<div dir="ltr"><br><div class="gmail_extra"><br><div class="gmail_quote">On Mon, Sep 8, 2014 at 10:19 AM, Olli Heiskanen <span dir="ltr"><<a href="mailto:ohjelmistoarkkitehti@gmail.com" target="_blank">ohjelmistoarkkitehti@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div dir="ltr">Hi Matthew,<div><br></div><div>Here's the debug output: </div><div><br></div><div><br></div><div><br></div><div><div><br></div><div><br></div><div><--- SIP read from UDP:PU.BL.IC.IP:5060 ---></div><div>INVITE <a href="mailto:sip%3A661@testers.com" target="_blank">sip:661@testers.com</a> SIP/2.0</div><div>Record-Route: <sip:PU.BL.IC.IP;r2=on;lr=on;ftag=856i7ei98p;nat=yes></div><div>Record-Route: <sip:PU.BL.IC.IP;transport=ws;r2=on;lr=on;ftag=856i7ei98p;nat=yes></div><div>Via: SIP/2.0/UDP PU.BL.IC.IP;branch=z9hG4bK3cb9.e83bde7c484eec5374f679e7232435af.0</div><div>Via: SIP/2.0/WS 8d833dsg4asm.invalid;rport=47184;received=CL.IE.NT.IP;branch=z9hG4bK3627044</div><div>Max-Forwards: 69</div><div>To: <<a href="mailto:sip%3A661@testers.com" target="_blank">sip:661@testers.com</a>></div><div>From: "660" <<a href="mailto:sip%3A660@testers.com" target="_blank">sip:660@testers.com</a>>;tag=856i7ei98p</div><div>Call-ID: oc0ppijresm05k2emsgt</div><div>CSeq: 3394 INVITE</div><div>Contact: <<a href="mailto:sip%3A660@testers.com" target="_blank">sip:660@testers.com</a>;gr=urn:uuid:81780308-9304-4e37-984f-a2e864b17bd3;alias=CL.IE.NT.IP~47184~5;alias=CL.IE.NT.IP~47184~5></div><div>Allow: ACK,CANCEL,BYE,OPTIONS,INFO,NOTIFY,INVITE,MESSAGE</div><div>Content-Type: application/sdp</div><div>Supported: gruu,outbound</div><div>User-Agent: SIP.js/0.6.2</div><div>Content-Length: 1862</div><div><br></div><div>v=0</div><div>o=- 9082254022026432015 2 IN IP4 PU.BL.IC.IP</div><div>s=-</div><div>t=0 0</div><div>a=group:BUNDLE audio</div><div>a=msid-semantic: WMS JtsXs3B8wXmgyUYVkUGWHBYVYgX0aSC38IWx</div><div>m=audio 10862 RTP/SAVPF 111 103 104 0 8 106 105 13 126</div><div>c=IN IP4 PU.BL.IC.IP</div><div>a=candidate:3350409123 1 udp 2122194687 192.168.0.101 65339 typ host generation 0</div><div>a=candidate:3350409123 2 udp 2122194687 192.168.0.101 65339 typ host generation 0</div><div>a=candidate:2301678419 1 tcp 1518214911 192.168.0.101 0 typ host generation 0</div><div>a=candidate:2301678419 2 tcp 1518214911 192.168.0.101 0 typ host generation 0</div><div>a=candidate:1190865175 1 udp 1685987071 CL.IE.NT.IP 65339 typ srflx raddr 192.168.0.101 rport 65339 generation 0</div><div>a=candidate:1190865175 2 udp 1685987071 CL.IE.NT.IP 65339 typ srflx raddr 192.168.0.101 rport 65339 generation 0</div><div>a=ice-ufrag:7N23UxBo9XUgx9pJ</div><div>a=ice-pwd:jL7AIeiJD5byGDSapfSftPRl</div><div>a=ice-options:google-ice</div><div>a=fingerprint:sha-256 93:E7:FC:E6:C2:74:71:2F:4F:81:43:7D:0C:A1:0F:C9:FC:3B:85:E6:44:2F:5A:39:05:79:DD:A6:0B:05:49:80</div><div>a=setup:actpass</div><div>a=mid:audio</div><div>a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level</div><div>a=extmap:3 <a href="http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time" target="_blank">http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time</a></div><div>a=rtpmap:111 opus/48000/2</div><div>a=fmtp:111 minptime=10</div><div>a=rtpmap:103 ISAC/16000</div><div>a=rtpmap:104 