Open Source SIP "PBX"?

Discussion in 'VOIP' started by Georg Schwarz, Apr 13, 2005.

  1. I am looking for some piece of Open Source software to implement a SIP
    "PBX" on a Unix system.
    Ideally, it should be able to do the following:

    * provide SIP registry server to local SIP phones
    * register itself to potentially multiple outside SIP registry servers
    to receive SIP calls from outside under different SIP addresses (which
    providers might have mapped to PSTN numbers).
    * Provide some screening logic for incoming SIP calls
    * Forward incoming outside SIP calls to local SIP phones according to a
    definable logic
    * Doing so it should act as a media gateway as well, i.e. it should be a
    complete application-level gateway between local SIP phones and the
    outside world (involves e.g. some SIP header rewriting)
    * It should do so as well for outgoing calls
    * It should be able to implement a private numbering plan/call routing
    logic for outgoing calls including the use of different SIP caller
    addresses depending on the destination.
    * It should be able to do codec conversions, i.e. allowing the use of a
    compressing codec for outside incoming and outgoing calls with a local
    SIP phone which only supports G.711.
    * It should be able to color traffic by setting the IPv4 ToS bits to a
    defined value for outgoing VoIP packets, thus setting a basis for
    CoS-aware QoS.

    Is Asterisk a good starting point to look for these features?
    Or is there any other software you can recommend taking a look at?


    --
    Georg Schwarz http://home.pages.de/~schwarz/
    +49 178 8545053
     
    Georg Schwarz, Apr 13, 2005
    #1
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  2. Georg Schwarz wrote:
    > I am looking for some piece of Open Source software to implement a SIP
    > "PBX" on a Unix system.

    ok.

    >
    > Is Asterisk a good starting point to look for these features?

    yes.

    > Or is there any other software you can recommend taking a look at?

    no.

    Wolfgang
     
    Wolfgang Barth, Apr 13, 2005
    #2
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  3. On 04/14/2005 12:45 AM, Georg Schwarz wrote:
    > I am looking for some piece of Open Source software to implement a SIP
    > "PBX" on a Unix system.

    [snip]
    > Is Asterisk a good starting point to look for these features?


    Yes.

    > Or is there any other software you can recommend taking a look at?


    Yes, have a look at http://YATE.null.ro/ too and CVS version. A newer
    version of YATE with lots of new features is also under development.

    --
    Dr Balwinder Singh Dheeman Registered Linux User: #229709
    CLLO (Chief Linux Learning Officer) Machines: #168573, 170593, 259192
    Anu's Linux@HOME Distros: Ubuntu, Fedora, Knoppix
    More: http://anu.homelinux.net/~bsd/ Visit: http://counter.li.org/
     
    Dr Balwinder S Dheeman, Apr 14, 2005
    #3
  4. Georg Schwarz

    B.M. Wright Guest

    Georg Schwarz <> wrote:
    > I am looking for some piece of Open Source software to implement a SIP
    > "PBX" on a Unix system.
    > Ideally, it should be able to do the following:


    > * provide SIP registry server to local SIP phones
    > * register itself to potentially multiple outside SIP registry servers
    > to receive SIP calls from outside under different SIP addresses (which
    > providers might have mapped to PSTN numbers).
    > * Provide some screening logic for incoming SIP calls
    > * Forward incoming outside SIP calls to local SIP phones according to a
    > definable logic
    > * Doing so it should act as a media gateway as well, i.e. it should be a
    > complete application-level gateway between local SIP phones and the
    > outside world (involves e.g. some SIP header rewriting)
    > * It should do so as well for outgoing calls
    > * It should be able to implement a private numbering plan/call routing
    > logic for outgoing calls including the use of different SIP caller
    > addresses depending on the destination.
    > * It should be able to do codec conversions, i.e. allowing the use of a
    > compressing codec for outside incoming and outgoing calls with a local
    > SIP phone which only supports G.711.
    > * It should be able to color traffic by setting the IPv4 ToS bits to a
    > defined value for outgoing VoIP packets, thus setting a basis for
    > CoS-aware QoS.


    > Is Asterisk a good starting point to look for these features?
    > Or is there any other software you can recommend taking a look at?
    >


    You might want to check out Sip Express Router (free, go google,
    no link handy) or sipXpbx (free as well, at http://www.sipfoundry.org).
    You'll have to look through the features list, but I believe most or all
    of those things are covered. These are lighter weight, probably better
    supported on non-Linux platforms, but lack the hardware support and some
    modularity of Asterisk (if you need those).
     
    B.M. Wright, Apr 14, 2005
    #4
  5. Georg Schwarz

    Guest

    The asterisk seems to be good and gives above functions
    But now a new group has statred for SIP PBX with opensource SW and you
    can visit the following site. It is also SIP IPBX.
    http://yate.null.ro


    www.softswitches.net
     
    , Apr 14, 2005
    #5
  6. B.M. Wright <> wrote:

    > You might want to check out Sip Express Router (free, go google,
    > no link handy) or sipXpbx (free as well, at http://www.sipfoundry.org).
    > You'll have to look through the features list, but I believe most or all
    > of those things are covered. These are lighter weight, probably better
    > supported on non-Linux platforms, but lack the hardware support and some
    > modularity of Asterisk (if you need those).


    do they include a media gateway? Can they act as an application level
    proxy for SIP calls?


    --
    Georg Schwarz http://home.pages.de/~schwarz/
    +49 178 8545053
     
    Georg Schwarz, Apr 16, 2005
    #6
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