Re: [asterisk-users] Forbidden call
On Fri, 12 Jun 2020, Jerry Geis wrote: Any chance you can configure the speaker to syslog to your host so you may get a clue why the speaker is rejecting? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Forbidden call
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" ;tag=as5e61ec66 To: Contact: 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" ;tag=as5e61ec66 To: 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" ;tag=as5e61ec66 To: ;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" ;tag=as5e61ec66 To: ;tag=6fK7TJdtnZb0JL.8C0aSd41SPe1goSxI Contact: 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" ;tag=as5e61ec66' Scheduling destruction of SIP dialog '361b4b803f214946320c0af84a9ac0c4@ X.X.X.X :5060' in 32000 ms (Method: INVITE) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Forbidden call
On Thu, 11 Jun 2020, Jerry Geis wrote: I have a call from a call file: This looks a lot more like an AMI event than a call file. In any case, it doesn't matter. Action: Originate Async: yes Channel: SIP/2012 Codecs: ulaw,alaw,gsm Context: dialout Exten: callprogress Priority: 1 Timeout: 2 Variable: SIPADDHEADER="Alert-Info: Ring Answer" ActionID: 100014 CallerID: Axis < 525 > The SIP/2012 is a IP Speaker on the computer. The error is: [Jun 11 15:44:45] WARNING[8132]: chan_sip.c:24191 handle_response_invite: Received response: "Forbidden" Why am I getting "Forbidden" ? Its a call file on my server It's not a call file permissions thing. That would be a different error and reported by something before chan_sip. the speaker is directly connected to my server. How is an IP speaker 'directly connected?' Do you mean directly from the Ethernet on the speaker to a NIC on the computer? It doesn't matter, just curious :) The only thing that will tell you what is going on is the packets. Crank up 'sip set debug on' and see if that yields a clue. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Forbidden call
I have a call from a call file: Action: Originate Async: yes Channel: SIP/2012 Codecs: ulaw,alaw,gsm Context: dialout Exten: callprogress Priority: 1 Timeout: 2 Variable: SIPADDHEADER="Alert-Info: Ring Answer" ActionID: 100014 CallerID: Axis < 525 > The SIP/2012 is a IP Speaker on the computer. The error is: [Jun 11 15:44:45] WARNING[8132]: chan_sip.c:24191 handle_response_invite: Received response: "Forbidden" Why am I getting "Forbidden" ? Its a call file on my server and the speaker is directly connected to my server. Thanks Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
