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<html> <body> <h2>sipcmd2 - simple SIP command line softphone</h2> <h3>Introduction</h3> <p> Updated sipcmd command line soft phone that makes phone calls, accepts calls, enters DTMF digits, plays back WAV files and records them. A useful testing tool for VoIP systems. Runs on Linux. </p> <h3>Install</h3> <h4>Dependencies</h4> <p> <h5>Amazon Linux 2 / RHEL 7 / CentOS 7</h5> <code> yum install opal-devel ptlib-devel && yum groupinstall 'Development Tools' </code> <h5>Ubuntu</h5> <code> apt-get install opal-dev ptlib-dev </code> <h5>Ubuntu 14.10</h5> <code> apt-get install libopal-dev </code> <h5>Ubuntu 12.10</h5> <code>apt-get install libpt-dev libopal-dev</code> </p> <h5>Debian</h5> <code> apt-get install libopal-dev libpt-dev </code> <br> <br> <h4>Download</h4> <code> git clone https://github.com/guisousanunes/sipcmd2.git </code> <br> <br> <h3>Compile</h3> <p> <code> make </code> <br> <br> (To disable debug messages, comment out DEBUG flag from Makefile) </p> <h3>Environment</h3> <p> If you compile the dependencies from source, make sure that libpt and libopal are in your <code>LD_LIBRARY_PATH</code>. The default installation location is <code>/usr/local/lib</code>. </p> <h3>Run</h3> <b>Run options:</b> <pre> -u <name> --user <name> username (required) -c <passw> --password <passw> password for registration -a <name> --alias <name> username alias -l <addr> --localaddress <addr> local address to listen on -o <file> --opallog <file> enable extra opal library logging to file -p <port> --listenport <port> the port to listen on -P <proto> --protocol <proto> sip/h323/rtp (required) -r <nmbr> --remoteparty <nmbr> the party to call to -x <prog> --execute <prog> program to follow -d <prfx> --audio-prefix <prfx> recorded audio filename prefix -f <file> --file <file> the name of played sound file -g <addr> --gatekeeper <addr> gatekeeper to use -w <addr> --gateway <addr> gateway to use </pre> <p> <code>-l</code> or <code>-p</code> without <code>-x</code> assumes answer mode. Additional <code>-r</code> forces caller id checking. <code>-r</code> without <code>-l</code>, <code>-p</code> or <code>-x</code> assumes call mode. <br> To register to a gateaway, specify <code>-c</code>, <code>-g</code> and <code>-w</code> <br><b>Example:</b><br><br> <code> ./sipcmd -P sip -u [username] -c [password] -w [server] -x "c<number>;w200;d12345" </code> </p> <br> <b>WAV file requirements:</b> <ul> <li>mono <li> 8 kHz sampling rate <li> 16 bits sample size </ul><b>The EBNF definition of the program syntax:</b>
<pre> prog := cmd ';' <prog> | cmd := call | answer | hangup | dtmf | voice | record | wait | setlabel | loop call := 'c' remoteparty answer := 'a' [ expectedremoteparty ] hangup := 'h' dtmf := 'd' digits voice := 'v' audiofile record := 'r' [ append ] [ silence ] [ iter ] millis audiofile append := 'a' silence := 's' closed := 'c' iter := 'i' activity:= 'a' wait := 'w' [ activity | silence ] [ closed ] millis setlabel:= 'l' label loop := 'j' [ how-many-times ] [ 'l' label ] </pre><b>Examples:</b><br><br> <code> "l4;c333;ws3000;d123;w200;lthrice;ws1000;vaudio;rsi4000f.out;j3lthrice;h;w2000;j4" </code> <br><br> Parses to the following:
<ol> <li> do this four times: <ol> <li> call to 333 <li> wait until silent (max 3000 ms) <li> send dtmf digits 123 <li> wait 200 ms <li> do this three times: <ol> <li> wait until silent (max 1000 ms) <li> send sound file 'audio' <li> record until silent (max 4000 ms) to files 'f-[0-3]-[0-2].out' </ol> <li> hangup <li> wait 2000 ms </ol> </ol> <br><br> <code> "lstart;a;lwai;w1000;j10lwai;d5;lafterwai;w1000;j10lafterwai;h;j0lstart" </code> <br><br> Parses to the following: <ol> <li> do this infinity times: <ol> <li> wait for a call and answer <li> wait 10 s (using label and jump) <li> send dtmf 5 <li> wait 10 s (using label and jump) <li> do this three times: <li> hangup </ol> </ol> <hr> <br> Credits for sipcmd creator tmakkonen in project <a href="https://github.com/tmakkonen/sipcmd">sipcmd</a>. </body> </html>