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Everything posted by Jan Boguslawski
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Hi @all please imagine a situation were user's work sometime's from home and like to use their own fixed or mobile phone to do the office work. This user's also like to use remotly the TAPI or Remote Call initiation via Web-Links like described in the wiki topic Click to Dial. Should this users be configured with Hot Desking to their home / mobile phone numbers or do we need to forward all calls to the number preferred by the home office worker? What might be the best way to provide the users a self maintenance service for changing their account configuration for this home office situation and back to standard office use at any time? In both cases: TAPI and Click2Dial-Web-Links is it simply a call (sip-invite) made by the system to the user? Thanks, Jan
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P-Asserted Identity / Outbound Caller ID Failure
Jan Boguslawski replied to Alex Kasperavicius's topic in Trunk Setup
Hi I agree with Alex. It seems since 3.1.1 some major changes had been implemented. This "country code" + ANI + "global number rewrite" features seems to complicate things (just my point of view,as I didnt understand it atm.) But what is very strange to me, it becomes impossible to recreate the "good old" behavior from older versions. An example: If you take a look at the wikipage for OCS the trunk settings for: With this settings it was possible to pass the OCS-Tel-URI via pbxnsip to another trunk for example AudioCodes Mediant 1000 VoiP gateway. So in PSTN your DID arrived at the callee. But now the number from the setting "Assume that call comes from user " is passed for every OCS user or the OCS Conference and Exchange Play on Phone, etc. Since 3.1.1 this seems not to work, or I simply dont understand how to recreate this behavior with the new features... Please advise! Thanks. Regards, Jan -
Hello Kaj, Hello pbxnsip, I guess we can find the solution for this challenge in this AudioCodes Document about: Configuring AudioCodes Gateways to Operate in TLS Transport Mode with Microsoft™ Mediation Server In step II it tells about the Mediations Server Preparation, and a hotfix for this TLS Setup. (it is not MTLS like normal between all OCS Roles) My verdict is, in your case: 1. OCS is doing an outbound call and presenting its certificate. Pbxnsip is simply accepting it. Does it make an crl (Certificate Revocation List) check to the rootCA? I guess not. 2. But OCS is not accepting the certificate presented by pbxnsip. Maybe the hotfix will ease this. Please give the AudioCodes document a try. btw: I am sure you dont need to buy a commercial certificate! If it does not work with the windows CA, it will never work with a payed one! Lets save the money Best regards, Jan
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Hello Tim, please carefully check / filter the OCS Event log in the event viewer for Event-ID 21034 You should find a lot of entries telling: I am sure the BPA report has had related warnings on that events. Please tell me about your findings. Regarding your other topics: yes all that you asked for is possible, but... It is a long story, with pros and cons and a lot of different approaches... I explained one here in another post. Please excuse that I can't go into all details here, it would be to much to write... Are you sure, you are aware of the headaches that Microsoft Office Communicator Phone Experience Device (Yes it is the official product name short term MS MOCPED )phones can cause to your colleagues and especially to you or Eric??? ;) Please be careful with that! If you think this is a must have, I strongly recommend you to evaluate and test a lot, also with your team! Regards, Jan
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Hi @all Unified Communications friends I like to spotlight you on an issue with OCS R1 and R2 regarding a call drop after exact 30 seconds. This happens in some integration scenarios with IP-pbx, some VoIP-Gateways and maybe also with VoIP-providers. It seems to be related how OCS components / roles check RTP streams on silence (both directions). One example of that can be found here with Cisco Unity. In pbxnsip you can face this problem e.g. when you put an external caller on hold in Office Communicator or R2-Attendant Console. Typically Mediation Server will send a bye after the 30 sek. Drago (in the forum discussion about the Unity issue) has an interesting explanation: Snom with their native OCS Edition phones also faced that problem in a similar way. But they have find a workaround J They are sending short none audible audio during the 30s. Maybe this workaround can be featured at an special pbxnsip trunksetting too? Or maybe a OCS native solution, which e.g. is done by AudioCodes Gateway's for Microsoft UC (sorry I dont know the details atm. but I can provide ACsyslog traces, where you might see what is happening if you put someone onhold in a direct OCS Mediation <-> AC Gateway scenario. I think they dont send none audible audio. Regards, Jan
