Great Office - Hummig KG

Hello, in the Section "Night Service" (Agent Group Setup) there is an option called "When primary agents are logged out, send calls to...". Works fine, so we do not need to use the Service Flags ("who cares to operate it? Is still anyone available to take calls or not? and so on"). Better to use this nice feature and if last agent is logging out, the calls will go to the Mailbox automatically. When we've used the Service Flags in the past, we were able to see the Night Service Status on BLF. Is there any possibility to do this with the above feature, too? (Visualize if all agents are logged out?) If not, is it possible at least to see if a specific extension is logged in or not? Regards, Lucas

Also wenn ich das recht verstanden habe, funktioniert die Durchwahl 10 dann, wenn eine Nebenstelle die Rufnummer 10 hat (Telefon läutet, wenn man die von außen die Durchwahl 10 ruft). Und wenn statt der Nebenstelle eine Hunt Group die 10 als interne Nummer bekommt, funktioniert's nicht mehr. Stellt sich die Frage, was passiert, wenn man intern (also von einer anderen Nebenstelle) die Hunt Group mit der 10 anruft. Funktioniert das wenigstens? Wenn nein, liegt es nicht an der Rufverteilung, sondern den Einstellungen der Hunt Group (z. B. ServiceFlag gesetzt und Nachtschaltungsziel falsch oder nicht gesetzt etc.)

Any information I should know? I've the same problem with PBXnSIP. Regardless which Streaming Software I use (I've followed all the steps of your instructions exactly) I always get terrible noise when selecting the RTP Source an listen to the MOH. Music Streaming itself should be no problem as I'm able to receive Audio with VLC at the Machine where the PBX is installed. I wonder, why I also get this noise even in case the streaming source is stopped, regardless of the Port specified in the pbx (yes, I restarted service upon changing the port). Best regards, Lucas

Natürlich müssen da auch die deutschen Audiodatein vorhanden sein. Download hier: http://downloads.snom.net/snomONE/audio/all_audio_prompts.zip

Hello, we use the DISA-Feature every day on all our windows mobile phones. There is a excellent Software, called Magicall (http://www.mobiion.com/magicall.html), so we never have to care about DTMF-Dialling, PIN and more. When we dial a number with the mobile phone (regardless if number is manual entered, call-back from call-history or from contacts) Magicall cares about dialling into the pbx with caller-id enabled only for that call, pbx detects the mobile phone, grants access, so the mobile phone has to dial only "1" for the outside line and dialling the destination number with DTMF, followed by '#'. Of course, the mobile phone user doesn't have to care about this at all. Just have to wait a few seconds upon this fully automatic dialling in the background is finished and the call is established. It works perfect! Another (and much more efficient) way would be "block-dialling", also called overlap-dialling (https://www.vc.dfn.de/en/video-conferencing/ways-of-access/isdn.html). A few years ago, a german Least-Cost-Provides offered the following service and I'm sure the PBX could do this too in future. You just call the access phone number of the provider (or your pbx), directly add the destination phone number and then press 'dial' on your Mobile phone. For example: Access phone number of the provider / your PBX is +49 211 1234 and destination number you want to be conneted to is 0897654321, so you dial +4921112340897654321. The service provider recognized your mobile phone (and charges your prepaid-account), gets the digits '0897654321' (like DDI-dialling) and connects directly to the destination number. No greeting, no PIN and(!) no answer of the call before destination number is answering. Once the destination number is answering, the pbx answers the mobile call and put both calls together. If not, PBX does not answer the call thus you don't have to pay for this call at all. Very nice, if you call from abroad. Annoying if you are charged for every call to the provider/pbx, even the destination number is not answering afterwards. Unfortunately this service-provider stopped this great service long time ago. But I think that's not a huge work to do this with the PBX. For now I can dial any 'long DDI' (this works for domestic calls with at least 30 digits here in germany). Any such long number is detected infull length the SIP-Trace, so I is just a question of programming to get this 'DDI-information' and forward it to the DISA-Service for further processing.

