<div>Is there even a valid use case for Freeswitch to set a channel to RINGING before the actual ring is signaled by the far endpoint? What other evidence would be helpful to diagnose the issue?</div><br clear="all"><div>
Hynek<br></div>
<br><br><div class="gmail_quote">On Tue, Feb 12, 2013 at 12:03 PM, Hynek Cihlar <span dir="ltr"><<a href="mailto:hynek.cihlar@gmail.com" target="_blank">hynek.cihlar@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div>When originating a call the respective call channel's call state is set to RINGING right after progress 100 is received. </div><div><br></div><div>Here's the captured flow:</div><div><br></div><div><div><font face="courier new, monospace">|Time | <src ip> |</font></div>
<div><font face="courier new, monospace">| | | <dst ip> | </font></div><div><font face="courier new, monospace">|2.848 | INVITE SDP (g711A g711U GSM telephone-eventRTP...e-101 CN) |SIP From: "" <sip:endpoint removed To:<sip:endpoint removed</font></div>
<div><font face="courier new, monospace">| |(5080) ------------------> (5060) |</font></div><div><font face="courier new, monospace">|2.848 | 407 Proxy Authentication Required |SIP Status</font></div>
<div><font face="courier new, monospace">| |(5080) <------------------ (5060) |</font></div><div><font face="courier new, monospace">|2.849 | ACK | |SIP Request</font></div>
<div><font face="courier new, monospace">| |(5080) ------------------> (5060) |</font></div><div><font face="courier new, monospace">|2.849 | INVITE SDP (g711A g711U GSM telephone-eventRTP...e-101 CN) |SIP From: "" <sip:endpoint removed To:<sip:</font><span style="font-family:'courier new',monospace">endpoint removed</span></div>
<div><font face="courier new, monospace">| |(5080) ------------------> (5060) |</font></div><div><font face="courier new, monospace">|2.850 | 100 Trying| |SIP Status</font></div>
<div><font face="courier new, monospace">| |(5080) <------------------ (5060) |</font></div><div><font face="courier new, monospace">|13.444 | 180 Ringing |SIP Status</font></div>
<div><font face="courier new, monospace">| |(5080) <------------------ (5060) |</font></div><div><font face="courier new, monospace">|13.445 | 183 Session Progress SDP (g711A g711U GSM tele...ne-eventRTPType-101) |SIP Status</font></div>
<div><font face="courier new, monospace">| |(5080) <------------------ (5060) |</font></div><div><font face="courier new, monospace">|13.445 | RTP (g711A) |RTP Num packets:230 Duration:4.574s SSRC:0x1E777E26</font></div>
<div><font face="courier new, monospace">| |(26056) <------------------ (19312) |</font></div><div><font face="courier new, monospace">|13.601 | RTP (g711A) |RTP Num packets:220 Duration:4.419s SSRC:0xA530C519</font></div>
<div><font face="courier new, monospace">| |(26056) ------------------> (19312) |</font></div></div><div><br></div><div>After the 100 Trying is received the switch executes switch_channel_perform_set_running_state (switch_channel.c) and the channel call state is set to RINGING and ESL event CHANNEL_CALLSTATE:RINGING is generated.</div>
<div><br></div><div>I would expect the channel call state to be set to RINGING only after 180 Ringing is received from the far endpoint.</div><div><br></div><div>Could anybody give me a hint what could be wrong or what steps to take to figure out? I am already out of ideas.</div>
<div><br></div><div>Thanks!</div><span><font color="#888888"><div>Hynek<br></div>
</font></span></blockquote></div><br>