Endpoint Configurator

Discussion in 'General' started by Gahao, Nov 15, 2010.

  1. Gahao

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    Did anyone set this function yet?

    How to set this function on Elastix Server?

    Besides, could you tell me which vendor's IP phone could work well with this function?


    Thanks for all your kindly replies.
     
  2. Gahao

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    Has no one used this function?
     
  3. iwaserman

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  4. iwaserman

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    hi,

    1. follow the link I posted before.
    2. get the module it self from the "Files" page and save it on your desktop.
    3. log into FreePbx (you can simply click on the "Unembedded freePBX" link in Elastix.
    4. open the Module Admin and select the Upload module option.
    5. After installing the module go to this link http://projects.colsolgrp.net/documents/27

    support is for Aastra, Snom, Polycom, Linksys, Grandstream, Yealink & Dreamweave

    good luck
     
  5. Lee Sharp

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    I am using it out of the box on 2.0 and have mixed feelings. When it works, it is smooth, amazing, nice, fantastic, and so on... But... It does not always work, it is unpredictable, and some setting are hard or imposable to find. It reminds me of my ex... B)

    It works OK on Aastra with some kicking. It does not work at all on the Grandstream ATAs I have. That is all I have tried it on. I have provisioned about 45 phones with it.

    So what is your question?
     
  6. dicko

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    The Aastra XML provisioning not only works but value adds an awful lot, sorry about the grandstream stuff, (I really mean it, I'm sorry you got stuck with it ;) )

    dicko
     
  7. Lee Sharp

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    Just to give specifics (as long as Dicko is here) let me give you the issues I have.

    The Aastra phones will not pull config right off the bat. I have to manually set the "configuration server" parameter. I thought it was an option 66 error, but my computers try to pxe boot off my phone system just fine. :)

    The config is set by ip address, not domain name. I have yet to figure out where to change that. Every time I find a file I think will change that, Elastix overwrites the damn thing on the next phone I provision. :S I want to use domain name, so I can take the phone to a hotel, and it will work.

    As for the Grandstreams, at least I only have 2. Unfortunately, one is in the conference room with an analog polycom phone, and it keeps dropping calls at 17 and a half minutes! http://www.elastix.org/en/component/kun ... nutes.html Live and learn... No Grandstream. No OpenVox. Don't ask me how I know... http://www.youtube.com/watch?v=w4Ksl95mD_g
     
  8. dicko

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    If you have a PXE server running, yes it is option 66, if you are using the Elastix dhcp server
    it is setup in /etc/dhcpd.conf

    the line that says

    option tftp-server-name "tftp://10.65.65.100";

    should be changed to

    option tftp-server-name "10.65.65.100";

    if you have "old" Aastras (9133/480 era)

    The phones will first read

    aastra.cfg from /tftpboot/ then 0008NNNNN.cfg the mac specific,

    add a -vv to the server_args in /etc/xinetd.d/tftp , restart xinetd and tail -f /var/log/messages to see "the action"
     
  9. Lee Sharp

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    DHCP is on my gateway, and is needed that way for the rest of the network. I have a blank for "Next Server" and "Filename" and that is it. The phones are 6730i's.
     
  10. dicko

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    nextserver is not option 66, if your sDHCP server is not capable of offering that option, you will have to hand enter it on each handset.

    If the Elastix box is on the same network as "the rest of the network" consider using the one o n the Elastix box.
     
  11. franklin

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    dicko, I just put up a new box. It is sitting behind the router that had previously been supporting another box perfectly. I changed all my forwarding rules to reflect the new box's LAN IP. It is passing voice perfectly, but I cannot hit the tftp. On support with Cisco now doing a syslog debug. Do I have to do anything to start the tftp? phones hit the other tftp just fine. thanks.
     
  12. franklin

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    Aye aye. I had two UDP forwarding rules on my Router, one was a UDP all, and the other port 69. neither would save when I checked the box to enable because of the conflict.

    But I DO still need help! at var/log I run tail -f messages | grep -i tftp

    On this new box, when the phone hits, I see START tftp pid=xxxxxx from=<myip>

    But I don't get the output that shows which files are being requested, etc. Do I have to increase the verbosity on the output? If so, can someone tell me exactly how to do this?

    Thanks
     
  13. dicko

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    server_args = -s -c -vv /tftpboot

    in /etc/xinet.d/tftp (v is for verbose, c for create (to write logs etc. back tio the server, but be careful in who gets access to port 69) )
     

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