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Friedom-Tech

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Posts posted by Friedom-Tech

  1. Hi,

     

    Polycoms are using 3.1.3.0439 and pbxnsip is version 3.4.0.3201 the server is on a local lan with the phones with no firewall at all. but when people listen to messages or even delete them sometimes the light on the polycom continues to flash...

     

    is there anything i can do about this?

  2. IMHO one of the points of DND is that the caller does not want to tell the caller that he or she is right now not willing to take calls. The caller should be in that foggy cloud of being busy or just not logged in.

     

    (One of my biggest concerns about the whole presence hype is that people don't want to have other people spy on them what they are doing right now, called the "presence state".)

     

    Hi,

     

    on DND i happen to agree with you but when callers are on the line it is very difficult for INTERNAL users to know this.

     

    i know for example with old fashioned nortel systems when somone is on the phone a popup will show on the screen of the caller saying line busy or on phone but the phone will still ring the person they originally dialed but at least they have the option to hang up knowing that the person they are trying to call is on the phone!

  3. One small trick: If the account does not have a PIN, the PBX does not offer the DISA option.

     

    But it makes sense. Outbound calls might cost a lot of money to the company when employees have a girls friend in some remote location on this planet. We have to look into adding an option to disable that feature, maybe per domain, maybe also per extension.

     

    Thats a nice trick the only problem is that these are regular phone users so what do they do about PIN for voicemail?

  4. If you are using Windows you can use the microphone in connection and select it from the MoH menu.

     

    if i am using a windows system will it support sound coming from a CD that is playing on the server itself?

     

    meaning if i put a music CD into the CD/DVD drive of the same windows system that is running pbxnsip will that work?

  5. Hi;

     

    i have two totally seperate pbxnsip windows boxes running out of two totaly different locations running version .3202 which are experiencing the following issues:

     

    1) Dropping trunk registrations sometimes up to 100 times in a matter of five minutes.

    2) on outbound calls the caller hears blank space but the person he/she is calling hear their phone ring...pick it up but cannot hear anything and the person who placed the outbound call does not even hear ringback never mind the person picking up the phone...this issue is occurring with multiple carriers.

  6. Hi,

     

    i have a client who has purchased a "sold on hold" message provider to be plugged into

    the "pbx" and used instead of music when folks are on hold. It appears

    to use either an RCA plug or mini phone jack and either 600ohm or 8 ohm

    output. Does anyone have experience with these type of systems? Where does

    it plug in to the phone system?

     

    Any help would be most appreciated.

  7. Well that is only possible with duplicating accounts and using static registrations that point to the other PBX. We (pbxnsip) love this approach because we sell twice as much licenses. :D If you have below 10 extensions I would use the direct destinations on the AA for this.

     

    there is no other way to do this?

     

    from the customers perspective this makes no sense...he feels like what is the point of VOIP if he cannot accomplish this???

  8. This can only work if the destination system believes that the call comes from an extension. So what you can try:

     

    • Set up a new trunk on the source system which is outbound only and that uses the credentials of an extension on the destination system. Having the trunk outbound only makes sure we are not screwing anything up on inbound traffic on the PBX. Choose a reasonably long extension name like "2127654321" so that the chance of mixing that number up is pretty low.
    • Create that extension with the same password on the destination system.
    • Assign 8xxx as the pattern in the source system dial plan and use that new trunk.
    • Give it a shot!

     

    this worked!! thanks!

     

    the only thing we still need to figure out is a way to be able to teach the auto attendant to recognize extensions on another system.

     

    in the same scenario where a client has two trunked pbx's but would like callers to call one number and be able to enter an extension number regardless if the number is on the NY system or the NJ system!

     

    can this be done?

  9. It seems that sometimes the format has still some issues. I am using GoldWave and I was always able to play the file back, even if GoldWave also complanied about the file being buggy. I believe we did fix something with the padding; but it seems that there is still something that causes problems.

     

    The other thing that could be a problem is that the email program translated LF into CRLF. You can check if this is the case by taking the WAV from the file system and play it back.

     

    Hi;

     

    i am reading the file directly from the file system...it is not getting emailed anywhere!

     

    can i tell my client this will not happen again? or do i need to be wary of this?

  10. Hi, i have call recording enabled for all calls in the system but tonight i have two files which don't want to open they are important recordings but i am getting the following message when trying to open them.

     

    "Windows Media Player cannot play this file. the player might not support the player type or might not support the codec.

     

    i tried opening it with itunes too and it wont open.

     

    please advise asap.

  11. Hi;

     

    i have a setup as follows:

     

    two pbx's trunked together.

     

    the current dial plan states that any calls to 3xx should go through to the NJ system (over the trunk between the systems). however if they want to transfer a call to voicemail of someone on the NJ system they cannot do that...i have tried putting in the DP that any calls to 83xx should go to NJ system but that doesn't work either. it gives a fast busy signal and logfile shows 404 Not found.

     

    any suggestions would be most appreciated.

  12. If you are doing plug and play the PBX will already set it up this way. If you manually configure the phone, probably you have to go to the web interface and change the outbound proxy there (depends on the phone type).

     

    what do you mean? i have the system setup using PNP!

     

    do i have to change setting on pbx...please advise where.

  13. Well, at least this is a sign that the phone is in trouble. Probably it tries the various DNS SRV entries that are available for the server when it fails from one transport layer to the other. What you can do is advertize only one transport layer or force the extension to use a specific transport layer (and thus rule out DNS SRV) with a outbound proxy like "sip:dns.name.com;transport=tls".

     

    how do i pass this config through to the phone?

  14. The Setup is Pbxnsip latest version and Polycom phones using the latest firmware.

     

    1) TRANSFERRING CALLS INTERNALLY

    When a user wants to transfer blind to his boss he can only do so by hitting transfer and then entering extension and hitting blind, in this case the boss will see the actual number of the person calling but the employee will not be able to tell him who is on the phone.

    I have feedback from a number of clients that find this frustrating because although they want their secretary to screen the calls, if the emplyee does NOT hit blind (because she wants to check with boss first if he wants to talk to the caller) then once she transfers all he sees are a bunch of calls from her so when he is handling multiple calls he has no idea which is which, this may be a polycom limitation but any way to get around this now or maybe in 4.0?

     

    2) CALL BACK WHEN USER IS ON PHONE

    Right now when a caller makes and internal call he/she have no way of knowing if the person is on a call/away from their desk, even if you offer camp on then it will still ring until the camp on message picks up. MOST of my clients have requested the option to have a message popup on their screen or for the camp on message to be immediately relayed to them if another user is on a call and be given the option to continue calling the person or to wait for call back/leave a message.

     

    3) USING THE CALL BACK FEATURE

    I would also like to clarify, how the call back feature works...if the user is away from their desk and the system offers the call back option...when exactly will it call the caller back?

     

    please advise on these issues asap.

  15. Source address for "Tom Smith"

    <sip:605@66.66.66.66>;tag=A00BCAF2-E2F3197 has changed from

    udp:55.55.55.55:1204 to tcp:55.55.55.55:55920

     

    the source address of the phones keep changing from UDP to TCP and vice versa.

     

    is there any reason for this? the client is also complaining that he gets URL call disabled on his polycom at least once a day?

     

    this is a hosted client and the hosted server is using latest version of pbxnsip, phones are using latest firmware.

  16. Did you try the "In call Record" star codes? this may have to be executed by the moderator during the conference.

    No...the domain admin needs ALL calls recorded as he monitors calls etc...he does not want to schedule every call but needs FULL call recording without his employees knowing he is recording!

     

    Pls advise asap

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