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shopcomputer

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  3. auto provisioning in 3.3 from outside lan

    in snom Phones

    Posted

    In 3.3, there is a change in the way you need to provision snom phones. The new link must be in the form http://ip:port/provisioning/snom300.htm (where the snom300 needs to be replaced with the actual phone model). We'll update the Wiki as soon as 3.3 is publically available.

    And on the phone I had to set the http client setting with the user name and password of the extension, then it worked.

     

    On the server generated folder, it created a folder named localhost, iinstead of the MAC_ADDRESS.

  4. auto provisioning in 3.3 from outside lan

    in snom Phones

    Posted

    Has this provisioning method been completely disabled from outside the lan?

    http://pbx.pbxnsip.com:9012/provisioning/s...0041324006D.htm

     

    What is the correct method to provision from outside the LAN, and obtain all the phone settings.

     

    I tried putting the MAC in the extension, and using http://pbx.pbxnsip.com:9012, the phone logs showed some redirects, time settings being applied however thats about it, it did not register.

     

    3.3.0.3165 (Win32)

  7. DTMF Problem Snom M3

    in snom Phones

    Posted

    This is starting to get a lot of folks very angry. It is tough to troubleshoot because it seems to be random but once it starts on a base the only thing that will fix it is a restart of the base.

     

    Tom

    Did you try the Snom.com forum?

  9. Trunk setup for correct ANI

    in General Setup

    Posted

    Oh you mean for the DISA feature when calling from the call phone?

    I think he is talking about, http://wiki.pbxnsip.com/index.php/Release_..._3.1#Cell_Phone

     

    Also, when the cell phone was used to place an outbound call, the PBX sometimes presented the Caller-ID of the cell phone, not the extension. The purpose was to hide it. That was fixed, and now users can use their cell phones to place outbound calls and present the caller-ID of the PBX.

     

    For most cases this problem still exists, I did manage to get it working with 1 ITSP, creating an inbound trunk and an outbound trunk, and some how forcing it to display the trunk ANI.

  10. Multicore technology utilization - when will be available?

    in PBX Lounge

    Posted

    That's a lot of work to accomplish this as you would need to separate everything including having separate web portals for each. This makes it increasingly difficult to manage. Do you have an estimate time line when this will automatically work making 1 login and binding all to 1 IP. This sounds like a duck tape job for now at least.

    Would be nice to be able to choose in admin how many processes to run, and maybe be able to map domains or functions to a process.

  11. Trunk setup for correct ANI

    in General Setup

    Posted

    Sorry I am not big with forums or posting comments via text as it is more complicated to interpret. Is there any phone support even if it is paid so I can convey my message.

     

    My scenario is dialing out to the PSTN where in some cases we need the calling party ANI to go through pbxnsip al the way to the destination #. This happenes when a PSTN (489-1500) user dials our DID 790-1509 which on the pbx is setup as extension 1509 with immediate call forward to 7631599. We want the original calling party (489-1500) ANI showed up on mobile 7631599 and this does happen with P-Preffered Identity setup on our trunk.

     

    everything has been working great.

     

    know we have a new scenario where we have 763-1599 calling the auto attendant and being authenticated by caller id and pressing 1 to place out bound calls to PSTN. we want to hide the 763-1599 ANI from end user receiving call on PSTN. but for that to work we have to setup trunk to P-Asserted Identity so it respects the ANI of the office extension (956-790-1509). so when we do this change we loos the previous working scenario.

     

    so that is where my question comes into play to have both scenarios working what setup can we have? could you make the option on the extension setup to have P-Asserted Identity and override the trunk setup when you see this option.

     

    If I understand this, he is trying to block the cell phones number from displaying as the outbound caller ID, this was supposedly fixed in version 3.2, thus far most ITSP's I tried still display the cell phone.

