Great Office - Hummig KG

A standard feature, available on all Panasonic PBX's is to answer to a paging-call. Someone pages a group of Extensions and calls for a staff and remains on the line. The called staff pick up the phone, dials a feature code, the page call is converted into a standard 2-pty phone call. Any similar function availably on pbxnsip-system?

Hello, I'd like to set up a solution for the following request. The pbx Auto Attendand can be called over a 800-Number (Toll Free). Just by dialling a Direct Destination I'd like to be asked to place an outbound call, without asking me for my extension and pin. For a minimum security, the direct Destination is not an 1-digit but 4-digit. I tried already: - Calling Card Account as Direct Destination, but Disadvantage: For my view Ext/PIN Authentification cannot be disabled (Any idea?) - Direct Destination set to *91-EXT-PIN but that wont work, I think the WIKI has a type-mismatch (http://wiki.pbxnsip.com/index.php/Basic_Calling - Dialling an outside Number with Prefix *91. This feature Code is default set to 'Add white list'. Not sure, if the descried Feature is available at all anymore) - Outbound loop call back to the pbx with ANI Number matching and providing me the DISA Menu. Works but waste line capacity and does have poor quality due to the echo on the additional loop call Any ideas?

Is there any possibility to get this nice feature working on group calls?

First I thought you've misunderstood me, but you're right: There is a XML-Tag 'button_speed_on_idle' which is default off. That was I was searching for. Thanks a lot!!

OK. What happens, if you create more button-profiles? Are they added to the empty fields or will some of the empty fields be renamed? Maybe a cleanup of the appropriate xml-File is necessary ;-)

For my customers, working up to now with traditional PBX's they are familiar with buttons which - blink, if the corresponding extension gets an call, by pressing the button the call can be picked-up (also possible with pbxnsip 'monitor extension') - light, if the corresponding extension is busy (also possible with pbxnsip 'monitor extension') - calls the extension, if the button is pressed while the extension is idle. The last I cannot setup in pbxnsip. I can assign a speed-dial without monitoring the extension or assign a monitor-button with nothing happen when pressed (extension is idle). Any idea to get this basic-function - available on every old pbx - work?

Obviously you have created some empty Button-Profiles. Go into the Domain-Setup / Settings / Buttons and delete the empty button-profiles.

We need a solution for those customers who intent to exchange from traditional PBX to SIP-PBX and want to benefit from the saving of SIP-Trunking. - Receiving of Faxes is essential - Faxnumbers are some extension numbers of the trunk, therefore one single Trunk-registration for the whole external communication (phone and fax). No possibility to divert fax-numbers for separate registration or remain them on PSTN-Lines (Example: 4885-0 (Telephone Switchboard) 4885-112 (Phone Extension) 4885-129 (Fax Extension). Trunk registered: 4885) - No PSTN available, No PSTN-Gateway applicable - SIP-Trunk Provider guarantee to support T.38 (QSC AG - Germany) We absolutely do not understand, how to switch to T.38. Of course, it should depend on the DDI called (Fax-Extensions). - How to setup the pbx in such a way, that certain Extensions (example above: 129) are communicating only T.38? - We tried to setup Exchange 2007 for receiving faxes, but we only get voicemails with the "fax-beep". Is there a possibility, to force to T.38 at Exchange? - As a workaround: We tried to forward all fax-calls to a complete different fax-numer on a seperate PSTN-Line. No way. The whole communication goes through the pbx instead of being rerouted from the SIP-Provider directly to the 3rd-pty PSTN-Fax-Machine. Any possibility to deflect the call out of the pbx? - Does anyone know a 3rd-pty Software to setup between the SIP-Provider and pbxnsip which is able to detect and handles Faxes and divert such calls away from the pbx? Any idea, hint or workaround? Without a solution we seem to be incapable to sell pbxnsip.

Thanks a bomb! I'm sure I gallop through this page, but read this matter over. Excellent Help!!

