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Has anyone used Freeswitch as a simple SIP TCP/UDP bridge?

    Question

  • If so, would you mind posting a sample copy of your SIP config file and dialplan?
    Tuesday, January 25, 2011 4:00 PM

All replies

  • http://www.gmc.cc.ga.us/drop/FreeSwitch_Lync_integration_guide.pdf

    Enjoy

     

    Drago


    http://ocsdude.blogspot.com | MVP Snom OCS Edition
    Tuesday, January 25, 2011 4:15 PM
  • I saw that guide but it didnt work right.  I did figure out a simple working config & can post if anyone else needs it...
    Wednesday, January 26, 2011 6:00 PM
  • yes please, also having trouble with drago's guide.
    Friday, May 20, 2011 7:04 PM
  • Hi,

    could you please post it as you promice to. Thank you in advance.

     

    Thursday, August 25, 2011 3:09 PM
  • Hi Guys,

     

    We have used Drago's guide which I believe is no longer available online anymore with some variations to suite our environment and after some troubleshooting we were upa nd running. We have set this up and running well twice now.


    Usually the problems we were having was making sure FS is binding to the correct IPS. And also we had problems with the dialplan we put all dialpnal rules in one file and did static pointing to the correct route in order for it to work correctly, i.e. $1@ip:port in dialplan bridge.
    Thursday, November 24, 2011 12:29 PM
  • Yes please post your config!
    Monday, November 28, 2011 6:27 AM
  • I do not have a config guide.

    Let me know where you are stuck and we will see how we can help.


    Kyriacos Aristodemou
    Monday, November 28, 2011 10:05 AM
  • I dont remember which settings were part of the defaults, but these should be the relevant sections.  You will need to change IP's and ports to match your environment.  Let me know if there is anything I missed :)

    From "vars.xml"

     <X-PRE-PROCESS cmd="set" data="sip_tls_version=tlsv1"/>

     

      <!-- Internal SIP Profile -->

      <X-PRE-PROCESS cmd="set" data="internal_auth_calls=true"/>

      <X-PRE-PROCESS cmd="set" data="internal_sip_port=5058"/>

      <X-PRE-PROCESS cmd="set" data="internal_tls_port=5059"/>

      <X-PRE-PROCESS cmd="set" data="internal_ssl_enable=false"/>

      <X-PRE-PROCESS cmd="set" data="internal_ssl_dir=$${base_dir}/conf/ssl"/>

     

      <!-- External SIP Profile -->

      <X-PRE-PROCESS cmd="set" data="external_auth_calls=false"/>

      <X-PRE-PROCESS cmd="set" data="external_sip_port=5081"/>

      <X-PRE-PROCESS cmd="set" data="external_tls_port=5082"/>

      <X-PRE-PROCESS cmd="set" data="external_ssl_enable=false"/>

      <X-PRE-PROCESS cmd="set" data="external_ssl_dir=$${base_dir}/conf/ssl"/>

    </include>

     

    From the conf/sip_profiles/internal folder, I called my config "lync.xml";

    <include>

      <gateway name="lync">

          <param name="username" value="none"/>

          <param name="password" value="none"/>

          <param name="caller-id-in-from" value="true"/>

          <param name="proxy" value="172.16.43.2:5068"/>

          <param name="register" value="false"/>

          <param name="register-transport" value="tcp"/>

      </gateway>

    </include>

     

    From the conf/sip_profiles/external folder, I called my config "trixbox.xml";

    <include>

        <gateway name="Trixbox">

          <param name="username" value="none"/>

          <param name="password" value="none"/>

          <param name="caller-id-in-from" value="true"/>

          <param name="proxy" value="172.16.39.7"/>

          <param name="register" value="false"/>

          <param name="register-transport" value="udp"/>

        </gateway>

    </include>

     

    From the conf/dialplan/public folder.
    00_outbound_did.xml;

    <extension name="outgoing">

      <condition field="network_addr" expression="^172\.16\.43\.2$"/>

      <condition field="destination_number" expression=".*" continue="on-true">

       <action application="bridge" data="sofia/gateway/Trixbox/${destination_number}"/>

      </condition>

    </extension>

    </include>

     

    01_inbound_did.xml

    <extension name="INBOUND">

      <condition field="destination_number" expression=".*" continue="on-true">

             <action application="bridge" data="sofia/gateway/lync/${destination_number}"/>

        </condition>

    </extension>

     

    Monday, November 28, 2011 3:59 PM
  • Hi Urnvs,

    I looked at what you said above quoted below:

    From the conf/dialplan/public folder.
    00_outbound_did.xml;

     

    <extension name="outgoing">

      <condition field="network_addr" expression="^172\.16\.43\.2$"/>

      <condition field="destination_number" expression=".*" continue="on-true">

       <action application="bridge" data="sofia/gateway/Trixbox/${destination_number}"/>

      </condition>

    </extension>

    </include>

     

     

    01_inbound_did.xml

     

    <extension name="INBOUND">

      <condition field="destination_number" expression=".*" continue="on-true">

             <action application="bridge" data="sofia/gateway/lync/${destination_number}"/>

        </condition>

    </extension>

     

     

    So basicly instead of doing normalisation rules in your free switch you are saying doing in your oubound xml your configuring it that what ever comes from lync sevre ip is forwarded to your sip trunk (trixbox) and in inbound everything goes to lync server. this works as outbound is checked first as the file name starts with 00.

    Please let me know if i understood that correctly and that you have tested this and it wroks well!


    Kyriacos Aristodemou
    Saturday, December 10, 2011 4:26 PM
  • I also think you need to configure acl.xml to allow domain names or ip address to communicate correctly, eitherwise they are blocked by FS and you get access denied by acl when the calls are trying to be routed.
    Kyriacos Aristodemou
    Saturday, December 10, 2011 4:28 PM
  • Hi URNVS
    I have been trying to implement your solution to the dial plan and it does not seem to work. Can you please assist?
    Thanks

    Kyriacos Aristodemou
    Tuesday, January 10, 2012 3:18 PM
  • I too am having problems with the dial plan. It says: "sofia/external/mobe@192.168.2.23 has executed the last dialplan instruction, hanging up."

    Basically I'm trying to do the bridging to a UCMA 4.0 service so that it can respond, and also make outgoing calls.

    Thanks!


    • Edited by JohnGalt Wednesday, May 08, 2013 9:46 PM
    Wednesday, May 08, 2013 9:43 PM