Hello, I would like to have different greetings depending on the current call (Transfer to Mailbox because extension was busy / Transfer to Mailbox because no answer / Transfer to Mailbox manually / Transfer to Mailbox because DND). Is it either possible to set up more mailboxes, but collect all recordings to ONE mailbox, so it isn't necessary to collect recordings from different mailboxes? Or is it possible to transfer the call to the mailbox in such a way the greeting is selected by transferring to special destination (e. g. Mailbox extension is 123, transferring the call to 123*1 plays Greeting #1, transferring the call to 123*2 plays Greeting #2 and so on)? Currently the call is redirected on busy / no answer / manually to one and the same mailbox extension 123. But the caller really doesn't know why he reaches the mailbox. The greeting can only be universal "You have reached the mailbox because either the person is busy, not at his/her desk or in a meeting". Not very good.

That sounds really interesting. What about an Office Pro 25? Which Version of Snom One I will get with that key? And, are Dongle-Licenses supported by Snom One? Regards, Lucas

Thank you for your reply. Is the outbound proxy the one you told about? It was already set, so the question is still what about this strange 'asterisk' extension in the call history? Trunk 23 in domain localhost Name: Bellsip Type: register To: sip RegPass: ******** Direction: Disabled: false Global: false Display: Bellsip RegAccount: ****** RegRegistrar: bellsip.com RegKeep: RegUser: ****** Icid: Require: OutboundProxy: proxy.bellsip.com Ani: ********* DialExtension: 00 Prefix: Trusted: false AcceptRedirect: false RfcRtp: false Analog: false SendEmail: true UseUuid: false Ring180: false Failover: never Privacy: false Glob: RequestTimeout: Codecs: CodecLock: true Expires: 3600 FromUser: 00 Tel: true TranscodeDtmf: false AssociatedAddresses: 00 InterOffice: false DialPlan: Colines: DialogPermission:

Hello Everyone, obviously someone has hacked a PBX of our customers. There are a lot of fraud long-distance calls in the call history. Fortunately the damage isn't very much, as the Trunk is Prepaid. But now I'm really worry, because the Hacker didn't hacked the trunk account directly. He came into the pbx in a way I cannot understand. The attached call history tells me there were made calls from an extension called test (asterisk) but we do not have such an extension. There are a lot of calls to almost the same number 00442073479999 during the whole night. Searching this number by Google results in being a popular number for fraud. For my point of view I can only imagine, that the hacker logged into administrator account of the pbx, created a extension and made those calls and delete the extension afterwards. But this seems to be implausible. So my questions are: Does the PBX have a backdoor (pre-programmed extension) a hacker could use? Unfortunately I only have this call history and no other information. Is there a log-file telling me all sucessful and failed login attempts? Does the blacklist apply only on failed SIP-Registrations or also on failed Web-Login? As an emergency procedure we set all Dial plans to PIN Enabled (except those calls which are covered by flat-rate, which are the most calls).

Hello everybody, when I call my mailbox I'd like to be informed about the time of each recording. But when I set the option 'Play Envelope Information' I always have to wait for the lenghty caller's number. By pressing '#' the system wont skip the envelope information only but the whole message so I have no option rather than wait a half minute to hear the whole envelope information include the caller's number. Does anyboy knows either - how to setup the pbx, to play only the time of the recording as envelope information - which button to press in order to skip the envelope information, starting playback of the message immediately? Best regards, Lucas Hummig

We use the feature "Play envelope information before playing the mailbox message" in Domain Settings. Works fine, but how to skip the Envelope Message occasionally? By pressing '#' the whole message is skipped. Is there any Input for just skipping the envelope information and starting immediately the message? Regards, Lucas Hummig

Update: Seems to be a problem with the certificate itself. Tried to install another certificate (from my IIS7). This is accepted by the pbx and works (but invalid due to the wrong domain of course). I will request a new certificate but don't know how to get the PBX's certificate request I need to issue a new cert. How to get it?

I've spent hours to get the pbx running with certificates, but without success. I've copied the certificates (Base64) into the upper section 'Certificates' and the Private Key into 'Private Key'. After clicking 'Save' I see 'Starfield Secure Certification Authority' above the Text Field 'Certificates' along with a 'delete'-icon which might show a successful upload of the certificate. After rebooting the pbx, https access is no longer possible at all. It is really not a problem of browser certificate errors only. The browser won't get a https-connection to the pbx at all. But with http I get access the pbx again, in order to delete the certificate. I wonder why the xml-file in the certificate folder of the pbx doesn't contain the private key. Is this normal? What about the domain section of the xml file? Does it match anything with the domain of the pbx? In this section we see 'Starfield Secure Certification Authority' Any idea?

