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Great Office - Hummig KG

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Everything posted by Great Office - Hummig KG

  1. How to check, if the Server the pbx is installed to (same Hardware, no changes) can reach the license Server? Is there a URL I can lookup in the browser or ping a Domain Name in order to exclude there is something wrong to reach the license Server?
  2. Hello, after upgrading to V5.2 the pbx asked us for Activation Code in order to retrieve the license. But it seems, that after pasting the Activation Code and pressing the "Send" button, nothing happens, even after waiting some minutes. We downgraded to V5.0 but still missing the license and aren't able to activate the System. Fortunately, this is only a test machine, but in productive Environment we would now running into disaster. For the future: How to backup the license, in order to prevent such a blackout after unsucessful activation issue? Just backup the pbx.XML? In case this is an issue with unsufficient Maintenance Period? Is there a difference between "no licence" (shown) and "invalid license / unsufficient Maintenance period" Where we can see, if there is retrieved a licence at all? Where is the license stored on the Server? Regards, Lucas
  3. Thanks a lot! Any Idea how to secure that Feature from abuse? Is there a way to allow only trusted callers to dial-in?
  4. Fine, we have a V5 System (Sandbox Server only) an will move our V4 there. So I understand, all we need are just these two Settings, only. Thus, if the DID matches to an Extension, the Extension is called. If not, the number is Interpret as an number to be dialled according the dialplan, isn't it? Can the "Accept Redirect" be kept to "No"? Or is that Feature based on allowing redirections? How to set, that not anybody is allowed to use that way of making calls? The field "Explicitly list addresses for inbound traffice" certainly means IP-Adresses rather than telephone numbers which must match. Only a few mobile phones should be allowed to use that trunk. Is there a way to White-list them?
  5. I found the Setting "Inter-Office Trunk" but neither "Trunk may terminate calls for remote Systems" nor a Dialplan-Selection (Version Are these Settings a Feature of V5 or can be found elsewhere than in trunk-Settings?
  6. I had tried a few things, but didn't got ahead. I think the Problem is, that an incoming call can be routed to an Extension only, no matter which dialplans (even inter-Domain settings) are set. Under 'Routing/Redirection' in the trunk Settings, the Manual tells me I can only enter Extension numbers. Again, we are searching for making Callthrough-Calls from our mobile-phones by dialling the Destination number as an DID. Assume that our Office-number is 089-5555-0 we would like to call 089-5555-0176123456 in order the pbx detects the 'DID' 0176123456 and redirects our incoming mobile-call to that external number. I think, the first step should be, to get an incoming call being forwarded to a pre-defined number (for testing). If this works, than we can care about forwarding individually by DID. I fail with that first step. If I type-in the test-Destination number any incoming call should be forwarded to, into the 'send call to extension' the call Fails, because there is no Extension with 0176123456. I'm sure dialplans will be ignored, as the System is only searching for extensions and doesn't cares about dialplans for Routing incoming calls, isn't it? What about the Redirect Feature? I found some description in the WIKI, but didn't understand how the redirection works. Is there a way to get the incoming call redirected to a pre-defined number or better the dialled DID (which is an external number)?
  7. Yes, that's sounds very good and it might be exactly we are searching for!! Any hints how to set this up (some Special Regex's)? Is there a way to proof incoming caller ID, so only registered callers can use that trunk-to-trunk Gateway?
  8. Sorry, I've mixed up "Dialling own Extension" and "Calling Card", but the question is still the same: Directly callthrough without Authentication by PIN, just by Caller ID and Destination-Number extracted from excess-length DID instead of entering via DTMF. If I understood you right, there is no way to do that?
  9. Hello everyone, we are using the PBX to make calls from our Mobile Phones (GSM-Cellphone) through the PBX in order to show the called Party the Office-Phone Number instead of the mobile Phone Number. We call the PBX, then press 1 for Outgoing Call, then type-in the Phone Number we want to be connected to and press #. Some Calling Card Providers offer Callthrough and we think it might be possible to run the same Service with Vodia PBX: Let's say the Calling-Card Extension is +498912345-678. Instead of calling that Extension and get the announcement "Press 1 for Outgoing Call" and so on, we dial +498912345678901761234567 The PBX gets that overlong DID and retrieves the Digits 01761234567 as the requested destination number and directly forwards the Call to that number. No message played, no DTMF Input neccessary. In case the Destination Number is busy the call isn't answered (and forwarded) through the pbx and the Calling Card User gets the Busy tone. Is there a way to configure the CallingCard Extension in such a way (e. g. Dialplan Regex)? Regards from Germany
  10. Sorry, found myself: PBX-Administration-Level: eMail / Messages: 'When a registration changes its address but keeps the call-ID' was set to yes! That's it!