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Hi Tim, you are welcome! Thanks for your compliment. OCS / Exchange / VoIP / pbxnsip / snom / AD / DNS / ISA etc. are just my "bread-and-butter" Please download, install and run OCS Best Practice Analyzer. from MS download center. When you run it the first time, please make sure you run it as Administrator and that you check for updates (inside the tool) before your first analyze run. The update step will download the latest info's about OCS R2 into the BPA. You run R2, don't you? Beside a lot of warnings and errors, the report results will warn you about telephone numbers that could not be "normalized". The Report will guide you to a file which contains all the "wrong" formatted numbers in your Active Directory. I guess you will see here all of the bussiness numbers starting with (301)... and the Mobile Numbers. Please update me about the results. Regards, Jan
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Hi DWAyotte, you are welcome. Sounds strange for me too. Please let's clarify the situation you are facing: You start a call via a hardphone (sip-phone registered at pbxnsip???, which model & brand?) and the callee! picks up? (what kind of callee -pstn, pbxnsip, ocs-user?) In that case your phone continous to ring? And the same happens when you start the call via communicator? Please try to explain your cases more detailed and describe who is calling who and with what kind of device/software! Regards, Jan
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Hi Tim, please add a leading plus in front of you number at the static registration in pbxnsip and at the tel URI at the OCS user account Enterprise Voice settings. Should look like: 1. sip:+2007@ocsmediationserver.domain.xx;transport=tcp 2. tel:+2007 OCS Mediation Server needs the E.164 format with the leading + As my example shows, you dont need to provide a real and complete international DID number with national digits etc. , but you need the plus! Additionally please start logging at the mediation server before testing and Analyze the Log in OCS Snooper Tool! Regards, Jan
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Exchange SP1 UM Voicemail Not Working.. Help!!
Jan Boguslawski replied to Eric's topic in Microsoft Exchange
Hi Eric, I think you can easily solve this problem using e.g. 9999* instead of 7* The 7* is just an example and my 9999* is also just an example. This is how we use it. No conflicts so far (since 2006 with MS Exchange Unified Messaging beta3 Ohh How time flies!). btw: dont forget to assign the exchange trunk another preference in your dialplan. best regards, Jan -
Hi Michael their are no silly questions, only silly answers What about an Snom 820. http://www.snom.com/fileadmin/user_upload/...820_frontal.jpg It's extremly flat and when you remove the food it would fit perfectly at walls, I guess. The handset would still be secured by a small pin. btw: I like my 820 on my desk best regards, Jan
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Hi deva, from what you described, my first impression is: 1. maybe a little misconfiguration in your setup. Did it ever worked the way you expect it? 2. sounds like a wrong or no diversion header info, so Exchange would not be able to recognise this is a redirected call and simply receives a call. Thats why it ask's for the PIN. Did it also ask for A's extension ??? Please describe your configuration more detailed! Maybe compare it carefully with the wiki page: http://wiki.pbxnsip.com/index.php/Microsoft_Exchange Additionally: enable Unified Messaging Diagnostic Logging at Exchange UM via Powershell: http://technet.microsoft.com/en-us/library/bb430783.aspx If I remember correctly you can do e.g. a one-liner like this: get-eventloglevel *UM* | set-eventloglevel -level Expert To change all UM related loggings. Please give it a try It's like voodoo Or try this one get-service | restart-service -force Best regards, Jan
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Hi Dave, No problems so far, we are on Rollup 7 now. But I share your precaution, everytime a new Rollup is released When will this product be RTM ? Best regards, Jan
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Hi Quentin, if you have OCS R2, why you dont try using the R2 Response Group Service? You can easily configure Workflows, Agents, Huntgroups - working our's, holidays etc. This will also respects OCS presence state and offers you a lot of basic and advanced ACD options. In your described case, user can redirect the second call in a "please wait a moment - I am in a call workflow". Configuration of this workflows can be very granular, but you will easily understand what is configured. You described so OCS R2 Attendant Console might be the better client for this user, instead of Office Communicator. Please give both option's a try Best regards, Jan