ISAC/32000</div><div>a=rtpmap:0 PCMU/8000</div><div>a=rtpmap:8 PCMA/8000</div><div>a=rtpmap:106 CN/32000</div><div>a=rtpmap:105 CN/16000</div><div>a=rtpmap:13 CN/8000</div><div>a=rtpmap:126 telephone-event/8000</div><div>a=maxptime:60</div><div>a=ssrc:1842567493 cname:aJCyVX5iKPNU6Gf8</div><div>a=ssrc:1842567493 msid:JtsXs3B8wXmgyUYVkUGWHBYVYgX0aSC38IWx 01a46fec-8a85-412d-9905-dcbefb8952b6</div><div>a=ssrc:1842567493 mslabel:JtsXs3B8wXmgyUYVkUGWHBYVYgX0aSC38IWx</div><div>a=ssrc:1842567493 label:01a46fec-8a85-412d-9905-dcbefb8952b6</div><div>a=sendrecv</div><div>a=rtcp:10863</div><div>a=rtcp-mux</div><div>a=candidate:GjpvUWJxlvHL7PZ6 1 UDP 1518214655 PU.BL.IC.IP 10862 typ host</div><div>a=candidate:GjpvUWJxlvHL7PZ6 2 UDP 1518214654 PU.BL.IC.IP 10863 typ host</div><div><-------------></div><div>--- (16 headers 42 lines) ---</div><div>Sending to PU.BL.IC.IP:5060 (no NAT)</div><div>Sending to PU.BL.IC.IP:5060 (no NAT)</div><div>Using INVITE request as basis request - oc0ppijresm05k2emsgt</div><div>Found peer '660' for '660' from PU.BL.IC.IP:5060</div><div> == Using SIP RTP TOS bits 184</div><div> == Using SIP RTP CoS mark 5</div><div>Found RTP audio format 111</div><div>Found RTP audio format 103</div><div>Found RTP audio format 104</div><div>Found RTP audio format 0</div><div>Found RTP audio format 8</div><div>Found RTP audio format 106</div><div>Found RTP audio format 105</div><div>Found RTP audio format 13</div><div>Found RTP audio format 126</div><div>Found unknown media description format opus for ID 111</div><div>Found unknown media description format ISAC for ID 103</div><div>Found unknown media description format ISAC for ID 104</div><div>Found audio description format PCMU for ID 0</div><div>Found audio description format PCMA for ID 8</div><div>Found unknown media description format CN for ID 106</div><div>Found unknown media description format CN for ID 105</div><div>Found audio description format CN for ID 13</div><div>Found audio description format telephone-event for ID 126</div><div>Capabilities: us - (gsm|ulaw|alaw|h263|testlaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)</div><div>Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x3 (telephone-event|CN|), combined - 0x1 (telephone-event|)</div><div>Peer audio RTP is at port PU.BL.IC.IP:10862</div><div>Looking for 661 in default (domain <a href="http://testers.com" target="_blank">testers.com</a>)</div><div>list_route: hop: <sip:PU.BL.IC.IP;r2=on;lr=on;ftag=856i7ei98p;nat=yes></div><div>list_route: hop: <sip:PU.BL.IC.IP;transport=ws;r2=on;lr=on;ftag=856i7ei98p;nat=yes></div><div><br></div><div><--- Transmitting (NAT) to PU.BL.IC.IP:5060 ---></div><div>SIP/2.0 100 Trying</div><div>Via: SIP/2.0/UDP PU.BL.IC.IP;branch=z9hG4bK3cb9.e83bde7c484eec5374f679e7232435af.0;received=PU.BL.IC.IP;rport=5060</div><div>Via: SIP/2.0/WS 8d833dsg4asm.invalid;rport=47184;received=CL.IE.NT.IP;branch=z9hG4bK3627044</div><div>Record-Route: <sip:PU.BL.IC.IP;r2=on;lr=on;ftag=856i7ei98p;nat=yes></div><div>Record-Route: <sip:PU.BL.IC.IP;transport=ws;r2=on;lr=on;ftag=856i7ei98p;nat=yes></div><div>From: "660" <<a href="mailto:sip%3A660@testers.com" target="_blank">sip:660@testers.com</a>>;tag=856i7ei98p</div><div>To: <<a