  11. Hello, I've set up an extension for a doorphone which works very fine. But every two minutes I get an email, telling me: Source address for "2N EntryCom IP Uni" <sip:19@mypbx.com>;tag=22804 has changed from udp:xx.xx.xx.xx:44710 to udp:xx.xx.xx.xx:44712 (Different ports every time, of course). I've checked all the settings to stop these emails, even restarted the pbx but without success. Since a few days I've got thousands of emails. I've checked these settings: Extension-Level / eMail-Settings: all eMail-Notifications are disabled Extension-Level / Registration-Settings: Log registration change = no logging Domain-Level: No settings found to change this behaviour PBX-Level: Status / Logging: All Log Levels = 0 Important / SIP-Logging = All NO Any idea which setting I've missed?
  12. Again, we have the same issue. We got just informed that there are registrations from the United States (noone of our users is there) are done. We get this message (where x.x.x.x is the IP of our PBX): REGISTER sip:x.x.x.x:5060 SIP/2.0 Via: SIP/2.0/UDP;branch=z9hG4bK5522310192a69a552ef6db8;rport From: "5005" <sip:5005@x.x.x.x:5060>;tag=552231082cf To: "5005" <sip:5005@x.x.x.x:5060> Call-ID: 310192a-10bf69a5-52ef6db8@x.x.x.x CSeq: 1 REGISTER Contact: "5005" <sip:5005@> User-Agent: VaxSIPUserAgent/3.1 Expires: 1800 Max-Forwards: 70 Content-Length: 0 I've I understood you right, the only way we have, is to keep our passwords secure. Is there a way to retrieve from the above message, which account of which tenant has been cracked? Not one of our multi-tenant pbx has an accout 5005. Do you mean settings made on the Trunk-Provider Side or settings on the Trunks of the SnomOne?
  13. Wie sind die Einstellungen bei Sipgate? Ist dort eingestellt, dass die anzuzeigende Absenderrufnummer von der Anlage kommt? Wenn bei Sipgate eine feste Absenderrufnummer eingestellt ist, werden die Einstellungen der Anlage ignoriert.
  14. Excellent. I'm sure this helps. Was that insider-knowledge or can we find those informations in the wiki or elsewhere?
  15. Where can I find more details about the Syntax of the "URL for sending text messages to cell phones" (at mypbx/reg_messages.htm)? We have a SMS-Gateway installed, supporting HTTP-API (http://www.diafaan.com/applications/web-service-sms-gateway/) and I'd like to use it with our pbx.
  16. We experience strange Problems when updating to 5.1.3. After Login the browser opens http://localhost/reg_domains.htm and shows us crazy stuff like this: #2 help:ADOlis1 #3 #localhost #reg_edit_domain.htm 1 help:ADOedi1#reg_edit_domain.htm 2 #reg_edit_domain.htm 7 #reg_edit_domain.htm 8 #reg_edit_domain.htm 9 #reg_edit_domain.htm max_extensions #reg_edit_domain.htm max_attendants #reg_edit_domain.htm max_callingcards #reg_edit_domain.htm max_hunts #reg_edit_domain.htm max_hoots #reg_edit_domain.htm max_srvflags #reg_edit_domain.htm max_ivrnodes #reg_edit_domain.htm max_acds #reg_edit_domain.htm max_conferences #reg_edit_domain.htm max_colines #reg_edit_domain.htm max_calls #reg_settings.htm max_regs #reg_edit_domain.htm admins #reg_create_domain.htm 1 help:ADOcre1#reg_create_domain.htm 2 #reg_create_domain.htm 8 #reg_create_domain.htm 9 #reg_create_domain.htm file By exchanging pbxctrl.old everything is fine (V 5.1.2). I wonder, why the filesize of the 5.1.3 is less (9.252 KB) than the 5.1.2 (11.020 KB). Exchanging the exe-file back causes the same error again. Even a complete deinstallation and new Installation starts with the above issue. Never had a Problem after updating before. Any help? We've installed the 64-bit Version on a Win2008R2 System...
  17. If the VaxSIPUserAgent would be one of our registered extensions, everything would be fine and the pbx's report changing of port could be ignored. Unfortunately neither the client's IP-Adress belongs to one of our branch offices nor does that extension exists on the pbx. So due to this change in port Information we stubled about, there is something (successfully?) registered on our pbx we didn't had allowed to be. Never heard about a VaxSIPUserAgent.
