I'm not getting inbound audio from <a href="http://bandwidth.com">bandwidth.com</a>. Their engineer said the invite that they're sending me looks like this:<br><br><span style="font-family: courier new,monospace;">INVITE sip:+15129616808@67.198.16.18:5060;transport=udp SIP/2.0.</span><br style="font-family: courier new,monospace;">
<span style="font-family: courier new,monospace;">Record-Route: <sip:216.82.224.202;lr;ftag=VPSF506071629460>.</span><br style="font-family: courier new,monospace;"><span style="font-family: courier new,monospace;">Record-Route: <sip:4.79.212.229;lr;ftag=VPSF506071629460>.</span><br style="font-family: courier new,monospace;">
<span style="font-family: courier new,monospace;">Via: SIP/2.0/UDP 216.82.224.202;branch=z9hG4bK6314.4f7b5d05.0.</span><br style="font-family: courier new,monospace;"><span style="font-family: courier new,monospace;">Via: SIP/2.0/UDP 4.79.212.229;branch=z9hG4bK6314.15486fb6.0.</span><br style="font-family: courier new,monospace;">
<span style="font-family: courier new,monospace;">Via: SIP/2.0/UDP 4.68.250.148:5060;branch=z9hG4bK506071629460-1207516720501.</span><br style="font-family: courier new,monospace;"><span style="font-family: courier new,monospace;">From: "BANDWIDTH COM" <<a href="mailto:sip%3A%2B19192282250@4.68.250.148">sip:+19192282250@4.68.250.148</a>>;tag=VPSF506071629460.</span><br style="font-family: courier new,monospace;">
<span style="font-family: courier new,monospace;">To: <<a href="http://sip:+15129616808@4.79.212.229:5060">sip:+15129616808@4.79.212.229:5060</a>>.</span><br style="font-family: courier new,monospace;"><span style="font-family: courier new,monospace;">Call-ID: <a href="mailto:HOUMGC0520090316161653037223@209.244.63.35">HOUMGC0520090316161653037223@209.244.63.35</a>.</span><br style="font-family: courier new,monospace;">
<span style="font-family: courier new,monospace;">CSeq: 1 INVITE.</span><br style="font-family: courier new,monospace;"><span style="font-family: courier new,monospace;">Contact: <sip:+19192282250@4.68.250.148:5060;transport=udp>.</span><br style="font-family: courier new,monospace;">
<span style="font-family: courier new,monospace;">Max-Forwards: 67.</span><br style="font-family: courier new,monospace;"><span style="font-family: courier new,monospace;">Content-Type: application/sdp.</span><br style="font-family: courier new,monospace;">
<span style="font-family: courier new,monospace;">Content-Length: 177.</span><br style="font-family: courier new,monospace;"><span style="font-family: courier new,monospace;">Remote-Party-ID: "BANDWIDTH COM" <<a href="mailto:sip%3A%2B19192282250@4.68.250.148">sip:+19192282250@4.68.250.148</a>>;party=calling ;screen=no;privacy=off.</span><br style="font-family: courier new,monospace;">
<span style="font-family: courier new,monospace;">.</span><br style="font-family: courier new,monospace;"><span style="font-family: courier new,monospace;">v=0.</span><br style="font-family: courier new,monospace;"><span style="font-family: courier new,monospace;">o=- 1237220213 1237220214 IN IP4 209.244.187.176.</span><br style="font-family: courier new,monospace;">
<span style="font-family: courier new,monospace;">s=-.</span><br style="font-family: courier new,monospace;"><span style="font-family: courier new,monospace;">c=IN IP4 209.244.187.176.</span><br style="font-family: courier new,monospace;">
<span style="font-family: courier new,monospace;">t=0 0.</span><br style="font-family: courier new,monospace;"><span style="font-family: courier new,monospace;">m=audio 60458 RTP/AVP 0 18 101.</span><br style="font-family: courier new,monospace;">
<span style="font-family: courier new,monospace;">a=rtpmap:101 telephone-event/8000.</span><br style="font-family: courier new,monospace;"><br>but asterisk is reporting it like this:<br><br><div style="font-family: courier new,monospace;">
INVITE sip:+15129616808@216.82.224.202:5060;transport=udp SIP/2.0<br>Record-Route: <sip:216.82.224.202;lr;ftag=VPSF506071629460><br>Record-Route: <sip:4.79.212.229;lr;ftag=VPSF506071629460><br>Via: SIP/2.0/UDP 216.82.224.202;branch=z9hG4bK6314.4f7b5d05.0<br>
Via: SIP/2.0/UDP 4.79.212.229;branch=z9hG4bK6314.15486fb6.0<br>Via: SIP/2.0/UDP 4.68.250.148:5060;branch=z9hG4bK506071629460-1207516720501<br>From: "BANDWIDTH COM" <<a href="mailto:sip%3A%2B19192282250@4.68.250.148">sip:+19192282250@4.68.250.148</a>>;tag=VPSF506071629460<br>
To: <<a href="http://sip:+15129616808@4.79.212.229:5060">sip:+15129616808@4.79.212.229:5060</a>><br>Call-ID: <a href="mailto:HOUMGC0520090316161653037223@209.244.63.35">HOUMGC0520090316161653037223@209.244.63.35</a><br>
CSeq: 1 INVITE<br>Contact: <sip:+19192282250@4.68.250.148:5060;transport=udp><br>Max-Forwards: 67<br>Content-Type: application/sdp<br>Content-Length: 175<br>Remote-Party-ID: "BANDWIDTH COM" <<a href="mailto:sip%3A%2B19192282250@4.68.250.148">sip:+19192282250@4.68.250.148</a>>;party=calling;screen=no;privacy=off<br>
<br>v=0<br>o=- 1237220213 1237220214 IN IP4 216.82.224.202<br>s=-<br>c=IN IP4 216.82.224.202<br>t=0 0<br>m=audio 60458 RTP/AVP 0 18 101<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-15<br></div><br><font face="arial,helvetica,sans-serif">as a result, I don't get incoming audio for obvious reasons. Is there any possibility that it's my asterisk configuration? I'm having a bear of a time getting to someone useful at my ISP, so I'm hoping to find that it's my problem.</font><br>