Carlton O'Riley
2004-Mar-05 12:20 UTC
[Asterisk-Users] RE: Asterisk-Users digest, Vol 1 #3012 - 11 msgs
I'm at a little bit of a loss here. I'm going to enclose my SIP output for this session that hopefully someone knows why I get the "SDP not available" message when using SIPPS to Asterisk. It registers great and when I call SIPPS it rings, but when it answers I get the same problem with the "SDP not available" message. Any help would be greatly appreciated. Thanks... ----------------- SIP Session messages --------------- INVITE sip:4670@192.168.3.53 SIP/2.0 Via: SIP/ 2.0/ UDP 192.168.3.69 ;branch=z9hG4bKnp492417237-4578560e192.168.3.69 From: <sip:101@192.168.3.53> ;tag=1d59b0b2 To: <sip:4670@192.168.3.53> Call-ID: 492417234-4578560b@192.168.3.69 CSeq: 1 INVITE User-Agent: Ahead SIPPS IP Phone Version 2.0.45.18 Expires: 180 Accept: application/ sdp Content-Type: application/ sdp Content-Length: 243 Contact: <sip:101@192.168.3.69> v=0 o=SIPPS 492417159 492417162 IN IP4 192.168.3.69 s=SIP call c=IN IP4 192.168.3.69 t=0 0 m=audio 10000 RTP/AVP 0 8 97 2 3 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:3 GSM/8000 SIP/2.0 407 Proxy Authentication Required Via: SIP/ 2.0/ UDP 192.168.3.69 ;branch=z9hG4bKnp492417237-4578560e192.168.3.69 From: <sip:101@192.168.3.53> ;tag=1d59b0b2 To: <sip:4670@192.168.3.53> ;tag=as6b4bb6b5 Call-ID: 492417234-4578560b@192.168.3.69 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE Allow: ACK Allow: CANCEL Allow: OPTIONS Allow: BYE Allow: REFER Contact: <sip:4670@192.168.3.53> Proxy-Authenticate: Digest realm="asterisk",nonce="6c08deed" Content-Length: 0 INVITE sip:4670@192.168.3.53 SIP/2.0 Via: SIP/ 2.0/ UDP 192.168.3.69 ;branch=z9hG4bKnp492418173-4587980e192.168.3.69 From: <sip:101@192.168.3.53> ;tag=1d59b0b2 To: <sip:4670@192.168.3.53> Call-ID: 492417234-4578560b@192.168.3.69 CSeq: 2 INVITE Proxy-Authorization: Digest username="101",realm="asterisk",uri="sip:192.168.3.69",nonce="6c08deed",nc=" 00000001",response="f058d443e9cebe0b23ca1adf0db21afb" Content-Type: application/ sdp Content-Length: 243 Date: Fri, 05 Mar 2004 19:16:01 GMT Contact: <sip:101@192.168.3.69> Expires: 180 Accept: application/ sdp User-Agent: Ahead SIPPS IP Phone Version 2.0.45.18 v=0 o=SIPPS 492417159 492417162 IN IP4 192.168.3.69 s=SIP call c=IN IP4 192.168.3.69 t=0 0 m=audio 10000 RTP/AVP 0 8 97 2 3 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:3 GSM/8000 SIP/2.0 100 Trying Via: SIP/ 2.0/ UDP 192.168.3.69 ;branch=z9hG4bKnp492418173-4587980e192.168.3.69 From: <sip:101@192.168.3.53> ;tag=1d59b0b2 To: <sip:4670@192.168.3.53> ;tag=as176f35b3 Call-ID: 492417234-4578560b@192.168.3.69 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE Allow: ACK Allow: CANCEL Allow: OPTIONS Allow: BYE Allow: REFER Contact: <sip:4670@192.168.3.53> Content-Length: 0 SIP/2.0 200 OK Via: SIP/ 2.0/ UDP 192.168.3.69 ;branch=z9hG4bKnp492418173-4587980e192.168.3.69 From: <sip:101@192.168.3.53> ;tag=1d59b0b2 To: <sip:4670@192.168.3.53> ;tag=as176f35b3 Call-ID: 492417234-4578560b@192.168.3.69 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE Allow: ACK Allow: CANCEL Allow: OPTIONS Allow: BYE Allow: REFER Contact: <sip:4670@192.168.3.53> Content-Type: application/ sdp Content-Length: 235 v=0 o=root 32181 32181 IN IP4 192.168.3.53 s=session c=IN IP4 192.168.3.53 t=0 0 m=audio 16246 RTP/AVP 0 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 m=video 11206 RTP/AVP CANCEL sip:4670@192.168.3.53 SIP/2.0 Via: SIP/ 2.0/ UDP 192.168.3.69 ;branch=z9hG4bKnp492418173-4587980e192.168.3.69 From: <sip:101@192.168.3.53> ;tag=1d59b0b2 To: <sip:4670@192.168.3.53> ;tag=as176f35b3 