<div dir="ltr">They're two separate issues. It looks like I referenced the wrong Jira in my reply to Michael's comment. The correct Jira for this issue, involving the 200 OK, is <a href="https://freeswitch.org/jira/browse/FS-7202">FS-7202</a>. Oops. I'll retest shortly.<div><br></div><div>Agreed, the 180-after-183 no audio issue described in <a href="https://freeswitch.org/jira/browse/FS-7207">FS-7207</a> is clearly different. Sorry for the confusion.<div><br></div><div><div><div><br></div><div><br></div><div>- Jeff</div></div></div></div></div><div class="gmail_extra"><br><div class="gmail_quote">On Mon, Mar 16, 2015 at 11:47 AM, Brian West <span dir="ltr"><<a href="mailto:brian@freeswitch.org" target="_blank">brian@freeswitch.org</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">If that is what is going on, then thats a critical detail that wasn't outlined in the JIRA, maybe we can get a new test on the latest code to verify?</div><div class="gmail_extra"><div><div class="h5"><br><div class="gmail_quote">On Mon, Mar 16, 2015 at 10:43 AM, Steven Ayre <span dir="ltr"><<a href="mailto:steveayre@gmail.com" target="_blank">steveayre@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 dir="ltr">Just an observation, but what you wrote in the Jira appears to be different from your original email.<div><br></div><div>'<span style="font-size:12.8000001907349px">In the debug I see where it detects the port change from 183 #1 to 183 #2. As such, I hear early media from both until the 200 OK. When the call connects, I see no such port change in the debug, and since it's still listening on the wrong port (from 183 #2), there is no audio. </span><span style="font-size:12.8000001907349px">I've seen some older posts where Anthony seemed against even the port change from 183 #1 to 183 #2, yet that seems to work okay today. It just doesn't sense the port change from 183 --> 200.'</span></div><div><br></div><div>So it sounded like you heard both ringbacks and nothing after 200. In the Jira you don't talk about the 200 at all, just issues before then. Do you have audio working on receiving 200 now?</div><div><br></div><div><br></div></div><div><div><div class="gmail_extra"><br><div class="gmail_quote">On 16 March 2015 at 14:18, Jeff Pyle <span dir="ltr"><<a href="mailto:jpyle@fidelityvoice.com" target="_blank">jpyle@fidelityvoice.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">Michael,<div><br></div><div>I don't believe so. I haven't seen any activity in the <a href="https://freeswitch.org/jira/browse/FS-7207" target="_blank">Jira</a>. </div><span><font color="#888888"><div><br></div><div><br></div><div>- Jeff</div><div><br></div></font></span></div><div><div><div class="gmail_extra"><br><div class="gmail_quote">On Fri, Mar 13, 2015 at 10:18 PM, Michael St-Georges <span dir="ltr"><<a href="mailto:m.stgeorges@csl-technologies.com" target="_blank">m.stgeorges@csl-technologies.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><span>Steven Ayre <steveayre@...> writes:<br>
<br>
><br>
><br>
> This sounds like it belongs on Jira so the issue can be tracked.<br>
><br>
</span>> On 27 January 2015 at 00:04, Jeff Pyle <jpyle-<br>
<div><div>eOOfO1YW0K2EOSkOl7zanAC/<a href="mailto:G2K4zDHf@public.gmane.org" target="_blank">G2K4zDHf@public.gmane.org</a>> wrote:<br>
> Hello,<br>
> The following is on FreeSWITCH Version<br>
1.5.15b+git~20150126T215733Z~c16f9ec1d9~64bit (git c16f9ec 2015-01-26<br>
21:57:33Z 64bit).<br>
><br>
> The design goal for this configuration is that of a simple transcoding<br>
SBC. SIP calls arrive with various supported codecs, SIP calls bridge<br>
out on PCMU. No users, no auth, etc. Overall, it seems to work but<br>
there is one call flow I'm struggling with.<br>
><br>
> The B-leg of calls are bridged to a PSTN gateway. If the gateway's<br>
signaling follows 100, 183, 200, all is well. But if the gateway sends<br>
multiple 183s with different RTP ports, there is no audio when the call<br>
goes to 200. See the following example:<br>
><br>
> - Gateway signals 183 with SDP indicating audio on port 16384. -<br>
Gateway signals 183 with SDP indicating audio on port 16386. - Gateway<br>
signals 200 with SDP indicating audio on port 16384 (same as original<br>
183).<br>
><br>
> In the debug I see where it detects the port change from 183 #1 to 183<br>
#2. As such, I hear early media from both until the 200 OK. When the<br>
call connects, I see no such port change in the debug, and since it's<br>
still listening on the wrong port (from 183 #2), there is no audio.<br>
><br>
> I've seen some older posts where Anthony seemed against even the port<br>
change from 183 #1 to 183 #2, yet that seems to work okay today. It<br>
just doesn't sense the port change from 183 --> 200. I don't know if<br>
this is a feature, bug, or misconfigured option. Thoughts are welcome!<br>
><br>
><br>
> - Jeff<br>
><br>
><br>
><br>
><br>
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Was this ever fixed? I have a similar problem with the behavior of a<br>
Samsung PBX that connects to a FreeSwitch.<br>
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<p><font face="courier new, monospace"><b><i><font size="4">Brian West</font></i></b><br><span style="font-size:x-small"><a href="mailto:brian@freeswitch.org" target="_blank">brian@freeswitch.org</a></span></font></p>
<p><font size="1" face="courier new, monospace"><img src="http://billing.freeswitch.org/templates/default/img/whmcslogo.png"><br></font></p><p><font face="courier new, monospace"><b><i>Twitter: @FreeSWITCH , @briankwest</i></b><br><a href="http://www.freeswitchbook.com" target="_blank">http://www.freeswitchbook.com</a><br><a href="http://www.freeswitchcookbook.com" target="_blank">http://www.freeswitchcookbook.com</a></font></p>
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