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Everything posted by rudik

  1. I am using v5.1.2 and have problems changing a service flag that is secured by pin code. If a 4-digit pincode is set the sevice flag asks for the pin code. While entering the pin code, after two digits the service flag always tells me that the pin code is wrong, after second try it always works? Tried pin code of different length or different pin code. Is this a bug ? Thanks for any clearification... Rudi
  2. Can you activate the SF manually: Yes ex dial the SF: Yes and are you able to map the SF account to a button on the snom phone: Yes But this does not work is the Service Flag is set Day/Night. Thanks for any clearification on this...
  3. I Upgraded to Zeta Perseids (Win64) but still not able to change the status of a day/night service flag manually. What can go wrong here ?? Thanks, Rudi
  4. Thanks for the answer, but i cannot get it to work. Maybe it is a new functonality wich is not in my version ? I am using
  5. Hi Guys, I received this request many times from my clients but still i cannot give them a clear answer why it is not possible to manually change(override) a automatic service flag . Can anybody give me a clear explanation or maybe a workaround ? Thanks, Rudi
  6. Also funny to see that the 'increase' voters are members of Snom Admins/supportgroups. The only other voter (snomonepbx) joined one week after starting this poll and is very knowledgable about the product Just my opinion...... RudiK
  7. Seeing the poll results makes me think Snom knows more about the market than the resellers/clients/users Or maybe they are correct that the new system will bring less sales but more revenues (for Vodia) because of contract. :blink: @snom: And this is what the market at this moment demands for. The current economic situation makes that no company would like to invest in long term financial commitments like some kind of contract. That is the reason why selling (talking as a reseller now) the old system (PBXNSIP) was not easy. Companies did not want to pay the yearly fee. Since the release of SnomBlue ("One System All Features No Limits") my snom sales doubled because it was easier to explain this one-time investment. Wether the company would grow or shrink, it did not cause extra costs. That was a strong argument; one time, long time investment. Going back to old system will, as expected, bring my sales back with 50% or, seeing the current economic situation even worse. Rudik
  8. I was a shock to me to find out that Voida goes back to limited functionality at higher prices. http://www.snomone.com/versions The 'old' SnomOne Functionality and Pricing gave enough financial flexibility to convince several Small Business clients. What to do with clients that already received a quotation from me ? Seems that "One System All Features No Limits" goes out the door :( Request from a reseller (me): Please review options and prices..... Thanks...
  9. Is it possible that this problem (bug) is also present in version 2011- (Win64). For many internal calls i have two records in my CDR and also two recordings. My settings are: Recording default for this domain: Record incoming calls from hunt group: Yes Record incoming calls from agent group: Yes Record incoming calls from extension: Yes Record outgoing calls to internal numbers: Yes Record outgoing calls to external numbers: Yes Record outgoing calls to emergency numbers: Yes Or is this "as designed" ? What would be the right setting for recording everything, but only once ? And also logging once in CDR ? Thanks, Rudi
  10. On this forum the link 'snom One' in the menu bar, just below the logo, opens the website 'www.snomone.com'. This site does not seem to exist ? I suggest the link should be changed. Regards, Rudi