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Hi DWAyotte, please carefully compare your setup with the OCS pbxnsip wiki page: http://wiki.pbxnsip.com/index.php/Office_C...ications_Server ! Trunk call: Could not identify user typically means you missed this part of the article: I am looking forward, to your feedback! Good Luck! Best regards, Jan
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Inbound calls through PBXnSIP to OCS
Jan Boguslawski replied to Louis Yssel's topic in Microsoft OCS
Hello Louis, are you still facing this problem? Or did you found a solution. I am not sure, but I guess I have seen something similiar. If possible, please update us with your status! Best regards, Jan -
Hi Andrew, what you like to do is more like a "Direct SIP" or better "Greenfield" scenario. The pbxnsip integration path is a "DUAL-Forking" scenario. I am sure the software logic in pbxnsip would be able to handle what you like to achieve. But a full fledged voip-pbx like pbxnsip is a bit oversized for this. A voip gateway might be enough (AudioCodes, Dialogic, Ferrari-Electronic, etc. Or if you prefer software only, try asterisk with openser or sipX. You will find a lot of online info in google about it. At least why dont try a native OCS R2 SIP trunking? Supported providers are Sprint and Global Crossing http://technet.microsoft.com/en-us/office/...8.aspx#trunking But you will also find solutions (not certified) for R2 and R1 SIP trunks, for example: SmartSIP installation at your Mediation Server for any VOIP-provider http://evangelyze.net/products.asp#SMARTSIP Best regards, Jan
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OCS Passing presense information with pbxnsip
Jan Boguslawski replied to gotvoip's topic in Microsoft OCS
Hi OCS R1 and R2 offer two options for pbxnsip. 1. classic RCC approach: pbxnsip need a OCS compatible presense server built-in, like an ESTOS CallControlGateway using ECMA TR78 uaCSTA (which offers in a second development step to become also a Remote Call Control Gateway). At least you will not only pass communicator / phone presence to both systems. Additionally you would be able to remote control (CTI) your pbxnsip registered phones, via Communicator. (enable's Enterprise Voice, [the tel: URI] + PBX integration [the RCC server URI] settings in AD User Account's) a very good explanation of OCS RCC can be found here in an microsoft technet magazine article from march 09: http://technet.microsoft.com/en-us/magazin...cc.aspx?pr=blog I am not a developer, but I guess this is not impossible to handle. ECMA TR78 is well documented, btw. 2. alternative Snom phone approach: learn how Snom's OCS edition phone's registers natively at OCS R1 and R2 and how they pass and receive the presence status. Put this capability in pbxnsip and register every pbxnsip account additionally at OCS. Personally I would prefer option 1, cause it offers the option for a DUAL-FORKING & RCC scenario with pbxnsip, which is the high class level of OCS - pbx integration. Beside Nortels CS1000 / OCS integration (which is a result of Microsofts & Nortels Innovative Communications Alliance) pbxnsip would be the first thirdparty pbx supporting this. Just for clarification: in an DUAL-FORKING & RCC scenario you still can use Communicator natively to make and receive calls, do conference etc.. RCC adds the option to remotely control your phone, exchange presence informations between the two systems and lets you define your preffered "device". So user's can decide what's the best endpoint at the moment, depending on the situation (office, home, hotel, etc.) btw.: in option 2 you need to know the User accounts and passwords from your OCS / Active directory users. In option 1 this is not necessary. Best regards, Jan -
Hi - Sometime\'s pictures say more than words! Wow! I am bowled over. You too ? Please poll! It will also be available with for 2007 R1 and R2 version of Best regards, Jan
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Update for you on Snom Conference Solution for OCS R1 / R2 new snom MeetingPoint at CeBIT (03.03. - 03.08. 2009) – with At CeBIT 2009 snom will present its conferencing solution snom MeetingPoint. The snom MeetingPoint features: Available with "snom OCS edition" to connect to Microsoft Office Communication Server 2007 (R2) More details to be announced on CeBIT Under the beautyfull designer dress and great soundsystem by Konftel a Snom (not a joke) is hidden. Like for the 3xx and 820, an OCS Edition will bei available. If you like to see more Detail regarding the default cool features and accessories by Konftel please take a look NOW Imagine this connected to your OCS enviroment Great isn't it? It was possible cause of the new partnership Please continue reading more about it here. Best regards Jan
