Hi Steve, - Your right - the file was AMI (copied the other one). By direct connect I simply meant the speaker is an extension on that server.
here is the SIP debug <--- SIP read from UDP:X.X.X.X:1024 ---> == Using SIP RTP CoS mark 5 Audio is at 16060 Adding codec ulaw to SDP Adding codec alaw to SDP Adding codec gsm to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to X.X.X.X :1024: INVITE sip:2012@ X.X.X.X :1024;ob SIP/2.0 Via: SIP/2.0/UDP X.X.X.X :5060;branch=z9hG4bK2555a6ef;rport Max-Forwards: 70 From: "Jerry Geis 101" <sip:XXXXXXXXXX@ X.X.X.X >;tag=as5e61ec66 To: <sip:2012@ X.X.X.X :1024;ob> Contact: <sip:XXXXXXXXXX@ X.X.X.X :5060> Call-ID: 361b4b803f214946320c0af84a9ac0c4@ X.X.X.X :5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 13.33.0 Date: Fri, 12 Jun 2020 12:18:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Alert-Info: Ring Answer Content-Type: application/sdp Content-Length: 285 v=0 o=root 1889524876 1889524876 IN IP4 X.X.X.X s=Asterisk PBX 13.33.0 c=IN IP4 X.X.X.X t=0 0 m=audio 16060 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- -- Called 2012 <--- SIP read from UDP: X.X.X.X :1024 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP X.X.X.X :5060;rport=5060;received= X.X.X.X ;branch=z9hG4bK2555a6ef Call-ID: 361b4b803f214946320c0af84a9ac0c4@ X.X.X.X :5060 From: "Jerry Geis 101" <sip:XXXXXXXXXX@ X.X.X.X >;tag=as5e61ec66 To: <sip:2012@ X.X.X.X ;ob> CSeq: 102 INVITE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- <--- SIP read from UDP: X.X.X.X :1024 ---> SIP/2.0 403 Forbidden Via: SIP/2.0/UDP X.X.X.X :5060;rport=5060;received= X.X.X.X ;branch=z9hG4bK2555a6ef Call-ID: 361b4b803f214946320c0af84a9ac0c4@ X.X.X.X :5060 From: "Jerry Geis 101" <sip:XXXXXXXXXX@ X.X.X.X >;tag=as5e61ec66 To: <sip:2012@ X.X.X.X ;ob>;tag=6fK7TJdtnZb0JL.8C0aSd41SPe1goSxI CSeq: 102 INVITE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- Transmitting (NAT) to X.X.X.X :1024: ACK sip:2012@ X.X.X.X :1024;ob SIP/2.0 Via: SIP/2.0/UDP X.X.X.X :5060;branch=z9hG4bK2555a6ef;rport Max-Forwards: 70 From: "Jerry Geis 101" <sip:XXXXXXXXXX@ X.X.X.X >;tag=as5e61ec66 To: <sip:2012@ X.X.X.X :1024;ob>;tag=6fK7TJdtnZb0JL.8C0aSd41SPe1goSxI Contact: <sip:XXXXXXXXXX@ X.X.X.X :5060> Call-ID: 361b4b803f214946320c0af84a9ac0c4@ X.X.X.X :5060 CSeq: 102 ACK User-Agent: Asterisk PBX 13.33.0 Content-Length: 0 --- [Jun 12 08:18:18] WARNING[12933]: chan_sip.c:24191 handle_response_invite: Received response: "Forbidden" from '"Jerry Geis 101" <sip:XXXXXXXXXX@ X.X.X.X >;tag=as5e61ec66' Scheduling destruction of SIP dialog '361b4b803f214946320c0af84a9ac0c4@ X.X.X.X :5060' in 32000 ms (Method: INVITE)
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