  11. Exactly that is also my problem !! Extension A calls extension B, after a few rings the call is forwarded to extension C. But extension B did not want to talk to extension C !! Two people have a un-needed interrupted in their daily business.... :angry: I assume the suggested solution of jano7878 would be the solution for that. I think this would be quite easy to implement. Maybe even implement an option to specify from witch numbers the forwarding rules would apply. If the caller is not in this list, the 'normal' forwarding rules would apply. Please implement..... Thanks, Rudi
  12. Hi all, I am currently implementing a new Snom One Blue PBX (2011- Win64)and have a strange (out of the box ?) CDR logging format. Instead of starting the file with <?xml version="1.0" encoding="utf-8"?> according to XML standard the example file called "93.xml" in the cdre-folder contains the following data ?? : === Start of file === TLVB c 1349258449.548 cid ,YjdlMWEzNzI4ZDVhMzhmNDhkYTkyM2NjMmIxNTM5YWQ. ct d d 1 e 1349258454.688 f +"Rudi Test" <sip:800@vijf> i ,YjdlMWEzNzI4ZDVhMzhmNDhkYTkyM2NjMmIxNTM5YWQ. o I p udp: r "Rudi" <sip:061111111@vijf> rec /recordings/20121003/120049-o-061111111-800.wav s 1349258441.81 t "Rudi" <sip:061111111@vijf> u 44 vq VQSessionReport: CallTerm LocalMetrics: Timestamps:START=2012-10-03T10:00:49Z STOP=2012-10-03T10:00:54Z CallID:YjdlMWEzNzI4ZDVhMzhmNDhkYTkyM2NjMmIxNTM5YWQ. FromID:"Test 5 New"<sip:800@>;tag=a817667b ToID:"061111111" <sip:01111111@>;tag=eb71e45556 SessionDesc:PT=8 PD=pcma SR=8000 FD=20 FO=160 FPP=1 PPS=50 PLC=3 LocalAddr:IP= PORT=52162 SSRC=0x48763f50 RemoteAddr:IP= PORT=37286 SSRC=0xe6cb2e02 x-UserAgent:snom-PBX/2011- x-SIPterm:SDC=OK SDR=AN PacketLoss:NLR=0.0 JDR=0.0 BurstGapLoss:BLD=0.0 BD=0 GLD=0.0 GD=0 GMIN=16 Delay:RTD=0 ESD=0 IAJ=0 QualityEst:MOSLQ=4.1 MOSCQ=4.1 RemoteMetrics: JitterBuffer:JBA=3 JBR=0 JBN=40 JBM=40 JBX=80 PacketLoss:NLR=0.0 JDR=0.0 BurstGapLoss:BLD=100.0 BD=40 GLD=0.0 GD=5920 GMIN=16 Delay:RTD=0 ESD=0 IAJ=15 QualityEst:MOSLQ=4.1 MOSCQ=4.1 x-UserAgent:eyeBeam release 1009l stamp 37965 y extcall === End of file === Clearly this does not have a XML header but some other (unknown to me ) header. As currently the CDR XML files should be analyzed by a seperate control-system these files are useless because the control system expects XML format. I have been reading the documentation and 'normal' XML should be default for Snom One? Unfortunately in this case it is clearly not default ? What settings should be changed to 'reset' the logging to the 'good old' XML ? The control-system runs great on the old version ( Win32), if possible the same format would be ideal. Thanks, Rudi
  13. It seems that the forwarded call still has the original Caller-Number. There is no 'ReWrite' which should replace the From field with de Trunk-ID. It might be possible there is nu A-number being send ? This is really geting urgent. Please help.... Thanks for any help....
  14. After analyzing the SIP trace i see (Packet example): INVITE sip:01234567@sip.esprittele.com;user=phone SIP/2.0 Via: SIP/2.0/UDP;branch=z9hG4bK-e065b4f437e49fcc77f4548dee2a3eb0;rport From: <sip:anonymous@;user=phone>;tag=39066 To: <sip:01234567@sip.esprittele.com;user=phone> Call-ID: d00454e7@pbx CSeq: 24852 INVITE Contact: <sip:07654321@;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE User-Agent: pbxnsip-PBX/ Related-Call-ID: 78f6ccb0bfe46b09@ P-Preferred-Identity: <sip:07654321@sip.esprittele.com> Replaced info: anonymous = Calling External Number (hidden in this case) 01234567 = External number to be called (forwarded) 07654321 = Trunk Phone number (ANI) = SIPTrunk Provider IP = IP PBX Call (Invite) is made with correct P-Preferred-Identity (07654321) but provider does not seem to use this. - 100 Trying - 407 Proxy Authentication Required - ACK sip:01234567@sip.esprittele.com;user=phone SIP/2.0 - INVITE sip:01234567@sip.esprittele.com;user=phone SIP/2.0 - 100 Trying - 603 Decline During this communication everywhere i see: "From: <sip:anonymous@;user=phone>" Provider finds a call 'From' number wich is not allowed because of billing purpose (in this case even anonymous). So call is ended (603 Decline) Is it possible to replace the 'From:' field with the Trunk ANI and fix my problem ? Thanks for any help....
