SGJ Solutions

I am strugling with the same issue, Provu is attempting to get assistance, can somone please get in touch with Provu it has been 2 days now

If I want to switch between admin level and user lever on pbxnsip. I have to close the browser and start again, otherwise if I try to re-enter the pbx address it logs me in as the previous log in. The Log off button which was on the top right in version 2 has gone walkies in Version 3 cheers

I have been testing the IM bit of pbxnsip. I sent an IM to ext 100 a snom 360 and yep it appears. However how to cear it off the Snom's display? I can't seem to clear it regards

Are we now able to use the UDB Dongle to licence a box, as I'm looking at having a hot swap unit and would like to be able to use the dongle. System is running SUSE10 Enterprise cheers

Actually this is intersting From the list of missed calls or call log click to dial gives you this http://192.168.168.250/remote_call.htm?dest=07976321995 However domain and personal address book gives you this link which fails https://192.168.168.250/remote_call.htm?use...est=07976321995 Seems the problem is the extra user=300%40pbx.sgj.int

This is the link in my version https://192.168.168.250/remote_call.htm?use...est=07976321995 I sometimes get not valid certificate as well (not always though) but always followed by 404 page not found. This is the same for personal address book and domain address book

Oh that's great many thanks for the list. In fact the list just shows the difference I need. To every one else on this forum, I agree that I can't spend weeks de-bugging and install. PBXNSIP on a linux platform takes less then 2 hours including install of SUSE. Then setup DNS and walk away in the knowlage that the thing stays up. Every pbxnsip I have installed has been very stable, even in the old version 1.5 days! thanks chaps

Which kicks this crap pbx into touch. If I see another customer and they mention this great free (in theory) pbx I will scream. I don't have the time of inclination to even put it on any of my servers. So has any one got a nice 10 point list of why PBXNSIP is better than this so called pbx (doesn't even have a conference server) cheers Still here selling pbxnsip as fast as I can

Yep that fixed it cheers

Nope made some progress though. The misses call and call log now work which is better. However personal and domain address book doesn't allow to dial from web. You get page cannot be displayed. Running 2448 on SUSE10

I was running 2.1.6.2446 which was the latest when I downloaded it. Will try latest version thanks

I know we have implimenmted T.38 codec detection and passthrough for faxes. But have we done the same for Video yet? The price of video phones are coming down and I know somone is going to ask the question at somepoint. cheers

OK any reason why? from a users log in on pbxnip. That the missed call and call log allows you to click to call back and your ext rings and call works But Domain and personal address book do not work If you click on them (page can't be displayed) Also address book lookup still fails on inbound DDI's? We really need the Alpha Tagging to work.

Anyone found a decent way in SUSE to get one pbxnsip install to copy auomatically its's config to a second backup box. So that if first box fails then the second box has an up to date config and can take over. Maybe do a transfer everynight to the second box

There is an option to rewind and forward while listening to a voicemail. This is under the extra optons in voicemail. However if you try to use this feature the playback becomes distorted and has random issues, echo playback slows down even.

With the latest version of pbxnsip, running with snom 320 handsets, the ring tone seems to be fixed by the phone system. Even if you try to alter the ringer on the handsets pbxnsip seems to play it's own ringer. How can we fix this? We need different ringers for different lines, ie main line and personal DDI Anyone come across this?

problem solved, it was the ITSP trunk messing about

I am struggling to work out why the automatic call recording doesn't seem to work. I can see the files being generated however with a cisco handset the file size is too small and prob just a blank file, with eye bean we get a larger file, however when you try to play back windows media it says not supported code. I can play back using Blaze meedia pro but all I get is half a second of white noise regards ok update We have found that ext to ext now works, it seems the calls via the trunk dont' get recorded correctly

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

Thanks SUSE10 link now works cheers Steve