Use pastebin to show us the console log of the call failing. Capture the output from these commands:<div><br></div><div>sofia status</div><div>sofia status profile <your_profile_name></div><div>sofia status profile <your_profile_name> reg</div>
<div><br></div><div>Then enable siptrace and make a test call, capturing the console output:</div><div>sofia global siptrace on</div><div><br></div><div>Put all that stuff in <a href="http://pastebin.freeswitch.org">pastebin.freeswitch.org</a> using "FreeSWITCH Log" as the syntax highlighting. Reply to this thread with the URL and the gang here will take a look.</div>
<div><br></div><div>-MC<br><div><br><div class="gmail_quote">On Sat, Dec 3, 2011 at 8:07 AM, ayobami <span dir="ltr"><<a href="mailto:ayobami@programmer.net">ayobami@programmer.net</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
I just git pulled the Freeswitch source code and compiled, everything went<br>
well, I mean the compilation, but the problem that arose now is sip phones<br>
could not call each other again, now the soft phones could register on the<br>
FS server because I could use them to call 5000, 4000 and some of my custom<br>
numbers, but when I tried to call an extension from another extension, the<br>
operator keep saying that the extension is not available, I have<br>
troubleshooted and dont know what to do, please people help me out.<br>
<br>
--<br>
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