  13. Call-through from Cellphone (DISA) shows Cellphone Caller-ID rather than PBX Caller-ID

    in Extension Setup

    Posted

    Dear Members,

     

    we have the problem exactly described in the Release Notes 3.1 to be fixed:

     

    "Also, when the cell phone was used to place an outbound call, the PBX sometimes presented the Caller-ID of the cell phone, not the extension. The purpose was to hide it. That was fixed, and now users can use their cell phones to place outbound calls and present the caller-ID of the PBX."

     

    If we use the DISA-Feature (Cellphone calls PBX and make external call through the PBX) the Caller-ID of the Cellphone is sent. We tried to set the Extensions-ANI with the PBX's Caller-ID, but this do not prevent to still send the very secret Cellphone-Caller-ID.

     

    Any idea?

    not fixed for me either.

  14. Snom Phone behind ISA connecting to public PBXnSIP server

    in Firewalls and NAT

    Posted

    Did anyone ever get a phone behind a Microsoft ISA server connect to a PBXnSIP server on the public internet.

     

    I tried following http://stewedprunes.com/blogs/stewed_prune.../08/29/502.aspx written for Vonage and changing the RTP port range, and also forwarding ports 5060-5080 tcp and udp it did not work.

     

    If I set transport=tcp, then I get http://wiki.pbxnsip.com/index.php/One-way_Audio.

  16. Cell Phone Schedule

    in Service Flags

    Posted

    I would just use one auto attendant and 24 service flags. One falg for each hour. Then use the night mode feature to distribute the call to each cell phone:

     

    Service Flag Account: 7100 7101 7102 7103 ...

    Night Service Number: 9787462777 9787462778 9787462779 9787462780 ...

     

    You can even make exceptions for the weekend and for the holidays!

     

    I am not following, there is only 1 service flag field in the autoattendant, and one night service field.

     

    If you seperate by spaces, the first service flag corresponds to the first night service number, second to the 2nd, etc.?

  17. Cell Phone Schedule

    in Service Flags

    Posted

    I am quoting on an opportunity for a nonprofit volunteer organization, they basically have 1 person on duty taking calls on their cell phone 24/7. Every hour there is someone else on duty.

     

    I am thinking how to configure this for them, the first thought that came to mind was creating a hunt group and adding all the volunteers as members of the hunt group, and setting the call cell phone option, with an explicit schedule allowing calls only during his on duty hour.

     

    However then I reminded myself about our no cell phone from hunt groups issue. I am back at the drawing board.

     

    What can I accomplish with a service flag? should I create scheduled service flags for each user and have an IVR forward to the users with manual registration?

  18. reformatting of calls

    in General Setup

    Posted

    Never seen this behavior in any CS410/425 old or new version deployed during the last 1+ yr. Our Default Dial plan is "*" "null" replacement.

     

    We set Domain PNP Plan to user must press enter...(People do this every day on their cell phones)

    On the trunk settings, there are options for changing its behavior.

  19. One way audio

    in General Setup

    Posted

    This morning we've experienced one way audio inward from callers. The callers can hear us but we cannot hear them. We've tested domain to domain, within the same domain and all has the same issue. We can call in to check voicemail fine. Call forwarding on extensions works also, which doesn't make sense. Calling in from cell to place call from pbxnsip also works fine.

     

    Please HELP!!!

     

    Do you have a wireshark? Can you check your firewall log?

  21. Routing incoming calls using Caller ID

    in Trunk Setup

    Posted

    Hi:

     

    Thanks for the information, "Send call to extension" sends all calls to the extension.

     

    Aliases works sendong the call by checking the called number.

     

    What I need is to route the call depending on the CALLER ID.

     

    Thanks

    I am affraid at the moment you would need to use an IVR node in order to route based on the caller ID.

  22. what is most recommended pstn gateway?

    in Trunk Setup

    Posted

    Hi,

     

    Thanks for sending that mp114 ini file.

    Unfortunately it didn't work.

    My ini has quite a bit of info after the "[ \TargetOfChannel ]". there's no chance the forum cut off the bottom of the file? If so your welcome to email me directly.

