Hi,<br><br>It worked finally with GSM Codec only enabled at client side..
Initially with G.711 (u-low) , G.711 (A-low) and GSM it didn't work. All
enabled<br>
<br>by setting [CLI] sip set debug on<br>I saw asterisk having following logs..<br><br> -- Remotely bridging SIP/macbook-00000041 and SIP/tharindu-00000042<br>set_destination: Parsing <<a href="mailto:sip%3Atharindu@192.168.1.3">sip:tharindu@192.168.1.3</a>:<div id=":1cg">
64540;ob> for address/port to send to<br>
set_destination: set destination to <a href="http://192.168.1.3:64540/" target="_blank">192.168.1.3:64540</a><br>Audio is at 5060<br>Adding codec 0x2 (gsm) to SDP<br>Adding non-codec 0x1 (telephone-event) to SDP<br><br><br>
-- <br><br>So I enabled GSM only .. then everything got solved.. :)<br><br>---<br><br>I would like to register SIP users to the server using kind of web service.. instead of manually entering extensions and users using configuration files..<br>
<br>TO achieve this could some body point some instructions. ??<br><br>One more thing.. <br><br>Is it possible to automatically reload the servers after some small time ?? instead of manually typing the command on [CLI] console. ??<br>
<br>Thanks and Kind Regards,<br><br>Tharindu Madushanka,<br><a href="http://tharindufit.wordpress.com">tharindufit.wordpress.com</a><br></div>