FreeSwitch UserGuide or clear example requested



  • Hello,

    I have been trying to use FreeSwitch since today,  and I have to say that I am a bit running in the dark.

    The various howto's I have found concerned mainly a scratch config and are not related in any way to the implementation done on pfSense.
    There is a very little description presented here http://doc.pfsense.org/index.php/FreeSWITCH
    But beside listing the functionality and telling you to buy your hardware in some places, this is useless, to say the less.
    The various images lack clear explanation and examples on the way things are mapped and organized together.

    The port seems very well organized and powerful, but without a clear explanation, even the best interfaces are useless.

    So if you have a working example on how things could be organized using the pfSense interface, I could surely use this.
    I am ready to create a clear and detailed howto…

    From my point of view a simple example which will include :

    • 2 SIP gateways
    • 2 SIP softphone
    • A simple dialplan
    • Routing incoming calls to the softphone
    • Routing outgoing calls to one of the SIP GW

    …should be enough for as a starting point.

    I certainly lack of experience in VoIP gateway, but I am certain I am not the only one out there…



  • I can not load my "internal" profile… I keep on having this error :

    2010-09-20 16:28:37.991317 [ERR] sofia.c:862 Error Creating SIP UA for profile: internal

    I also don't why FreeSwitch is binding on my external IP instead of my internal one…

    Las but not least if I remove my install and reinstall the port, I keep on having the same old config files that I have modified… 
    What do I have to do to have a fresh install ??



  • Is anyone using with success this package ??

    An example of various config files will be very useful for me.

    I am a bit lost…

    Thanks



  • I am trying to figure this one out myself.
    Going to let you know how it went.



  • Ok, today was a good day !

    After three days of fight with this software. Good news is It is working… ! ;-)

    Bad news is that you have to guess everything by yourself… no documentation.
    Hopefully once I have put all the bits and pieces together, I'll write a "REAL" tutorial !!!

    So, from what I have been finding out :

    • You have to have a registered phone (SIP phone in my case) - this is obvious… But if you haven't used that soft before, It is difficult to guess… Specially if you have been using Asterisk BECAUSE: the proxy binding is on port 5080 and not 5060 .

    • Once you have registered your phone It'll show in the status page as :

    =================================================================================================
    Name            external
    Domain Name      N/A
    DBName          sofia_reg_external
    Pres Hosts     
    Dialplan        XML
    Context          public
    Challenge Realm  auto_to
    RTP-IP          82.xx.tt.242
    Ext-RTP-IP      82.xx.tt.242
    SIP-IP          82.xx.tt.242
    Ext-SIP-IP      82.xx.tt.242
    URL              sip:mod_sofia@82.xx.tt.242:5080
    BIND-URL        sip:mod_sofia@82.xx.tt.242:5080
    HOLD-MUSIC      local_stream://moh
    OUTBOUND-PROXY  N/A
    CODECS          PCMU,PCMA,GSM
    TEL-EVENT        101
    DTMF-MODE        rfc2833
    CNG              13
    SESSION-TO      0
    MAX-DIALOG      0
    NOMEDIA          false
    LATE-NEG        false
    PROXY-MEDIA      false
    AGGRESSIVENAT    false
    STUN-ENABLED    true
    STUN-AUTO-DISABLE false
    CALLS-IN        0
    FAILED-CALLS-IN  0
    CALLS-OUT        0
    FAILED-CALLS-OUT 0

    Registrations:

    Call-ID:    ABFFB889A4819E1E6DC67155A2C32F1F90879F58
    User:      102@82.xx.tt.242
    Contact:    "user"
    Agent:      Acrobits Softphone/4.2
    Status:    Registered(UDP)(unknown) EXP(2010-09-21 22:11:50)
    Host:      pfsense.xx.tt.biz
    IP:        192.168.2.19
    Port:      45593
    Auth-User:  102
    Auth-Realm: 82.xx.tt.242

    =================================================================================================

    • I strongly suggest that you connect in debug mode with:

    cd /usr/local/freeswitch/bin

    ./fs_cli -H 82.xx.tt.242 -p your_pass

    _____ ____    ____ _    ___           
              |  / |  / | |  | |         
              | |
      _
    \  | |  | |    | |           
              |  ) | | |__| | | |           
              |
    |  |
    /  _|
    |__|



    Type /help <enter>to see a list of commands

    +OK log level  [7]
    freeswitch@82.xx.tt.242@internal> sofia status profile internal

    • The big trouble with all this config is that you are binding on the WAN interface of your FW which is quite weird, there must be a reason, but It is not obvious (at least from the user PoV).

    • Once you are connected, you will be able to use various command such as dialing 5000 from your client which will lead to a voice menu…

    I have been able to place some calls, not yet to receive them, but I am certain It'll come in time.</enter>



  • Ok, I have solved this… binding on LAN is ok (was due to siproxd beeing started at the same time as FreeSwitch).

    I can not load my "internal" profile… I keep on having this error :

    2010-09-20 16:28:37.991317 [ERR] sofia.c:862 Error Creating SIP UA for profile: internal

    I also don't why FreeSwitch is binding on my external IP instead of my internal one…

    Las but not least if I remove my install and reinstall the port, I keep on having the same old config files that I have modified…
    What do I have to do to have a fresh install ??

    Finaly the fault was all mine.
    After investigating why the port 5060 was taken, I remembered that I had the "siproxd" package installed and started.

    This is the reason why local trafic could not be started.


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