  15. Since a harddisk crash from a older PBXNSIP installation, i re-installed on a new system and am trying to restore the old configuration. I have problems implementing the following: Incoming external call has to be forwarded to external number (outgoing). Tried everything but call is not being forwarded. Testing from an internal number (extension), the forwarding to the external number does work. I remember it was some kind of security in PBXNSIP but i can't remember how to enable this.. Relying on your memory..... Thanks, RudiK
  16. Searching through the forum i found that the 'old' pbxnsip supports use of only one core from a multi-core processor. Has this changed in Snom-One PBX software (32 of 64 bit) or still only uses one core ? We are searching for new hardware for our new PBX. Is it true that a dual core processor might perform better than a quad core processor as long as the internal clock speed of the dual core is higher than the clock speed of the quad core ? Lets start a discussion about this... RudiK
  17. At this moment it is difficult to get a complete log, i will get it soon. I did not assume the problem could be related to the incoming invite because the Hunt Group setting "Show Calling Party Name" does actually work. The Hunt Group member shows the resolved name of the caller. What could be the relation between incoming Invite and the option "Show Group Name + Calling Party Name" ? Thanks....
  18. This is unfortunately not the case Please have a look at the Invite and let me know what is missing. For privacy reasons following information is replaced: %PBXIP% = Internal IP from SnomOne PBX %MOBNUM% = My Mobile phonenumber for test (number is present in PBX Adress Book in multiple formats) %PBXNUM% = Phone number of PBX %PROVSIPSRV% = Server Adress of SIP Provider (from incoming Trunk) INVITE sip:600@%PBXIP%:1035;transport=tls;line=sqmkp8ek SIP/2.0 v: SIP/2.0/TLS %PBXIP%:5061;branch=z9hG4bK-e0a8212dd259f76a1ac8a405e60e40da;rport f: "HuntGroup1" <sip:%MOBNUM%@%PROVSIPSRV%;user=phone>;tag=11998 t: <sip:%PBXNUM%@%PROVSIPSRV%;user=phone> i: 43430db0@pbx CSeq: 10737 INVITE Max-Forwards: 70 m: <sip:600@%PBXIP%:5061;transport=tls> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snom-PBX/2011- c: application/sdp l: 315 Thanks, Rudik
  19. We are using multiple incoming numbers witch each their own hunt group. There are a few central phones wich are placed in several hunt groups. The person picking up the phone should answer with a according greeting matching the incoming number. The current possibilities are: - Show Group Name (Works) - Show Calling Party Name (Works, from resolving by global address book) We would like to have a combination of both: - Show Group Name + Calling Party Name It this possible ? Thanks....
  20. I have the same problem, how can i make the internal address book resolve the incoming numbers to their names ? Is this possible ? Thanks...
  21. Hi there, Can anybody tell me where i can find the latest version of the PAC or tell me how to get into the WAC ? I need a littlebit of help here.... Thanks, Rudi
  22. Hi There, I hope this post will not be removed Does SNOM has something like a mobile phone client like SWYX does ? http://www.swyx.com/products/devices-and-softphones/mobile-integration.html Or does SNOM rely completely on soft-clients from other developers (for the mobile environment) ? Thanks for any answer, Rudi
  23. Hi everybody, I have a question about mobile users licensing. In a situation with SNOM blue: We have an account with a snom phone registered. As a second device a mobile phone with a SIP-Soft-Client is connected to the same account. Does the mobile phone count as one of the 40 non-snom devices that can be connected ? Or, because the account already has a snom registered, a second device does not count for licensing ? Or: can anybody tell me how SNOM PBX detects that there is a SNOM phone registered ? What is the logic behind the "40-non-snom-device" system ? Thanks, Rudi
  24. Hi Matt, I see you have some experience with the portech (MV-3XX) GSM Gateway. I need a GSM Gateway and i am currently diving into the possibilities of the device. Unfortunately i am quite new to the GSM Gateway equipment and i have difficulties finding the technical possiblities of the device. Maybe you/or_anyone can tell me if the following is possible: There is a SIP trunk to a 'normal' SIP provider. Coming through this trunk a call arrives from external phone number 'A'. This call is 'send' to a extension. The 'regular' phone is unavailable and the call is forwarded to a mobile number through the Portech GSM Gateway. I want to know if the mobile phone user can see that the call originates from phone number 'A'. I hope the example above is clear enough to help me get the answer to my question. Thanks in advance. With kind regards, RudiK
  25. Type: gateway => try making this a proxy Display: SPA-3102 RegAccount: 620 RegRegistrar: RegUser: 620 OutboundProxy: => always problems until i changed port number => PSTN Line on SPA3102: => I always used the external (WAN) for SIP connection (, not LAN!! Line enable: yes SIP port: 5060 => 5061 Proxy: (PBX's IP) Outbound proxy: => Keep empty Good Luck....
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