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Hi nexSIP, the MS Supportability Guide states: PKI Support A PKI (public key infrastructure) is required. Certificates that are issued from the following types of CAs (certification authorities) are supported, depending upon the requirement: • Certificates issued from an internal CA • Windows Server 2003 SP1 Enterprise CA (recommended) • Windows Server 2003 SP1 Standalone CA (supported, but not recommended) • Certificates issued from a public CA Communicator Web Access MTLS certificates must be issued by the same CA that issued the MTLS certificates for the Office Communications Server 2007 server or servers. Note Office Communications Server 2007 will support certificates with a length of up to 1024 bits. Office Communications Server 2007 server certificates must be configured with an enhanced key usage (EKU) extension for server authentication. You can download it here Support Guide This is complety contrast to the statement in the Edge Server Deployment Guide: KeyLength = 1024 Must be a power of 2 between 1024 and 4096, inclusive But the Supportability Guide was published later and includes life experience from e.g. Microsoft Field Engineers and Consultants. I guess this document is more up to date and their were issues with more than 1024 bit certificates. I dont know what kind of problems that might be. If you follow the Support Guide very strict, your expensive 2048 bit UCC certificate is at minimum not supported, at maximum not (fully) compatible with OCS. Just test all official clients (OC, AC, Outlook (also Play On Phone), Communicator Mobile, Communicator Web Access, Tanjay (!) and check if all work together with your 2048 certificate. Maybe you can renew the certificate during time of contract, like offered by digicert. Best regards Jan
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Speech Server Not Hanging Up Calls
Jan Boguslawski replied to AndrewW's topic in Microsoft Speech Server
Hi Andrew I am not an Speech Server expert, but you can run the SNOOPER Tool from the ressource kit. Make a trace with it at Speech Server. One with pbxnsip and one with Communicator. At least let us compare the SNOOPER Trace here. Hope you are not too disapointed for getting such an extremly late response. Regards, Jan -
Hi Enercia, one solution might be the Exchange UM autoattentend! As you can see in the picture, there is an option called "Allow callers to send voicemessages", which is off by default ExUM AA setup: You can find more detailed info @ managing-unified-messaging-auto-attendant part 1- 4 !!! Maybe you will be suprised about all options the ExUM AA can offer by default and how you can extend it! regards, Jan
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Hi IPNOX, please try to troubleshoot this at ExchangeUM first, if this is reproduce-able behavior. Please activate UM Diaglostic Logging at Expert Level: http://technet.microsoft.com/en-us/library/bb430783.aspx I prefer the PowerShell steps. Then wait until the problem comes back. In the event-viewer you might be able to see the cause for the issue in nearly clear text. Please report this errors or warning to me. btw: Please tell me more about you deployment! You should not e.g. run pbxnsip and ExUM on the same machine, or combined with any other SIP-Solutions. sorry for this delayed response, but I can only check the forum every once in a while. regards, Jan
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Communicator external access & outlook integration
Jan Boguslawski replied to Coen's topic in Microsoft OCS
Congratulations! I guess the cause and the solution are not R2 or OCS schema related. I remember that one of the latest Exchange Rollups for SP1 can cause issues like that, and one of the newer once fixed it. Especially if your patching is automated or managed by another colleague you will not know about these changes. An OCS, R2 exactly OC client related reason might be, that in R1 your external certificate for OCS Edge Server and Exchange OWA + Webservices ssl publishing certificate need to be from the same commercial root CA, or you use one / the same SAN certficate on Edge and Exchange. Not the case with OC R2 anymore, as far as I know. Please try checking Exchange Autodiscover Test Results from the internet by hold down STRG/ CTRL & leftclicking Outlook-Icon in the TASKBAR. See if all of you Ex-services URL's are correct! If you like, post it here. Second: test all of the service by Outlook on the Internet. Perform an 1) Offline Adressbook Download, 2) Schedule a Meeting and see if you can see the Outlook FREE BUSY Status of the Collegues you added to the meeting 3) Do a PLAY on Phone in Outlook to see if the UM-webservices are working also from Internet / Anywhere Acces (RPCoverHTTP). Please report any issues here. Thanks. btw: I checked your blog. Maybe this might be interesting for you, regarding sharepoint: ITaCS SharePoint Post Installation Tool The ITaCS Change Passwort web part 2.0 beta Regards, Jan