Posts with the name or tag of FreeSWITCH;

by emran

Dialing out via Gateway in FreeSWITCH

12:43 pm in FreeSWITCH by emran

Add the following to default.xml (in prefix/conf/dialplan) for 11 digit dialing (US) via your newly created gateway configuration above. Remember, the default configuration provided by FreeSWITCH does not include a diaplan rule for outbound calls via the gateway you’ve defined. You will need to add something similar to the following.

<!– Dial any 10 digit number (1223334444) or 1+10 number (11223334444) here –>
   <extension name=”Long Distance – isoftswitch”>
    <condition field=”destination_number” expression=”^(1{0,1}\d{10})$”>
      <action application=”set” data=”effective_caller_id_number=11223334444″/>
      <!– If your provider does not provide ringback (180 or 183) you may simulate
        ringback by uncommenting the following line. –>
      <!– action application=”ringback” /–>
      <action application=”bridge” data=”sofia/gateway/isoftswitch/$1″/>
     </condition>
   </extension>

by emran

Nokia TLS Setup / Symbian S60 TLS Setup for FreeSWITCH

12:35 pm in FreeSWITCH, Symbian by emran

This configuration is only for SIP TLS. SRTP hasn’t been tested.

  • To import a root certificate into the device, it must be in DER encoded format. To setup a certificate that is create using gentls_cert script use the following command
openssl x509 -in /usr/local/freeswitch/conf/ssl/CA/cacert.pem -out /usr/local/freeswitch/conf/ssl/CA/cacert.cer -outform DER
  • Copy the new cacert.cer to the device, and run it in file manager. This will prompt you the save it as a trusted certificate. Do this and select all trust levels during installation.
  • Setup your SIP Profile using “sips:” instead of “sip:” in Public Username, Proxy Server Address and Register Server Address, and set the appropriate port for your FreeSWITCH installation.

ATTENTION!!! Set the “Transport type” to “AUTO” or it will not work!

by emran

how to send SIP traffic Flood by sipsak?

9:50 pm in Asterisk, FreeSWITCH, OpenSIPS, Yate by emran

sipsak -F -vvv -u 500 -H 192.168.11.33 -l 5060 -s sip:192.168.11.52

by emran

How to capture SIP Packet with Tcpdump ?

9:49 pm in Asterisk, FreeSWITCH, OpenSIPS, Yate by emran

tcpdump -i eth0 -n -s0 -v udp port 5060

by emran

Free SIP Out Bound Proxy Service

9:36 pm in Asterisk, FreeSWITCH, OpenSIPS, Yate by emran

Your free SIP Outbound Proxy Service, bypass local  ISPs’ blocking, start using your broadband phone.

http://freesps.googlepages.com/

by emran

IAX2 Failover in FreeSWITCH Dialplan

9:32 pm in FreeSWITCH by emran

Here is the dialplan settings:

<extension name=”FreeSWITCH routing to IAX VENDORS”>
<condition field=”destination_number” expression=”^8888$”>
<action application=”set” data=”hangup_after_bridge=true”/>
<action application=”set” data=”continue_of_fail=true”/>
<action application=”set” data=”call_timeout=5″/>
action application=”set” data=”no_media=true”/>
<action application=”bridge” data=”iax/marko:moofoo atXXX.XXX.XXX.242 /1234567890″/>
<action application=”bridge” data=”iax/marko:moofoo at XXX.XXX.XXX.243 /1234567890″/>
</condition>
 </extension>