Leonmeijer

Hello, I was trying to set up an IVR node that only playes a sound and then redirects the call so I used: !E!100! !T!100! in the DTMF match list, actually it didn't work utnill I restarted the service, I've detected that there are more functions that need a restart of the service, why not announce that on the page of these settings? Also trunks somethimes need a restart of the service somehow... I also want to fully DISABLE user input, the wave file should keep playing also if the user presses a button on the phone, only the # sign shouuld stop playing and follow the !E!100! dtmf, how to do this?

I'm using version: 4.0.1.3438 (Win32) (Beta). The sips are registering via UDP/TLS (I think also TLS...) I don't know where to set it to TCP but I guess UDP is ok for sip not? I tried various refres times, 20 seconds, 2 minutes, 10 minutes, 20 minutes and 1 hour but after some ammount of time the refresh stops working and all the sip registrations are lost... the status shows a " 200 OK" message but the logfile (which is set to the maximum logging level) shows only SIP registrations from the phone, all the trunk data is not logged. I've also seen once that the trunk status was "blank" after 1 day. Also, if I kill the connections in the firewall the pbx shows "408 Request timed out (retry after 60 seconds)" somethimes but that message remains untill I manually click the "register" link, the 60 seconds do nothing... It's very strange but can't find a way to solve this...

As iv'e said, it's NOT the firewall but if you really think so, of cource I tried disabling the firewall but exact the same thing happens, even if I have the PBX as DMZ host without ANY firewall on the network this thing happenes. What I see in the log is some 401 and 404 messages from INCOMMING connections from 1 of the IP's of the sip host, the sip.sipnl.nl and sip.voipcheap.com have more then 1 ip set and some times when I check the Access thing in PBX I see that the pbx has blocked one of these IP's so it looks like the pbx plays the firewall here, but I will do a reconfigure... Let the snom connect the sip server and forward the calls to the pbx which forwards the calls back, quick and dirty but works fine, maybe in a stable release of V4 it will work again but I don't know why but it works bably. When I worked for another company I had the same issue, for them I wrote a program that logs in the PBX webinterface and virtually "clicks" the register button, guess such a solution I will have to make again...

Now what! A few days later, in my firewall I see NO SIP connections. I log in to the PBX (I use the beta 4 btw, maye that's it... on A P3), I click Trunks and what I see is: All 4 of my trunks, the status column for all trunks is "EMPTY!" then an edit button and a delete button, how is this possible, PBX just drops the connections or something... this is really strange, anybody got any ideas? I use 2 accounts of VoipCHeap.com and others of sipnl.net (both different providers). Edit, the only sip connection in the firewall is a connection from the phone to the pbx...

Hello all, I'm having troubles with my trunks, they disconnect every 8 minutes, I've set keep-alive to 2 minutes and the Proposed Duration (s) is set to it's default. Actually, when I look at the connections in my FireWall I see the "age" of the connection is about e.g. 30 minutes, but the connection Timeout is reset to 8 minutes everytime, when these 8 minutes expires, the SIP registration is lost (the webinterface shows that it's just fine...) then, after the 2 minutes of " keep-alive" have expired, the connection is recreated but it looks like it's not really keeping them alive, now very often I have problems with the inbound calls, when e.g. I set the keep-alive to "blank" I can have inbound for 8 minutes only after the re-registering. The registration of my Snom phone also gets the 8 minutes but it re-registeres directly when the connection is lost, what does the PBX doe to check if the connection is disconnected somehow? it looks like "nothing" in this case, or did I just miss an option somewhere? I'm certain that it's not a firewall setting.

The upgrade has succesfully started will test it from now. Thanks for the support Leon

Yeah, why givin' up the good old stuff if it just works fine Only thing I found at the moment is that the pbx software "stops" at memory address 00546838, looking there with the debugger shows the processorcode (assembly) to be screwed up, and an " Invalid Instruction " error returns and then the program shuts down.

I tried disabling Apache and all other software that is using "ports" also the FireWall but still the PBX is not updating / installing. Still the "Service does not respond to the control function" error when typting net start "pbxnsip pbx" on updating...

I've configured the web interface to run on port 800 because the webserver uses port 80 and 443, does the new version maybe use port 443 too, and if so, how can I change it then, all other ports are useable for PBXnSIP. Leon

Hello all, I receive an error at the end of the installation of PBXnSIP 4 beta, the windows app log shows: Product: PBX -- Error 1722. There is a problem with this Windows Installer package. A program run as part of the setup did not finish as expected. Contact your support personnel or package vendor. Action _0CF9C641_7AE5_407D_8D5B_CDE6A976C38A, location: C:\Program Files\pbxnsip\PBX\pbxctrl.exe, command: --dir "C:\Program Files\pbxnsip\PBX" --service-install The error message popsup is " There is a problem with this Windows Installer pacage. A program run as part of the setup did not finish as expected. Contact your support personnel or package vendor." When I click OK the setup rolls back. When trying a manual update, the PBX service is unable to start. So now i'm back at the latest 3.4. The .net installer completed succesfull. I have the latest windows installer. Trying to install on Windows Server 2003 Enterprise Edition English 32 bit. It's an old compaq DeskPro system Pentium III 800Mhz 384MB of memory. Andy ideas what might cause this problem?

