<div dir="ltr"><div>Hi Dovid,</div><div><br></div><div>Thanks for that. Can you explain how the Progress() and/or Playback() actually help the NAT problem? I'm trying to figure out how it tells Asterisk the correct address to send the RTP to.</div><div><br></div></div><br><div class="gmail_quote"><div dir="ltr" class="gmail_attr">On Thu, 3 Dec 2020 at 16:10, Dovid Bender <<a href="mailto:dovid@telecurve.com">dovid@telecurve.com</a>> wrote:<br></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div dir="auto">David,</div><div dir="auto"><br></div><div dir="auto">You should be able to do that via the agi as well. </div><div><br><div class="gmail_quote"><div dir="ltr" class="gmail_attr">On Wed, Dec 2, 2020 at 20:32 David Cunningham <<a href="mailto:dcunningham@voisonics.com" target="_blank">dcunningham@voisonics.com</a>> wrote:<br></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div dir="ltr"><div>Hi Dovid,</div><div><br></div><div>We're using Enswitch so it uses AGI rather than a regular Asterisk dialplan, but perhaps sending it to a custom-made Asterisk context with the lines you suggest could be the best way forward.</div><div><br></div><div>Thank you for that.</div><div><br></div></div><br><div class="gmail_quote"><div dir="ltr" class="gmail_attr">On Thu, 3 Dec 2020 at 13:01, Dovid Bender <<a href="mailto:dovid@telecurve.com" target="_blank">dovid@telecurve.com</a>> wrote:<br></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div dir="ltr">David,<div><br></div><div>Does Asterisk send a 180 or a 183 with SDP? We have found that using these two lines help (where xc is a 500ms blank sound file)</div><div>Exten => _X.,n, Progress()<br>Exten => _X.,n, Playback(xc,noanswer)<br></div><div><br></div></div><br><div class="gmail_quote"><div dir="ltr" class="gmail_attr">On Wed, Dec 2, 2020 at 4:30 PM David Cunningham <<a href="mailto:dcunningham@voisonics.com" target="_blank">dcunningham@voisonics.com</a>> wrote:<br></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div dir="ltr"><div>Hello,</div><div><br></div><div>We have a problem with a SIP doorbell device which sends media one way only, and NAT at the receiving device.<br></div><div><br></div><div>When the doorbell button is pressed
it makes a call to a configured destination. Since the doorbell only sends
and doesn't receive it sends the INVITE with sendonly in the SDP, and
the destination then replies with a 200 OK with recvonly in the SDP.
<br>
<br>The problem is that the destination is behind NAT, and its reply
contains a private network IP in the SDP. Normally Asterisk when nat=yes
works around that by adjusting the destination for RTP to be the
address it actually receives audio from, however because this device is
recvonly Asterisk never receives audio from it. This means Asterisk
keeps trying to send the doorbell's RTP to the private network IP which of course
fails, and the destination never gets the RTP from the doorbell.</div><div><br></div><div>Does anyone know how to work around this issue?</div><div><br></div><div>Thank you in advance,<br></div><div><br></div>-- <br><div dir="ltr"><div dir="ltr"><div><div dir="ltr"><div><div dir="ltr"><div><div dir="ltr"><div><div dir="ltr"><div>David Cunningham, Voisonics Limited<br><a href="http://voisonics.com/" target="_blank">http://voisonics.com/</a><br>USA: +1 213 221 1092<br>New Zealand: +64 (0)28 2558 3782</div></div></div></div></div></div></div></div></div></div></div></div>
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