     

    matt

    I have nothing after that, if you post your audiocodes and PBXnSIP logs, I am sure we can figure out your problem.

  23. what is most recommended pstn gateway?

    in Trunk Setup

    Posted

    Hello,

     

    Thanks for that post...that is what I was looking for...

    I'm not at the computer to test it but had some questions:

     

    #1- CNONCE = '0a123bcf' and PASSWORD = '787899'. Is this a password setup in pbxnsip?

    #2- is the ip address of the pbxnsip server in this senario 192.168.0.10?

     

    My config is here:

    http://forum.pbxnsip.com/index.php?showtopic=2003

     

    if you want to critique it.

     

    Once again, thanks.

    matt

    The IP of the pbxnsip is the 192.168.0.10, I don't know what that password is, it certainly is is not to the PBXnsip, as it is a gateway trunk not registered.

    According to http://wiki.pbxnsip.com/index.php/Audiocodes those 2 values should be left at their default values.

  24. what is most recommended pstn gateway?

    in Trunk Setup

    Posted

    Here is a sample config from a mp114

    ;**************

    ;** Ini File **

    ;**************

     

    ;Board: MP-114 FXS_FXO

    ;Serial Number: 768949

    ;Slot Number: 1

    ;Software Version: 5.20A.027.004

    ;DSP Software Version: 204IM => 520.13

    ;Board IP Address: 192.168.0.6

    ;Board Subnet Mask: 255.255.255.0

    ;Board Default Gateway: 192.168.0.1

    ;Ram size: 32M Flash size: 8M

    ;Num DSPs: 1 Num DSP channels: 4

    ;Profile: NONE

    ;-----------------------------------------

     

     

    [sYSTEM Params]

     

    SyslogServerIP = 10.1.1.89

     

    [bSP Params]

     

    PCMLawSelect = 3

    LocalOAMIPAddress = 192.168.0.6

    RoutingTableHopsCountColumn = 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0

     

    [ATM Params]

     

     

    [Analog Params]

     

    MinFlashHookTime = 100

    FXSLoopCharacteristicsFilename = 'MP11x-02-1-FXS_16KHZ.dat'

     

    [ControlProtocols Params]

     

     

    [MGCP Params]

     

     

    [MEGACO Params]

     

    EP_Num_0 = 0

    EP_Num_1 = 1

    EP_Num_2 = 0

    EP_Num_3 = 0

    EP_Num_4 = 0

     

    [PSTN Params]

     

     

    [sS7 Params]

     

     

    [Voice Engine Params]

     

    VoiceVolume = 1

    RFC2833PayloadType = 101

    CallProgressTonesFilename = 'usa_tones_12.dat'

     

    [WEB Params]

     

    LogoWidth = '339'

     

    [sIP Params]

     

    ENABLECALLERID = 1

    MAXDIGITS = 11

    REGISTRATIONTIME = 3600

    ISTWOSTAGEDIAL = 0

    GWDEBUGLEVEL = 5

    ENABLEEARLYMEDIA = 1

    ISUSERPHONE = 0

    CNONCE = '0a123bcf'

    PASSWORD = '787899'

    ENABLEVOICEDETECTION = 1

    ALTROUTINGTEL2IPMODE = 0

    ISFAXUSED = 1

     

    [VXML Params]

     

     

    [iPsec Params]

     

     

    [Audio Staging Params]

     

     

    ;

    ; *** TABLE DspTemplates ***

    ; This table contains hidden elements and will not be exposed.

