<html><head><meta http-equiv="Content-Type" content="text/html; charset=us-ascii"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; line-break: after-white-space;" class="">Google search for SIP ALG problem to see if this is relevant for your case.<div class=""><br class=""></div><div class=""><div class="">Regards,</div><div class="">Adrian</div><div class=""><br class=""></div><div class=""><div><br class=""><blockquote type="cite" class=""><div class="">On 13 Jan 2021, at 13:08, Mark Allen <<a href="mailto:mark@allenclan.co.uk" class="">mark@allenclan.co.uk</a>> wrote:</div><br class="Apple-interchange-newline"><div class=""><div dir="ltr" class="">Hi all - I've been banging my head against this but not succeeding.<div class=""><br class=""></div><div class="">Our setup...</div><div class=""><br class=""></div><div class=""><font face="monospace" class="">UAC 192.168.x.x</font></div><div class=""><font face="monospace" class=""> | </font></div><div class=""><span style="font-family:monospace" class="">Router 5.x.x.x</span></div><div class=""><span style="font-family:monospace" class=""> |</span></div><div class=""><span style="font-family:monospace" class="">(internet)</span></div><div class=""><span style="font-family:monospace" class=""> | </span></div><div class=""><span style="font-family:monospace" class="">Firewall 46.x.x.x maps</span></div><div class=""><span style="font-family:monospace" class=""> | directly to</span></div><div class=""><span style="font-family:monospace" class="">OpenSIPS 192.168.x.x Mid-registrar</span><br class=""></div><div class=""><span style="font-family:monospace" class=""> |</span></div><div class=""><span style="font-family:monospace" class="">Asterisk 192.168.x.x</span></div><div class=""><span style="font-family:monospace" class=""><br class=""></span></div><div class=""><br class=""></div><div class=""><font face="monospace" class="">Current situation: </font></div><div class=""><font face="monospace" class="">- UAC can register on Asterisk via OpenSIPS</font></div><div class=""><font face="monospace" class="">- UAC can call destination registered on Asterisk on local n/w to Asterisk box</font></div><div class=""><font face="monospace" class="">- Destination extension rings and can pick up call</font></div><div class=""><font face="monospace" class="">- There is no audio either way & call drops after about 30 secs (Asterisk kills call with "</font>Requested channel not available" <span style="font-family:monospace" class="">because not RTP traffic is reaching destination)</span></div><div class=""><span style="font-family:monospace" class=""><br class=""></span></div><div class=""><span style="font-family:monospace" class="">I have tried passing audio through Mediaproxy on OpenSIPS box but with no success. Using Wireshark I can see RTP traffic initiated at both ends, but it doesn't reach the other end either way.</span></div><div class=""><span style="font-family:monospace" class=""><br class=""></span></div><div class=""><span style="font-family:monospace" class="">Is there some definitive guide to setting this up correctly or are there specific steps that I need to follow? </span></div><div class=""><font face="monospace" class=""><br class=""></font></div></div>
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