Call-ID: 492417234-4578560b@192.168.3.69 CSeq: 2 CANCEL Warning: 399 "SDP not available" Date: Fri, 05 Mar 2004 19:16:01 GMT Content-Length: 0 User-Agent: Ahead SIPPS IP Phone Version 2.0.45.18 SIP/2.0 200 OK Via: SIP/ 2.0/ UDP 192.168.3.69 ;branch=z9hG4bKnp492418173-4587980e192.168.3.69 From: <sip:101@192.168.3.53> ;tag=1d59b0b2 To: <sip:4670@192.168.3.53> ;tag=as176f35b3 Call-ID: 492417234-4578560b@192.168.3.69 CSeq: 2 CANCEL User-Agent: Asterisk PBX Allow: INVITE Allow: ACK Allow: CANCEL Allow: OPTIONS Allow: BYE Allow: REFER Contact: <sip:4670@192.168.3.53> Content-Length: 0 SIP/2.0 200 OK Via: SIP/ 2.0/ UDP 192.168.3.69 ;branch=z9hG4bKnp492418173-4587980e192.168.3.69 From: <sip:101@192.168.3.53> ;tag=1d59b0b2 To: <sip:4670@192.168.3.53> ;tag=as176f35b3 Call-ID: 492417234-4578560b@192.168.3.69 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE Allow: ACK Allow: CANCEL Allow: OPTIONS Allow: BYE Allow: REFER Contact: <sip:4670@192.168.3.53> Content-Type: application/ sdp Content-Length: 235 v=0 o=root 32181 32181 IN IP4 192.168.3.53 s=session c=IN IP4 192.168.3.53 t=0 0 m=audio 16246 RTP/AVP 0 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 m=video 11206 RTP/AVP SIP/2.0 200 OK Via: SIP/ 2.0/ UDP 192.168.3.69 ;branch=z9hG4bKnp492418173-4587980e192.168.3.69 From: <sip:101@192.168.3.53> ;tag=1d59b0b2 To: <sip:4670@192.168.3.53> ;tag=as176f35b3 Call-ID: 492417234-4578560b@192.168.3.69 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE Allow: ACK Allow: CANCEL Allow: OPTIONS Allow: BYE Allow: REFER Contact: <sip:4670@192.168.3.53> Content-Type: application/ sdp Content-Length: 235 v=0 o=root 32181 32181 IN IP4 192.168.3.53 s=session c=IN IP4 192.168.3.53 t=0 0 m=audio 16246 RTP/AVP 0 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 m=video 11206 RTP/AVP SIP/2.0 200 OK Via: SIP/ 2.0/ UDP 192.168.3.69 ;branch=z9hG4bKnp492418173-4587980e192.168.3.69 From: <sip:101@192.168.3.53> ;tag=1d59b0b2 To: <sip:4670@192.168.3.53> ;tag=as176f35b3 Call-ID: 492417234-4578560b@192.168.3.69 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE Allow: ACK Allow: CANCEL Allow: OPTIONS Allow: BYE Allow: REFER Contact: <sip:4670@192.168.3.53> Content-Type: application/ sdp Content-Length: 235 v=0 o=root 32181 32181 IN IP4 192.168.3.53 s=session c=IN IP4 192.168.3.53 t=0 0 m=audio 16246 RTP/AVP 0 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 m=video 11206 RTP/AVP SIP/2.0 200 OK Via: SIP/ 2.0/ UDP 192.168.3.69 ;branch=z9hG4bKnp492418173-4587980e192.168.3.69 From: <sip:101@192.168.3.53> ;tag=1d59b0b2 To: <sip:4670@192.168.3.53> ;tag=as176f35b3 Call-ID: 492417234-4578560b@192.168.3.69 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE Allow: ACK Allow: CANCEL Allow: OPTIONS Allow: BYE Allow: REFER Contact: <sip:4670@192.168.3.53> Content-Type: application/ sdp Content-Length: 235 v=0 o=root 32181 32181 IN IP4 192.168.3.53 s=session c=IN IP4 192.168.3.53 t=0 0 m=audio 16246 RTP/AVP 0 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 m=video 11206 RTP/AVP SIP/2.0 200 OK Via: SIP/ 2.0/ UDP 192.168.3.69 ;branch=z9hG4bKnp492418173-4587980e192.168.3.69 From: <sip:101@192.168.3.53> ;tag=1d59b0b2 To: <sip:4670@192.168.3.53> ;tag=as176f35b3 Call-ID: 492417234-4578560b@192.168.3.69 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE Allow: ACK Allow: CANCEL Allow: OPTIONS Allow: BYE Allow: REFER Contact: <sip:4670@192.168.3.53> Content-Type: application/ sdp Content-Length: 235 v=0 o=root 32181 32181 IN IP4 192.168.3.53 s=session c=IN IP4 192.168.3.53 t=0 0 m=audio 16246 RTP/AVP 0 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 m=video 11206 RTP/AVP