Leonmeijer

I enabled the Global radiobutton and the Accept Redirect on the trunks. Each domain has it's own trunk because echt domain is another company's so they have their own outbound number. And what is two domains both contains extesion number e.g. "100"? Or do I need to set some other thing in the trunk config? Edit: I also set the "tel:xxx" alias names but... nothing

Some time ago iv'e seen a topic like this but can't find it anymore, but I want to call with an extension from one domain to another I tried to set the trunks to "global" but this didn't take effect, how do I need to configure this?

I mean that the TAPI driver is not visible in the Phone and Modem options in control pannel after install.

I installed it on my 2008 server but it doesn't show in the Phone and Modem options.

Oke, then I will search further there. Another off-topic question, will there come a tapi for Windows 2008 (Terminal server) Server?

The addres is filled in but I replaced the phone number with the bold text (the bold text contains the phone number to dial but I removed the phone number). Doing this with another phone it works fine.. except from this snom 360 phone.

Hello, I'm trying to get the TAPI driver working. When I try to call using my SNOM360 th PBX log shows: [7] 2008/10/02 10:04:11: SIP Rx udp:127.0.0.1:1034: NOTIFY sip:127.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:1034;branch=z9hG4bK-r433tyk1h8osuw4fjcwz;rport From: <sip:32@hofstee>;tag=23487 To: <sip:32@hofstee> Call-ID: eilt76f47xpey71wsqio CSeq: 2343 NOTIFY Max-Forwards: 70 Contact: <sip:32@127.0.0.1:1034> Event: x-tapi Content-Type: application/x-tapi Content-Length: 59 Action: MakeCall Line: 1 Call: 250 Address: Phone number to call [7] 2008/10/02 10:04:11: SIP Tx udp:127.0.0.1:1034: SIP/2.0 200 Ok Via: SIP/2.0/UDP 127.0.0.1:1034;branch=z9hG4bK-r433tyk1h8osuw4fjcwz;rport=1034 From: <sip:32@hofstee>;tag=23487 To: <sip:32@hofstee> Call-ID: eilt76f47xpey71wsqio CSeq: 2343 NOTIFY Content-Length: 0 What could be the problem? the snom is not rinning and the Snom sip-trace log doesn't show the call.

Hello, I want to transfer a call to an extension that's busy, then I ask the caller " Do you want to wait?" they usually say yes, now I need to put them on hold and wait manually till the person I want to transfer the caller to is available, I want that PBX does that for me, when I press "transfer" and enter the number of an extension that's busy, the call must wait (now it just hang up the call, that's not quit friendly:-( ). Where to config this?

From extensin to AT it goes fine, but from the trunk it cuts off a part of the wave file. It would be a nice feature to set a delay in trunk settings or AT settings. I copied the "ringing" wav file to the AT's wave file 2 times, the first is now cut of and then you hear the 2nd and then the AT's wave file, the only thng now is that the caller can press 1, 2 etc while hearing the ringing sound.

I can't find (in PBX, Wiki and forum) how to set a delay before the AT picks up, now it directly answers but it causes things like somebody doesn't here the complete message. So I want to hear the " ringing" tone for e.g. 5 sec before the AT picks up, where can I set this option?

I turned of the offer camp, but now the caller is just disconnected, I want them to hear "All our extensions are busy at the moment, please hold the line" till the extension is available and then connect to the extension.

I would like to suggest to be able to write "scripts" for PBX. So you can for expamle build your own "telephone scripts" for AA's and so on something like aa.init; aa.play('a wave file') if key(#) goto call1 call1: while extension(3).calling do begin if busy then soundplay('busy.wav'); // the phone is busy while busy do begin soundplay('we_are_busy.wav'); wait 30s soundplay('stillbusy.wav'); end; end; ... Something like this... I think it would be very usefull to have an own Scripting Language inside

Well.. I don't know but here in Holland that's quit normal "just wait" and you hear moh and "THe extension(s) are currently busy please hold the line" and them MoH (you can probably include a feature that "THe que-time is longer then 15 minutes, please call back later - drop - this is used somethimes too)...

How can I disable the "press 1 to receive a callback" when an extension is busy, I just want the user to hear the MoH and hear the "the extension is busy at the moment" after some seconds, and then just wait till the extension becomes available. How can I configure this? if not possible, please include it, our clients are asking this many times:)

I just prefere an option to extension redirect when there is no phone connected/regged... instead of dropping the call

Yup I know, so my suggestion is now to make it possible to be able to personalize this global messages.

Ok I will try that with the hunt group. I know about the personalized AA message, but I want to have a personal message for "Please enter the extension number" wich comes when the user doesn't press a # after the personal welcome message, these messages must be different for each AA, I can edit that wave file but then every AA will have the same one.

Ok, then I think i will have to upgrade my license.. About the recording, Of cource I can overwrite the original prompts but what when using 2 auto attendends, then you hear "No input found, for ... pres ..." for two diferent AA's that will be a big mass:P so that's why I suggest choose a wav file or recording for the "please enter the extension number" per AA.

Are the following things possible with the AA? 1st. When the AA answers the user kan press 1 for e.g. Networking and 2 for e.g. Sales. - When 1 is pressed extension 20 will be called with dial-tone (e.g.) "Internal Call" - When 2 is pressed extension 20 will be called but with dial-tone (e.g.) "Tone 1" So the person can hear wherefor the caller is calling. 2nd. When the AA answers you hear a recorded message something like "for networking press 1" (in my case in dutch) but after this recording you hear "Please enter the exention number" can i replace this with a personal wave? ("No input given, for network press 1....")? would be nice:)

Also redirect on busy droppes the call, and also in a hunt group if you have 3 cases 34 35 36 and 35 is not connected, 36 will never be reached because the caller will be "dropped" at 35... What can I do... it's very frustrating....

The deveice didn't register yet (Yesterday I've updated the PBXnSIP software to the latest version) but it still doesn't redirect. Edit: The option "Forward all calls" is NOT set. I only set the redirect when busy and redirect when no answer. Maybe a suggestion to add an option "Redirect when offline/not connected" ?

When an external incomming call reaches my PBXnSIP it will be send to the first extension: 32 mailbox disabled, redirect after 10 seconds to extension: 33 mailbox disabled, redirect after 10 seconds to extension: 34 mailbox disabled, redirect after 10 seconds to extension: 32 ... This works fine, but when e.g. extension 34's network cable is unplugged the call will be dropped... instead of being redirected, how can I configure PBXnSIP so that the call will be forwarded if the extension is nog "connected/logged in"? to prevent dropping of the call. I tried it with an agent group but then the caller doesn't hear the phone ringing and all phones will go to ring, a hunt group did the same problem, when it reached the disconnected/unplugged phone/extension it droppes the call.

thx, FIxed the problem using IP filter list and ip routing list

Hello, The pbxnsip machine we're running is located at the office. Now I want to login to it from home with EyeBeam software so I configured it with the IP of the office at home and with the domain, registering is fine.. incoming (forwarded calls) are fine but when I call to an extension on the PBX I hear "nothing" but i see "Call established" on eyebeam. Calling a remote phone.. the phone rings, when anwer both sides hear nothing. Do I need to configure something for remote calling? (The firewall is configured correctly, even turning it off doesn't work).

Logging is enabled for everything, when I start the call: [5] 2008/02/21 15:24:35: Dialplan hofdial: Match 0613082423@10.38.53.199 to <sip:613082423@solcon.nl;user=phone> on trunk hoftrunk When the call hangs-up: [5] 2008/02/21 15:25:06: BYE Response: Terminate 3c26c0dfa4de-9l9vuvv37uql how can I do a sip trace?