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  • Aykut, Wed 01 of Aug, 2007 [07:53 UTC]: Hi all, does anybody know about Thomson ST2030 SIP phone. I have upgraded it to latest version (1.56) but "Hold" and "Conf" features are not working after the upgrade ?? Do you know any solution or do you have Ver. 1.52 ?? Where can I find it?
  • Edward J Brown, Tue 31 of Jul, 2007 [23:33 UTC]: Has anybody experienced Choppy voice quality when using a Linksys SPA942 in an Asterisk Conference bridge? It works fine with my polycom and Cisco, but sucks with my Linksys.
  • www.astawerks.com, Fri 27 of Jul, 2007 [18:00 UTC]: does anyone use asterisk on top of clark connect? does it work good?
  • simon, Fri 27 of Jul, 2007 [14:16 UTC]: Hi All, Has anyone here managed to get the Cisco79x1 to successfully fail over to the backup proxy. I have 2 asterisk servers , handsets all register and function, except that backup proxy function doesn't work. Any working example would be very apprecia
  • Matthew Richmond, Thu 26 of Jul, 2007 [03:40 UTC]: using the page() application to page across our building...often the meetme conferences don't disconnect after the caller hangs up. Anyone else having this problem. (using Polycom phones)
  • Matthew Richmond, Wed 25 of Jul, 2007 [02:58 UTC]: thanks Nicholas Blasgen! I haven't worked with AGI before, but there's always a first! Thanks again!
  • Nicholas Blasgen, Tue 24 of Jul, 2007 [19:18 UTC]: Matthew Richmond, AGI will handle all that for you.
  • sam, Mon 23 of Jul, 2007 [16:39 UTC]: need help - certain voicemail extension will stop working and recording voicemail on asterisk - anyone know why and how to fix it? Thanks
  • john haji, Mon 23 of Jul, 2007 [14:55 UTC]: free calls to pakistan
  • bong, Sat 21 of Jul, 2007 [19:09 UTC]: hi good day to all can anyone help me how to configured the nortel sip to the signaling server and how to activate in mobile w/ sip compatible without mcs
Server Stats
  • Execution time: 0.38s
  • Memory usage: 2.18MB
  • Database queries: 32
  • GZIP: Disabled
  • Server load: 2.68

sipX_screenshots

sipX - The SIP PBX for Linux - Web Interface Screenshots


sipX is managed using a simple to use Web interface. The following screenshots are available below:


  • 1. Configure Devices - This screen allows the setup of phones and enable them to automatically register with sipX as they are powered up. Profiles can be defined hierarchically in groups and specific configuration files required by the specific model phone are automatically pushed down to the phone as it registeres. This allows the management of large populations of phones across a distributed environment.

  • 2. Configure Services - This screen allows the configuration of the different services offered by sipX.

  • 3. Configure Users - This screen allows the creation of a new user. For every user a profile is created that defines specific user permissions, available services as well as service attributes. This profile can be created hierarchically using groups. Once a user is created successfully, devices are assigned to users in order to project specific user features out to the phones the user wants to use.

  • 4. Retrieve Voicemail - For every user sipX automatically creates a personal Web user portal. Using this portal the user can perform all the different operations offered by the system, from defining speed dial numbers to the retrieval of voicemail messages as well as the personalization of both the phone and the phone service.

Additional Information:

1. Configure Devices

Image

2. Configure Services

Image

3. Configure Users

Image

4. Retrieve Voicemail

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Created by fm77c, Last modification by JimWilson on Mon 11 of Apr, 2005 [16:09 UTC]

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