These are on outbound calls through pennytel, correct? Find out if the 183 that they send has media or not. Most likely they&#39;re doing something silly like sending a 183 w/media but not actually sending media. A pcap analyzed in wireshark would help you see exactly what they are doing. Once you know that then you can probably figure out what the next step should be.<div>
<br></div><div>-MC<br><br><div class="gmail_quote">On Mon, Mar 21, 2011 at 3:30 AM, Mark Campbell-Smith <span dir="ltr">&lt;<a href="mailto:mcampbellsmith@gmail.com">mcampbellsmith@gmail.com</a>&gt;</span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
Hi!<div><br></div><div>I have an issue with no ringback or audio that has just started in the last few weeks.  No changes in FS, which has been rock solid for a long time.  I think my voip provider Pennytel has upgraded or changed their servers.</div>

<div><br></div><div>Anyway, the difference in signalling is as follows:</div><div><br></div><div>This does not work:</div><div><div>INVITE <a href="mailto:sip%3Anumber@sip.pennytel.com" target="_blank">sip:number@sip.pennytel.com</a> SIP/2.0</div>

<div>SIP/2.0 100 Trying</div><div>SIP/2.0 401 Unauthorized</div><div>ACK <a href="mailto:sip%3Anumber@sip.pennytel.com" target="_blank">sip:number@sip.pennytel.com</a> SIP/2.0</div><div>INVITE <a href="mailto:sip%3Anumber@sip.pennytel.com" target="_blank">sip:number@sip.pennytel.com</a> SIP/2.0</div>

<div>SIP/2.0 100 Trying</div><div>SIP/2.0 183 Session Progress</div><div>SIP/2.0 180 Ringing</div><div>SIP/2.0 200 OK</div><div>ACK sip:<a href="http://202.85.243.105:5061" target="_blank">202.85.243.105:5061</a> SIP/2.0</div>
<div>BYE sip:<a href="http://202.85.243.105:5061" target="_blank">202.85.243.105:5061</a> SIP/2.0</div>
<div>SIP/2.0 200 OK</div><div><br></div><div><span style="white-space:pre-wrap">	</span></div><div>This works</div><div>INVITE <a href="mailto:sip%3Anumber@sip.pennytel.com" target="_blank">sip:number@sip.pennytel.com</a> SIP/2.0</div>

<div>SIP/2.0 100 Trying</div><div>SIP/2.0 401 Unauthorized</div><div>ACK <a href="mailto:sip%3Anumber@sip.pennytel.com" target="_blank">sip:number@sip.pennytel.com</a> SIP/2.0</div><div>INVITE <a href="mailto:sip%3Anumber@sip.pennytel.com" target="_blank">sip:number@sip.pennytel.com</a> SIP/2.0</div>

<div>SIP/2.0 100 Trying</div><div>SIP/2.0 183 Session Progress</div><div>SIP/2.0 200 OK</div><div>ACK sip:<a href="http://202.85.243.105:5061" target="_blank">202.85.243.105:5061</a> SIP/2.0</div></div><div><br></div><div>
And I see after the Ringing signal in the case that does not work (there is no ringing signal in the case that does work):</div>
<div><div>2011-03-21 20:28:25.677621 [DEBUG] sofia.c:4641 Channel sofia/external/0393011079 skipping state [proceeding][180] received after a RINGING</div></div><div><br></div><div>Can anyone point me to a reason for the difference in signalling and what I can do to get audio 100% of the time?</div>

<div><br></div><div>THanks</div>
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<br></blockquote></div><br></div>