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/$1″/>
</condition>
</extension>
12:35 pm in FreeSWITCH, Symbian by emran
This configuration is only for SIP TLS. SRTP hasn’t been tested.
- To import a root certificate into the device, it must be in DER encoded format. To setup a certificate that is create using gentls_cert script use the following command
openssl x509 -in /usr/local/freeswitch/conf/ssl/CA/cacert.pem -out /usr/local/freeswitch/conf/ssl/CA/cacert.cer -outform DER
- Copy the new cacert.cer to the device, and run it in file manager. This will prompt you the save it as a trusted certificate. Do this and select all trust levels during installation.
- Setup your SIP Profile using “sips:” instead of “sip:” in Public Username, Proxy Server Address and Register Server Address, and set the appropriate port for your FreeSWITCH installation.
ATTENTION!!! Set the “Transport type” to “AUTO” or it will not work!
9:50 pm in Asterisk, FreeSWITCH, OpenSIPS, Yate by emran
sipsak -F -vvv -u 500 -H 192.168.11.33 -l 5060 -s sip:192.168.11.52
9:49 pm in Asterisk, FreeSWITCH, OpenSIPS, Yate by emran
tcpdump -i eth0 -n -s0 -v udp port 5060
9:36 pm in Asterisk, FreeSWITCH, OpenSIPS, Yate by emran
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9:32 pm in FreeSWITCH by emran
Here is the dialplan settings:
<extension name=”FreeSWITCH routing to IAX VENDORS”>
<condition field=”destination_number” expression=”^8888$”>
<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>
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