<div dir="ltr"><div dir="ltr">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. <div><br></div><div>here is the SIP debug </div><div><--- SIP read from UDP:X.X.X.X:1024 ---><br><br><br>  == Using SIP RTP CoS mark 5<br>Audio is at 16060<br>Adding codec ulaw to SDP<br>Adding codec alaw to SDP<br>Adding codec gsm to SDP<br>Adding non-codec 0x1 (telephone-event) to SDP<br>Reliably Transmitting (NAT) to 

X.X.X.X

:1024:<br>INVITE sip:2012@

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:1024;ob SIP/2.0<br>Via: SIP/2.0/UDP 

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:5060;branch=z9hG4bK2555a6ef;rport<br>Max-Forwards: 70<br>From: "Jerry Geis 101" <sip:XXXXXXXXXX@

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>;tag=as5e61ec66<br>To: <sip:2012@

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:1024;ob><br>Contact: <sip:XXXXXXXXXX@

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:5060><br>Call-ID: 361b4b803f214946320c0af84a9ac0c4@

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:5060<br>CSeq: 102 INVITE<br>User-Agent: Asterisk PBX 13.33.0<br>Date: Fri, 12 Jun 2020 12:18:18 GMT<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE<br>Supported: replaces, timer<br>Alert-Info: Ring Answer<br>Content-Type: application/sdp<br>Content-Length: 285<br><br>v=0<br>o=root 1889524876 1889524876 IN IP4 

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<br>s=Asterisk PBX 13.33.0<br>c=IN IP4 

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<br>t=0 0<br>m=audio 16060 RTP/AVP 0 8 3 101<br>a=rtpmap:0 PCMU/8000<br>a=rtpmap:8 PCMA/8000<br>a=rtpmap:3 GSM/8000<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-16<br>a=maxptime:150<br>a=sendrecv<br><br>---<br>    -- Called 2012<br><br><--- SIP read from UDP:

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:1024 ---><br>SIP/2.0 100 Trying<br>Via: SIP/2.0/UDP 

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:5060;rport=5060;received=

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;branch=z9hG4bK2555a6ef<br>Call-ID: 361b4b803f214946320c0af84a9ac0c4@

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:5060<br>From: "Jerry Geis 101" <sip:XXXXXXXXXX@

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>;tag=as5e61ec66<br>To: <sip:2012@

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;ob><br>CSeq: 102 INVITE<br>Content-Length:  0<br><br><br><-------------><br>--- (7 headers 0 lines) ---<br><br><--- SIP read from UDP:

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:1024 ---><br>SIP/2.0 403 Forbidden<br>Via: SIP/2.0/UDP 

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:5060;rport=5060;received=

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;branch=z9hG4bK2555a6ef<br>Call-ID: 361b4b803f214946320c0af84a9ac0c4@

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:5060<br>From: "Jerry Geis 101" <sip:XXXXXXXXXX@

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>;tag=as5e61ec66<br>To: <sip:2012@

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;ob>;tag=6fK7TJdtnZb0JL.8C0aSd41SPe1goSxI<br>CSeq: 102 INVITE<br>Content-Length:  0<br><br><br><-------------><br>--- (7 headers 0 lines) ---<br>Transmitting (NAT) to 

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:1024:<br>ACK sip:2012@

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:1024;ob SIP/2.0<br>Via: SIP/2.0/UDP 

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:5060;branch=z9hG4bK2555a6ef;rport<br>Max-Forwards: 70<br>From: "Jerry Geis 101" <sip:XXXXXXXXXX@

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>;tag=as5e61ec66<br>To: <sip:2012@

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:1024;ob>;tag=6fK7TJdtnZb0JL.8C0aSd41SPe1goSxI<br>Contact: <sip:XXXXXXXXXX@

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:5060><br>Call-ID: 361b4b803f214946320c0af84a9ac0c4@

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:5060<br>CSeq: 102 ACK<br>User-Agent: Asterisk PBX 13.33.0<br>Content-Length: 0<br><br><br>---<br>[Jun 12 08:18:18] WARNING[12933]: chan_sip.c:24191 handle_response_invite: Received response: "Forbidden" from '"Jerry Geis 101" <sip:XXXXXXXXXX@

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>;tag=as5e61ec66'<br>Scheduling destruction of SIP dialog '361b4b803f214946320c0af84a9ac0c4@

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:5060' in 32000 ms (Method: INVITE)<br></div></div></div>