<div><div dir="auto">On latest versions I’m trying the sipdump module, the one benefit I’m liking over its alternatives is that you can enable/disable dumping to file via RPC..</div></div><div dir="auto"><br></div><div dir="auto">So in normal operation you are not dumping all tls traffic, and then you can enable it only when you need to troubleshoot something.. </div><div dir="auto"><br></div><div dir="auto">Just throwing this out there in case it’s useful :) </div><div dir="auto"><br></div><div dir="auto">Cheers,</div><div dir="auto">Joel. </div><div><br><div class="gmail_quote"><div dir="ltr" class="gmail_attr">On Sun, Aug 2, 2020 at 05:15 Karsten Horsmann <<a href="mailto:khorsmann@gmail.com">khorsmann@gmail.com</a>> wrote:<br></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="auto"><div>Hi,</div><div dir="auto"><br></div><div dir="auto"><br></div><div dir="auto">use siptrace to see what Kamailio sees on the tls side or use the message buffer pv $mb</div><div dir="auto"><br></div><div dir="auto">Like this (please check Syntax at your own, copy pasted that with my smartphone so no guarantees) </div><div dir="auto"><br></div><div dir="auto">It shows the raw message (unencrypted) before tls module it handles or after its decrypted AFAIK. </div><div dir="auto"><br></div><div dir="auto"><div dir="auto">event_route[network:msg] {</div><div dir="auto"> if (is_incoming()) { xlog("L_INFO", "Received message '$mb' \n");</div><div dir="auto"> } else {</div><div dir="auto"> xlog("L_INFO", "Sending message '$mb' \n");</div><div dir="auto">}</div><div dir="auto">} </div><div dir="auto"><br></div><div dir="auto"><br></div>Kind regards </div></div><div dir="auto"><div dir="auto">Karsten Horsmann </div><div dir="auto"><br></div><div dir="auto"><br><div class="gmail_quote" dir="auto"><div dir="ltr" class="gmail_attr">tangd122 <<a href="mailto:tangd122@gmail.com" target="_blank">tangd122@gmail.com</a>> schrieb am Sa., 1. Aug. 2020, 19:40:<br></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Thank you for your replies! Outbound calling is now completely working. <br>
<br>
For incoming calls from SIP trunk to Teams there's still no ACK and CANCEL<br>
messages. Which causes disconnect in 20 seconds and no audio. <br>
<br>
I think there's something wrong with the routing or via headers in the 200<br>
OK but don't know what. <br>
<br>
The complete sip dump: <a href="https://paste.projectdev.org/imiluwecaf" rel="noreferrer noreferrer" target="_blank">https://paste.projectdev.org/imiluwecaf</a> <br>
<br>
Unfortunately i can't see the sip messages from the SIP provider side. <br>
<br>
<br>
<br>
--<br>
Sent from: <a href="http://sip-router.1086192.n5.nabble.com/Users-f3.html" rel="noreferrer noreferrer" target="_blank">http://sip-router.1086192.n5.nabble.com/Users-f3.html</a><br>
<br>
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