Hi! I really get crazy with that ringtones. I cannot follow where the ringtone-files comes from, where they are stored and how to get them to the phone. And: If I understand right, all of the Ringer-Settings of the Snom Phone are are circumnavigated as the phone is instructed by the pbx to play GUI-sounds instead of triggering the ringer according to the phone settings. [2]4/5/2009 16:16:37: gui_object:: Received Alert Info alert_info_host=127.0.0.1:80, alert_info_file=/Bellcore-dr2 [5]4/5/2009 16:16:37: Dialog -10/5 going to early [2]4/5/2009 16:16:37: gui_object::synthesizer: Playing Bellcore melody [2]4/5/2009 16:16:37: gui_object::synthesizer: Playing Bellcore melody [2]4/5/2009 16:16:37: gui_object::synthesizer: Playing Bellcore melody [2]4/5/2009 16:16:38: gui_object::synthesizer: Playing Bellcore melody Is there any possibility to disable this on the pbx and just let the SIP-Command "BELL" triggering the ringer? The unattractive side-effect is, that the phone currently bells furious in the first seconds due to the quad GUI-Command (see above). No idea, where to find the reason for that behaviour. It occurs only on Agent Group Calls, normal Calls will only raise one GUI-Command

Good idea! But where to find a template or a description for that file? It isn't available on my system and I have to create it manually. And: where exactly to store this file in order that works? Root directory? \pnp_parms? Could you provide me a sample file? Thanks a lot

Hello, did anyone had success in selecting different Ring Tones on Agent Group Calls or Hunt Group Calls? For me it doesn't matter which ring tone is selected, always the same Standard Melody is played. I wonder where this ringtone comes from, because it is none of the 1-9 Snom's Melody which can be triggered for demo-play. I tried all settings in pbxnsip as well as all Settings in the User-Interface of the Phone Handset. Changed the Ringtone in the SIP-Identity, changed the Default Ringtone in the Preferences Menu as well as all other options in this tab like special ringtones with alert-info internal/external/group. But now I found something interesting in the Phones SIP-Trace. What the hell is: [2]3/5/2009 23:20:29: gui_object:: Received Alert Info alert_info_host=127.0.0.1:80, alert_info_file=/Bellcore-dr2 Should it be really neccessary to tinker within the millions of expert-settings (alert-info / specify ringtone-files) of each snom phone rather than just select another ringtone in the pbxnsip menu in order to get different ringtones? I remember I've heard some day a different ring tone (maybe on previous software versions on pbxnsip) but cannot remember if I had made special settings accidentally. Would be very nice to be able to distinguish different lines audible. Regards from Germany, Lucas Hummig

I've read the above mentioned description, but didn't found a solution to setup the auto-Attendand in such a way the call is answered, than nothing or just a beep is played and then a fixed UM-Voicemailbox is called without any other interconnected messages or user intervention by pressing any buttons. I need to setup as much auto-attendands as I need UM-Voicemailboxes for direct transfer. Any idea?

I have the exactly same Problem: Transferring Calls directly to VM (Exchange) costs me a 2nd licence for each user. All the following attempts were in vain: - Creating ONE Extension for all Users, but adding ALL VM-Numbers in the Field Extension Number, separated by space Didn't work: The Call is always forwarded to the VM of the FIRST Extension Number only - Creating Hunt, Group, Auto Attendand, Agent Group with forwarding to Exchange (Dial plan set to use the Exchange trunk) It seem silly to attach a 3CX as an sub-pbx in order to do not waste Extension-Licences... So I think about another method of resolution: Is it possible to just insert the follow SIP-Command in the Trunk-Setup in some way (and create as much trunks as we need Exchange UM-Users)? Diversion: <tel:12345>;reason=no-answer;screen=no;privacy=off (where 12345 is the UM-Mailbox-Number) (I've heard people working with Asterisk can do this without any problems in their pbx)

*** NEW DETAILS ON THIS PROBLEM-REPORT *** In the History we found this entry: [0] 2009/04/22 23:46:58: License suspended: There are too many extensions [0] 2009/04/22 23:47:01: Cannot send packet: No license We definitely didn't made any changes on the Extensions. After restarting the service, the licence informations are shown as follows: Extensions: 10/10 Accounts: 107/300 As a workaround we've deleted the PAC-Console-Extension (so we have only 9 Extensions of 10 used) and hope the error won't appear any more. But obviously there is a bug counting the number of extensions. Again, without deleting the 10th Extension the PBX will work some time... Any idea?