Agree, but I just can't get that, a powerfull PBX system such as PBXnSIP just doesn't support fax2mail (or something) and a "simple" fritz!box modem/router supports it...

That could be handy but what I need is a voicemail for the caller who is at the top of the queue (waiting for the extension to be picked up) when the extension is rinning eg 15 seconds it must redirect to the voicemail.

Yup, especially when you host everything yourself like a SIP server:P

I understand about the pricing, but I don't care paying for that, I don't have a fritz!box but, this peace of hardware costs only 160 euro's, that means that - the fax module costs at least 60 euro's because a LinkSys, Sitecom or speedtouch modem without SIP/fax costs 99 euro's, I actually don't see the point there but.... I hope it would be included someday :-), for now I let my SIP provider move the fax to the mail, they do it for free, the only disadvantage their sip server has is, when fax is enabled, voicemail must be enabled too, but can live with that for now;)

Oke, it was easyer as i thought, put a &mode=posts in the URL and that changed it for lifetime... thanks though.

If it can run easily external, it should be logically included in PBXnSIP, tell me if i'm wrong but... PBXnSIP is a PBX system or.... not?

Hello, I've noticed a change to either the theme of the forum or my account settings, but can't find the way to change this thing back: When I open a thread, I see the first message in the thread and below that I see: "Posts in this topic" and below that a set of links to all replies instead of the replies.... I'm using SuperNova screenreader to read the forum using speech and braille, this way of showing the topics is very irritating and annoying, how can I change this setting so just all the replies are shown and when opening the thread the last page / reply is shown, can't remember I made any changes to the theme settings or account settings... Please help;)

Will PBX v4 support Fax 2 Mail, even my Fritz!Box modem/router supports it... thought PBXnSIP is a modern PBX system? But I'm finding many things that are just not there to switch to another system soon...

Even when I configure my SNOM360 to redirect after 10 secs to 8<extensionnumber> doesn't redirect the call in the agent group :@ why, why, why...... Please include such a feature in v4.0

Hmm, that seemed to be a one time only thing... now it works normally but then the second problem comes up, while playing the message, pressing a key on the phone stops the wave file from playing and just left the caller alone in the dark (nothing is happening, there is silence and the caller can wait but.... the IVR node stops functioning) I want to disable ALL USER INPUT ont he IVR node... how can I do this?

Removed the space, now the wave file is played 1,75 times... then the call is redirected. I've recorded the text " blablabla" and the pbx played: blablabla - blablabl - redirect with a little silence between the first and the second play of the blablabla text.

Hmm, i tried the E and "action: bye" as shown on the wiki but... the IVR keeps on repeeting the wave file and isn't doing anythibng, pressing any key stops the wave but nothing happenes... Is this introduced in a later version? and I want to fully disable the keypad while the wave is playing so the wave isn't stopped when someone presses a key.

Yes but it keeps repeating the file so e.g. you have a wave saying " Hello" calling it it does " hellohellohellohellohellohello" untill the user presses a matching key, when using the global expresions or extenion matching it forwards directly without playing the wave... So I want to have it simple, just a wave file tobe played without a lot of configureable things.... like SWEEX pbx does, just click, drag and drop and there it is....

Hello, I want the following config: Call comes in to the trunk and goes to a " wave file" in the " wave file" settings I want to set weather to wich extension to " forward" the call or do forward it to /dev/null (drop the call) like an out of office notification: "Company is opened from .... untill .... the connection will now be dropped" or: " Welcome to company, you will now be forwarded to our technical department" -> forward to extension. - I don't want to use AA because NO buttons are allowed to be pressed (also * and #). - I don't want to use a hunt or agent group because it's no group it's just a notification. How could this be done? if I can't it's a feature request;)

Thanks for this topic! I have the same issue! Here the feature works, it redirects te call to 8<extension>, the voicemail BUT: when the extension is busy the call goes to the waiting music, OK, but then it redirects to the voicemail TOO. I want the PBX to follow the voicemail rules in the agent group, I don't care about which voicemail answers the first, the first that redirects, the first that answers. How difficult is it to add this feature??? I'm really frustrated about some features PBX is just not including or things are not possible, I'm really thinking about switching to another PBX system which has got all the features needed and if it isn't available the user can just write a peace of script to add it.... how difficult??? Regards, Leon