OK. Solved now. (Better reading all replies before adding comments) This was the solution: "This was a kind of workaround because most phones display only the "From" information, but you want to see who is being called. In a perfect world, the SIP phone would do the job and display both from and to. "

That's the same on our system. It's definitely a bug, because the number is shown when you choose in the Group's configuration to show only the caller-ID without Group name. Some of my Snom-Phones have a large Display. During Ring-state the Caller ID is shown in the bottom line of the telephone Display. But as soon as I lift the receiver this number is being replaced to the number the caller has dialed. I've opened a Support-Ticket #OIT-521482, but not solved yet. Obviously the 'From'-Field (SIP-Trace) has been changed from V3 to V4: V3-SIP Command: From: "GroupABC: (01234567890)" ;tag=60047 V4-SIP Command: From: "GroupABC:" ;tag=60047 If someone has an idea if there is any possibility to manually edit the From-Command, please let us know. Regards from Germany Lucas

Hello, quite easy: [1] Under: Accounts/Edit Agent Group/Algorithm/Number of agents added per stage - set the number of phones which should be ring from the first moment [2] Under: Accounts/Edit Agent Group/Ringback tone - choose "no ringback tone, continue to play music" Best regards, Lucas Hummig

We have one PBXnSIP License for Office Pro 25 (25 Extensions) left over. The license does have an upgrade-protection until Jan 2010, comes with a USB-Dongle and can be used on every computer regardless its MAC-Address. We will give off at a bargain price. Contact us by PM if you're interested.

Another question: Will it be possible to transfer a call directly to the MS Exchange UM Voicemail? Currently this works only by wasting additional extensions (licences!), which are set to "turn on Mailbox immediately". Transferring directly to the Exchange Trunk to doesn't work as Exchange will answer the call with "Welcome to MS Exchange, please enter Extension..." instead of routing the call to the appropriate UM-Mailbox because MSExch only looks to the "Forwarded from"-Tag which is the Ext-No. of the Switchboard, the call will be transferred from. A solution would be a setting "Select From-Field when transferring a call: From=To"

I've heard that queue positioning become available in 4.0. Is this still true? (The system should tell the caller "You are in position # in this queue". 3CX implemented this feature since a long time).

Could you tell me these both star codes? I didn't found them in the function list of 4.0 Beta we have installed and are working with. Thanks a lot!

Since some days, we have installed V 4.0 (beta). At extension level you now have the option to turn on cellphone ringing also on group calls! Works very fine!!

Sorry for the delayed feedback. I didn't got informed about your answer and stumled today over it. Let me ask you something: As the service provider does support T.38 by contract and the ATA definitely supports T.38 I'm searching the problem in the matter described at our WIKI: As soon as the receiver detects that the sender wants to send a fax, it tries to re-negotiate the used codec to T.38. With this change comes a change of the used ports, as the T.38 actually does not even use RTP. I do see that the AA detects the CNG Tone and "dial" the F which cause ringing the Fax ATA. But where (which log level / which messages to track) to check if the re-negotiaton to codec T.38 and change of the used ports happened or failed? Any special settings on the PBX necessary? Again, in the list of codecs I do not find T.38 but this seems to be normal, isn't it?

Hello, in order to intercom to another extension I need permissions in my extension to which accounts I'm allowed to intercom. No problem so far. Now I need to Intercom from outside (Parent's control over children). This doesn't work due to missing permissions when coming over the Auto Attendand. If I call the AA from my (privileged) extension it works fine, but not if coming from PSTN. Note: To prevent misuse I've set the direct destination at the AA to an 6 digit Number... Any idea? Setting the ANI to the extension with the privileges doesn't work. Any global setting disabling any restriction for intercom?

I've had the same problem. But solved it by setting the Snom-Phone to: Setup/Advanced/Behaviour/Enable Intercom = yes Or by butting the following command into my snom_3xx_custom.xml <settings> <phone-settings e="2"> <intercom_enabled perm="">on</intercom_enabled> </phone-settings> </settings> Hope this will help you, too. Regards, Lucas Humig

After some weeks of operation, we can confirm, that the described problem never recured since decreasing the number of extensions to 9 (and keeping 1 extension unused). Hope we can use all extensions of licence in V4.