Win 2003 R2 SP 2

Since Upgrading to 3.3 we experience repeatedly a full system outage due to missing licence. We can see the licence code and it is correct. But above the licence code we read: Current license state: No licence License duration: Permanent All we need to do, is just restarting the service, than the licence is coming out. This works for app. 3 to 48 h. What's the best practise to localize the problem? - The history is full of non-relevant events and messages: Which Level is appropriate? - It seems that the system misses the licence from one second to another. - It seems to have no bearing with a 3-minute-limit or similar. - We have an Dongle-based NFR-Licence with Upgrade Protection Jan 2010 Maybe a defective Dongle could cause this problem? When the sytem "hangs" again, we only see there is "no licence". No idea to check if the dongle is "active". Any tool to do this? Regards

Sounds promising... For now I solve some requested feature by connecting a sub-PBX (3CX) registered as an extension on the pbxnsip and forwarding the calls to my mobile. This has the advantage, that as long the cell-call lasts, my extension is busy and the next incoming call is hold in a queue. As soon as my cellphone is free, the next call is forwarded to it. So the question is: When the V4 can be expected approximately? Will it provide hunt-group calls be also routed to cellphones? Regards, Lucas

Sounds excellent. Just for correct understanding: Is it possible in this way to write some special hard-coded addon-settings into an XML-file put in the tftp directory which contains not the complete phone's setting but just these special settings (e. g. backlight always on) which are processed for all phones of this model, AFTERWARDS every provisioning of the phones according to the current configuration done through the pbxnsip web-interface?

I tried this with binding to each other identity, but without success. The problem must be elsewhere. Now I tried to dial *85 and I get the announcement "Call cannot be parked" too. I compared all the remaining SIP-related settings on the Snom Phone for each identity and they all seem to be the same... Hold it! In the meantime I got help on another topic (Multiple SIP-Registrations by pnp-Settings). NOW IT WORKS! It depends wether I set the additional SIP-Registrations by hand on the SNOM-Phone or by pnp-settings of the PBX! The buttons remain "to all SIP-Identities on the Phone" and it works...

Oh yes! These demo effects! I'm so sure I tried to store more MAC-Addresses seperated by space and I got always the message that the MAC-Address is too long. Obviously I've got two spaces during copy-and-paste and didn't recognized that. Thanks a lot! Selecting the SIP-Identity for outbound calls are really no problem on SNOM-Phones. Either select them or set a default and deny selecting SIP-Identities. But you are right. Call Pickup and call parking get very messy, especially using buttons.

I think I've heard somewhere, that I can assign more MAC-Adresses to one extension, so this extension becomes available on more than one device. Is it possible to set the 2nd, 3rd, 4th SIP-Identity on SNOM-Phones over PnP-Mechanism of the PBX or do we really have to manually login to every phone and do the settings by hand? What we intend to do (maybe anyone has an better solution for that?) is: - assigning on every phone a 2nd and 3rd (...) SIP-Identity, so that two extensions of the PBX are registered by every phone in the office as additional SIP-Identites. - When a call is received on one of this multiple registered extensions (each one is the only one agent of an Agent Group), all phones are ringing at the same time (I know: this can be done much easier), BUT: As long as this call lasts, the appropiate agent group is busy and the next incoming call is put in the Queue, REGARDLESS on which phone the call was accepted. I do not have a better idea, since the following alternative solution don't solve the problem: If we do it without multiple registrations, and put each phone with it's different extension number into the (personal's VIP) agent group, the other phones would ring on the next incoming call to this agent group (which is used only for one special Person). There seem to be no way to set the Agent Group "busy on busy" (or maximum Calls on this Agent Group = 1). Any idea?

Now I have the half solution found myself. There is a totally different behaviour between setting up a Park Orbit Button in the Web-Interface of the SNOM Phone and setting up a Park Orbit Button in the Button-Settings of PBXnSIP. The last does something magic: Pressing the button while a call is active, it dials *85EXT (where Extension is the specified Park Orbit Extension in the Button Settings of PBXnSIP). Pressing this button while a call is in the Park Orbit causes the phone to dial *60RANDOM (where RANDOM is a irreproducable Number) I absolutely cannot understand, but that work. BUT: Now I still have an unresolved problem. This only works as long as calls received on SIP-Identity #1 of the SNOM Phone. On all other Identities (same PBX of course) I doesn't work. I always get the voice-message that the call couldn't be parked. Any idea?

Hello, within the Web-Interface of every SNOM-Phone there are millions of settings which cannot be set either in the phone nor in the pbx. I wonder if it is possible to implement some of these settings (e. g. Backlight always on, LDAP-Settings) into the provisioning files automatically created on the PBX, without editing files for each extension. I think I have heard anywhere that it is possible to edit a template file, which is the basis of the system-created provisioning files. Any idea? Regards, Lucas

Poor workaround: Create a trunk, register a extension at this trunk (loop connection), dial out over this trunk so you'll present your office trunk number and no longer the Cellphone's. Disadvantage: Increased Echo.