    ; This table exists on board and will be saved during restarts

    ;

     

    ;

    ; *** TABLE PREFIX ***

    ;

    ;

     

    [ PREFIX ]

    FORMAT PREFIX_Index = PREFIX_DestinationPrefix, PREFIX_DestAddress, PREFIX_SourcePrefix, PREFIX_ProfileId, PREFIX_MeteringCode;

    PREFIX 0 = *, 192.168.0.10, *, 0, 255;

     

    [ \PREFIX ]

     

    ;

    ; *** TABLE CoderName ***

    ;

    ;

     

    [ CoderName ]

    FORMAT CoderName_Index = CoderName_Type, CoderName_PacketInterval, CoderName_rate, CoderName_PayloadType, CoderName_Sce;

    CoderName 0 = g711Ulaw64k, 20, 0, 255, 0;

     

    [ \CoderName ]

     

    ;

    ; *** TABLE TrunkGroup ***

    ;

    ;

     

    [ TrunkGroup ]

    FORMAT TrunkGroup_Index = TrunkGroup_TrunkGroupNum, TrunkGroup_FirstTrunkId, TrunkGroup_FirstBChannel, TrunkGroup_LastBChannel, TrunkGroup_FirstPhoneNumber, TrunkGroup_ProfileId, TrunkGroup_LastTrunkId, TrunkGroup_Module;

    TrunkGroup 0 = 1, 255, 1, 1, 8, 0, 255, 0;

    TrunkGroup 1 = 1, 255, 2, 2, 9, 0, 255, 0;

    TrunkGroup 2 = 2, 255, 3, 3, , 0, 255, 0;

    TrunkGroup 3 = 1, 255, 4, 4, , 0, 255, 0;

     

    [ \TrunkGroup ]

     

    ;

    ; *** TABLE PstnPrefix ***

    ;

    ;

     

    [ PstnPrefix ]

    FORMAT PstnPrefix_Index = PstnPrefix_DestPrefix, PstnPrefix_TrunkGroupId, PstnPrefix_SourcePrefix, PstnPrefix_SourceAddress, PstnPrefix_ProfileId;

    PstnPrefix 0 = *, 2, *, *, 0;

     

    [ \PstnPrefix ]

     

    ;

    ; *** TABLE TxDtmfOption ***

    ;

    ;

     

    [ TxDtmfOption ]

    FORMAT TxDtmfOption_Index = TxDtmfOption_Type;

    TxDtmfOption 0 = 4;

     

    [ \TxDtmfOption ]

     

    ;

    ; *** TABLE TrunkGroupSettings ***

    ;

    ;

     

    [ TrunkGroupSettings ]

    FORMAT TrunkGroupSettings_Index = TrunkGroupSettings_TrunkGroupId, TrunkGroupSettings_ChannelSelectMode, TrunkGroupSettings_RegistrationMode, TrunkGroupSettings_GatewayName;

    TrunkGroupSettings 0 = 1, 3, 255, ;

    TrunkGroupSettings 1 = 2, 3, 255, ;

     

    [ \TrunkGroupSettings ]

     

    ;

    ; *** TABLE EnableCallerId ***

    ;

    ;

     

    [ EnableCallerId ]

    FORMAT EnableCallerId_Index = EnableCallerId_IsEnabled;

    EnableCallerId 0 = 1;

    EnableCallerId 1 = 1;

    EnableCallerId 2 = 1;

    EnableCallerId 3 = 1;

     

    [ \EnableCallerId ]

     

    ;

    ; *** TABLE TargetOfChannel ***

    ;

    ;

     

    [ TargetOfChannel ]

    FORMAT TargetOfChannel_Index = TargetOfChannel_Destination, TargetOfChannel_Type;

    TargetOfChannel 2 = 53, 1;

    TargetOfChannel 3 = 58, 1;

     

    [ \TargetOfChannel ]

  25. what is most recommended pstn gateway?

    in Trunk Setup

    Posted

    hi,

     

    I have an audiocodes mp-114 fxo and i can't seem to get it working with the wiki config.

     

    (I had it working on another system)

     

    i've posted the .ini if you want to see if there are obvious mistakes on it.

     

    tx

    matt

    I do not see your ini, please attach it again. Are you using the MP114 FXO, FXS, or 2FXO/2FXS. If you can post your PBxnsip and MP114 logs, that will be helpful too.

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