<div dir="auto">Hi Ernie,<div dir="auto"><br></div><div dir="auto">When one-way audio appear (no matters if there is a VPN or NAT server on the diagram) I simply :</div><div dir="auto"><br></div><div dir="auto">* Enable SIP debug on Asterisk server. Excecute 'sip set debug ip x.x.x.x' on Astrisk CLI, where x.x.x.x is the IP of the phone or SIP peer you want to debug.</div><div dir="auto"><br></div><div dir="auto">* Make a test call and replicate the issue.</div><div dir="auto"><br></div><div dir="auto">* Stop debug with 'sip set debug off'.</div><div dir="auto"><br></div><div dir="auto">* Follow the SIP conversation. Verify that the INVITE message has the correct IP on the contact field and any other related fields.</div><div dir="auto"><br></div><div dir="auto">* On SDP handshake, verify that the ports where the sound is send, is correct.</div><div dir="auto"><br></div><div dir="auto">Normally, one-way audio is faced when one audio stream (example the called audio) is send to the correct IP and Port destination, on the other audio stream (example the caller audio) don't.</div><div dir="auto"><br></div><div dir="auto">Last, if Asterisk is 'behind' another server, you need tell Asterisk what is the external IP so it can inform this IP to your clients.</div><div dir="auto"><br></div><div dir="auto">If you dont want to follow the SIP conversation on plain text, you can make a packet capture on the Asterisk server, instead of SIP debug.</div></div><div class="gmail_extra"><br><div class="gmail_quote">El 19 abr. 2017 16:38, "Mark Wiater" <<a href="mailto:mark.wiater@greybeam.com">mark.wiater@greybeam.com</a>> escribió:<br type="attribution"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">On 4/18/2017 7:40 PM, Ernie Dunbar wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Server network: <a href="http://192.168.0.0/24" rel="noreferrer" target="_blank">192.168.0.0/24</a><br>
OpenVPN network: <a href="http://10.8.0.0/24" rel="noreferrer" target="_blank">10.8.0.0/24</a><br>
Asus network: <a href="http://192.168.1.0/24" rel="noreferrer" target="_blank">192.168.1.0/24</a><br>
<br>
The Asterisk SIP registration appears to be responding properly to this - this is what I see when I do a 'sip show peer' for an Aastra phone that's connecting through the VPN (Asterisk output is truncated):<br>
<br>
ToHost :<br>
Addr->IP : <a href="http://10.8.0.6:5060" rel="noreferrer" target="_blank">10.8.0.6:5060</a><br>
</blockquote>
<br>
If the Asus network is <a href="http://192.168.1.0/24" rel="noreferrer" target="_blank">192.168.1.0/24</a>, and the phone is registering as 10.0.8.6, it looks like NAT is taking place. Would your asterisk server know how to route traffic to <a href="http://192.168.1.0/24" rel="noreferrer" target="_blank">192.168.1.0/24</a>?<br>
<br>
I've always used site-to-site OpenVPN tunnels where the vpn's terminate on the gateway for both the phones and the asterisk server. I've always had rock solid connections between phones and Asterisk.<br>
<br>
-- <br>
______________________________<wbr>______________________________<wbr>_________<br>
-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" rel="noreferrer" target="_blank">http://www.api-digital.com</a> --<br>
<br>
Check out the new Asterisk community forum at: <a href="https://community.asterisk.org/" rel="noreferrer" target="_blank">https://community.asterisk.org<wbr>/</a><br>
<br>
New to Asterisk? Start here:<br>
<a href="https://wiki.asterisk.org/wiki/display/AST/Getting+Started" rel="noreferrer" target="_blank">https://wiki.asterisk.org/wik<wbr>i/display/AST/Getting+Started</a><br>
<br>
asterisk-users mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-users" rel="noreferrer" target="_blank">http://lists.digium.com/mailma<wbr>n/listinfo/asterisk-users</a><br>
</blockquote></div></div>