I'm not getting inbound audio from bandwidth.com. Their engineer said the invite that they're sending me looks like this:
INVITE sip:+15129616...@67.198.16.18:5060;transport=udp SIP/2.0. Record-Route: <sip:216.82.224.202;lr;ftag=VPSF506071629460>. Record-Route: <sip:4.79.212.229;lr;ftag=VPSF506071629460>. Via: SIP/2.0/UDP 216.82.224.202;branch=z9hG4bK6314.4f7b5d05.0. Via: SIP/2.0/UDP 4.79.212.229;branch=z9hG4bK6314.15486fb6.0. Via: SIP/2.0/UDP 4.68.250.148:5060;branch=z9hG4bK506071629460-1207516720501. From: "BANDWIDTH COM" <sip:+19192282...@4.68.250.148<sip%3a%2b19192282...@4.68.250.148> >;tag=VPSF506071629460. To: <sip:+15129616...@4.79.212.229:5060>. Call-ID: houmgc0520090316161653037...@209.244.63.35. CSeq: 1 INVITE. Contact: <sip:+19192282...@4.68.250.148:5060;transport=udp>. Max-Forwards: 67. Content-Type: application/sdp. Content-Length: 177. Remote-Party-ID: "BANDWIDTH COM" <sip:+19192282...@4.68.250.148<sip%3a%2b19192282...@4.68.250.148>>;party=calling ;screen=no;privacy=off. . v=0. o=- 1237220213 1237220214 IN IP4 209.244.187.176. s=-. c=IN IP4 209.244.187.176. t=0 0. m=audio 60458 RTP/AVP 0 18 101. a=rtpmap:101 telephone-event/8000. but asterisk is reporting it like this: INVITE sip:+15129616...@216.82.224.202:5060;transport=udp SIP/2.0 Record-Route: <sip:216.82.224.202;lr;ftag=VPSF506071629460> Record-Route: <sip:4.79.212.229;lr;ftag=VPSF506071629460> Via: SIP/2.0/UDP 216.82.224.202;branch=z9hG4bK6314.4f7b5d05.0 Via: SIP/2.0/UDP 4.79.212.229;branch=z9hG4bK6314.15486fb6.0 Via: SIP/2.0/UDP 4.68.250.148:5060;branch=z9hG4bK506071629460-1207516720501 From: "BANDWIDTH COM" <sip:+19192282...@4.68.250.148<sip%3a%2b19192282...@4.68.250.148> >;tag=VPSF506071629460 To: <sip:+15129616...@4.79.212.229:5060> Call-ID: houmgc0520090316161653037...@209.244.63.35 CSeq: 1 INVITE Contact: <sip:+19192282...@4.68.250.148:5060;transport=udp> Max-Forwards: 67 Content-Type: application/sdp Content-Length: 175 Remote-Party-ID: "BANDWIDTH COM" <sip:+19192282...@4.68.250.148<sip%3a%2b19192282...@4.68.250.148> >;party=calling;screen=no;privacy=off v=0 o=- 1237220213 1237220214 IN IP4 216.82.224.202 s=- c=IN IP4 216.82.224.202 t=0 0 m=audio 60458 RTP/AVP 0 18 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 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.
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