  18. Hello everyone, I'm getting lots of eMails from our PBX with following message (where xx.xxx.xxx.xxx is the IP of my PBX): Source address for "205" <sip:205@xx.xxx.xxx.xxx:5060>;tag=8351913a6a365 has changed from udp: to udp: The attachment enclosed shows: REGISTER sip:xx.xxx.xxx.xxx:5060 SIP/2.0 Via: SIP/2.0/UDP;branch=z9hG4bK8351913a65d52461352a4d206;rport From: "205" <sip:205@xx.xxx.xxx.xxx:5060>;tag=8351913a6a365 To: "205" <sip:205@xx.xxx.xxx.xxx:5060> Call-ID: 13a65d52-c9894613-52a4d206@xx.xxx.xxx.xxx CSeq: 1 REGISTER Contact: "205" <sip:205@> User-Agent: VaxSIPUserAgent/3.1 Expires: 1800 Max-Forwards: 70 Content-Length: 0 So far nothing fancy, except, that there is no extension 205 at all! A few minutes I get the same eMail, but with other extension numbers all not set up on this PBX. I've checked the call history and there are no calls from those extensions, but I get nervous that obvoiusly someone is hacking my pbx. How to get rid of this situation? Is it possible that someone can register an extension just by SIP-Connect without creating the extension before at admin's interface? Regards
  19. Hello, the problem seems to be solved. Suddenly, the issue disappeared and we might have found the reason. An Android Smartphone was registered to the extension, which had this issue. Obviously the registered smartphone didn't answered fast enough or not at all due to slow internet connection (GPRS). Another phone had a registration to this extension too (forgot that this was done for experimental purposes a few month ago). So it seemed to be a problem of the whole system, but wasn't. LH
  20. Sent the full Logfile as PM. Operating system is Windows Server 2008 R2 Standard SP1. (VMware Environment)
  21. Thanks a lot for the fast response!! I think we can exclude the SIP-Serviceprovider for this issue, as we get the problem on all SIP-Trunks (different providers!) as well as internal calls, too. Again, incoming calls aren't interrupted during Auto-Attendand and Queue. The 38-seconds begins as soon someone picks up the phone. Also it is neither only a problem of the SIP-Phone nor only a problem of the local IP-Infrastructure (LAN / Gateway). We have the issues on different phones and different locations (and different public IP-Addresses). Even the call is answered on mobile-phone (meaning without any LAN, only Trunk-in Trunk-out) we have the same situation. But we think that obviously the issues are only on one (first) of four tenants. Currently we are testing this deeper. We've set a multi-tenant PBX. By the way, we have a 2011- Coma Berenicids (Win64) (SnomOne Blue) running. I've attached three Logfiles: Logfile_Internal_Call.txt One extension called another extension. Call was disconnected after 38 seconds after the call was completed. Logfile_Trunk_Call.txt An external call (from pbx into pbx using public phone number and connected over Auto Attendand). Call was disconnected 38 seconds after the call was completed by extension Logfile_Test_Call.txt One extension called a Agent Group. No one answers the call, caller hears Music on Hold. Call isn't interrupted at all. Neither restarting the PBX, nor restarting the Windows OS (Firewall disabled completely) does resolve this problem. Datacenter Support assure that they didn't made any changes and don't limiting the server communication in any way (e. g. port restriction). Never had such a big problem before. We are using SnomOne / PBXnSIP for five years without any bigger problem. Hope you can help me! Best regards Logfile_Test_Call.txt Logfile_Internal_Call.txt Logfile_Trunk_Call.txt
  22. Hello, since a few hours we have EXACTLY the same problem. All incomping calls terminate after 38 seconds and we see it is caused by "The call port 104 was erased forcefully". The system is working since many months without any problem. Absolutely nothing has been changed. PBX is running in a datacenter so we do not have any access to routers and gateways. OK. I see it might be an issue at the network environment and I will inform the tech-support of the datacenter. It doesn't seem to be a problem with the trunk provider. The incoming caller can wait some minutes in the queue, but as soon an agent takes the call, the 38 seconds are counting after the call will be disconnected. Any other ideas?
  23. Hello, really no one, who uses this callback-feature and can tell me if it is really working or notor if I do something wrong?
  24. Hello, We'd like to Setup a calling-card account in order to be able to request a callback, without paying for the call to the system. The Snomone-Manual describes exactly what we want to do (Page 179 - "Callback"): "The User calls into the system and request a callback. Although this mode requires more steps to establish the call, it may reduce telephone costs even further, as the call to the system does not get connected and the caller is not charged for the call." For my understanding the Manual does not answer the question if the system accept ANY callback request, meaning, it will callback any number which makes a request-call. If only pre-registered callers are able to make a request-call, than the question is, where to enter this Caller-ID's. We aren't able to set this up. When we call the calling-card-account, the call IS answered and we're requested to enter a PIN. Even in Domains, where the mobile-phone's number is recognized properly (Personal Virtual Assistant) the behaviour is the same. Does anybody have tested, if the system really offers this functionality described in the Manual?
  25. I was really surprised to read that it would be possible to manually change the status of the service-flag while it is in Day/Night-Mode. This was a feature request I know since PBXnSIP-Times. Always they say it isn't possible, meaning EITHER automatic OR manual. I've checked this with V2011- and as soon the Service-Flag is set from Manual into Day/Night-Mode noone can change the status manually. If the service-flag account is dialled by any user, the PBX is telling that the Function isn't available. My customers always used the workaround using (wasting) three buttons: (1) Day-Night-Mode (2) Forced Day-Mode (and ignore (1)) (3) Forced Night-Mode (and ignore (1)) As far as I know, the situation is the same even in V5.
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