as well. I will try again as I might have missed a setting. Just to recap, the caller should only hear three sound files, "Hello, music, sorry everyone is busy leave a message". All customizable of course.

Correct. The reason for this request is that some phone numbers in the hunt are dedicated offices, they know a customer is calling. Other numbers are personal cell phones, and the user's don't know if a customer or the wife is the one calling. This needs to be customizable per hunt entry because of this.

What I need to leave a hosted PBX company. I must have this exact call flow. **************************************************** Auto Attendant: - Business Hours and After Hours awareness. - When PBX answers call, customizable sound file ("press 1 for sales...") - If no press after X seconds go to extension or hunt group. - AFTER the above events have occurred, customizable play sound or NO sound. - Send caller to hunt group or extension. Hunt Group (FollowMe/FindMe): - Dial internal extension (100) for X seconds then external (555-5555), etc.. - Having the external number listed here instead of a dummy extension is easier to manage complex routing. - If a hunt entry does answer, then a per hunt entry customizable "Press 1 to accept" or play sound file. - At all times when dialing different numbers/extensions play music on hold with no interruptions unless desired. - If no hunt entries have answered send to Voicemail with customizable sound "all reps are busy, please leave a message". **************************************************** The above requirements will keep a caller in the PBX and won't update them with events that are unimportant to them. This call flow is also geared/designed to present a company, not a person or even a group of people. The PBX only plays three sounds... hello, music, human, or I'm sorry. To summarize The caller hears an auto attendant, they press a button or wait, they hear music, then they hear a human voice or go to voicemail. None of the windows compatible PBX's offer this exact set of capabilities. Until they do I am stuck with RingCentral or an Asterisk solution. I will be forced to make a decision soon and that decision will be to stay with RingCentral another year. I hope to check back on the status of PBXnSIP and find it a purchasable product. Well wishes to the developers and keep up the good effort.

Just to confirm, when a caller has been sent to a Hunt Group, MoH does not work? I can't get it to, and I suppose this is why. Any other features that don't work but appear to?

CallCentric is a good provider but the people you're speaking to could be over a bad connection. My suggestion would be to connect your phone directly to CallCentric, by passing PBXnSIP all together to prove that point. If these calls your making are overseas you may want a provider that has a server in that location.

The sipJack looks interesting but not as flexible for other uses should we ever want to run something else or use the device for something else like a fit-PC2i would be. Has anyone used a fit-PC2i? I admit that I have never worked with such small devices but we are looking into getting smaller devices for purpose built servers. Thoughts?

Matt, I can't thank you enough. I was searching for VoIP related things and found your video review of the sipJack (sheeva pbxnsip) just today infact. In about an hour I had PBXnSIP working were 3CX would not. Yeah about that post, what is weird is that when I first posted it, it parsed the string "PBXnSIP" and replaced it with "other PBX". That is when I added the spacing. I don't trust 3CX now at all.

I look forward to learning PBXnSIP vs. 3CX. Went round and round trying to get 3CX to call a cell phone for X number of seconds, it never would return back to the PBX, just kept ringing. PBXnSIP works as expected so that is a relief. We are going to be migrating from RingCentral (excellent company by the way). I will be testing PBXnSIP all next week to make sure it dots every "i". Looks good so far from my hour or so of playing around.

Never mind, license key was inserted.