href="mailto:sip%3A661@testers.com" target="_blank">sip:661@testers.com</a>></div><div>Call-ID: oc0ppijresm05k2emsgt</div><div>CSeq: 3394 INVITE</div><div>Server: I Am the Devil</div><div>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE</div><div>Supported: replaces, timer</div><div>Contact: <sip:661@PU.BL.IC.IP:5070></div><div>Content-Length: 0</div><div><br></div><div><br></div><div><------------></div><div> -- Executing [661@default:1] NoOp("SIP/660-00000007", "general : Dialed 661") in new stack</div><div> -- Executing [661@default:2] Dial("SIP/660-00000007", "SIP/661,3600,rt") in new stack</div><div> == Using SIP RTP TOS bits 184</div><div> == Using SIP RTP CoS mark 5</div><div>Audio is at 18366</div><div>Adding codec 100003 (ulaw) to SDP</div><div>Adding codec 100002 (gsm) to SDP</div><div>Adding codec 100004 (alaw) to SDP</div><div>Adding codec 100017 (testlaw) to SDP</div><div>Adding non-codec 0x1 (telephone-event) to SDP</div><div>Reliably Transmitting (NAT) to PU.BL.IC.IP:5060:</div><div>INVITE sip:661@PU.BL.IC.IP:5060 SIP/2.0</div><div>Via: SIP/2.0/UDP PU.BL.IC.IP:5070;branch=z9hG4bK4e738afa;rport</div><div>Max-Forwards: 70</div><div>From: "660 win8" <<a href="mailto:sip%3A660@testers.com" target="_blank">sip:660@testers.com</a>>;tag=as73376885</div><div>To: <sip:661@PU.BL.IC.IP:5060></div><div>Contact: <sip:660@PU.BL.IC.IP:5070></div><div>Call-ID: <a href="mailto:2f70cc9567be50a46ba2879d4391a7dc@testers.com" target="_blank">2f70cc9567be50a46ba2879d4391a7dc@testers.com</a></div><div>CSeq: 102 INVITE</div><div>User-Agent: I Am the Devil</div><div>Date: Mon, 08 Sep 2014 15:15:37 GMT</div><div>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE</div><div>Supported: replaces, timer</div><div>Content-Type: application/sdp</div><div>Content-Length: 437</div><div><br></div><div>v=0</div><div>o=root 630896079 630896079 IN IP4 PU.BL.IC.IP</div><div>s=Asterisk PBX 11.11.0</div><div>c=IN IP4 PU.BL.IC.IP</div><div>t=0 0</div><div>m=audio 18366 RTP/SAVPF 0 3 8 101</div><div>a=rtpmap:0 PCMU/8000</div><div>a=rtpmap:3 GSM/8000</div><div>a=rtpmap:8 PCMA/8000</div><div>a=rtpmap:101 telephone-event/8000</div><div>a=fmtp:101 0-16</div><div>a=ptime:20</div><div>a=connection:new</div><div>a=setup:actpass</div><div>a=fingerprint:SHA-256 CE:EE:D9:28:EA:B0:6E:D0:CE:4F:5A:9A:FB:53:66:74:83:47:18:37:2F:76:C1:6D:10:C0:EE:FF:A4:56:F4:05</div><div>a=sendrecv</div><div><br></div><div>---</div><br clear="all"></div></div></blockquote></div><br></div><div class="gmail_extra">That's not really DEBUG output - just VERBOSE output from the CLI with 'sip set debug on'.<br><br></div><div class="gmail_extra">That aside, your initial e-mail provided the configuration for SIP peer 660, but not for SIP peer 661. In your dialplan, you are dialling SIP peer 661:<br><br><div> -- Executing [661@default:2] Dial("SIP/660-00000007", "SIP/661,3600,rt") in new stack</div><div> == Using SIP RTP TOS bits 184</div><div> == Using SIP RTP CoS mark 5<br><br></div><div>What is their configuration?<br></div><br></div><div class="gmail_extra"><br>-- <br><div dir="ltr"><div>Matthew Jordan<br></div><div>Digium, Inc. | Engineering Manager</div><div>445 Jan Davis Drive NW - Huntsville, AL 35806 - USA</div><div>Check us out at: <a href="http://digium.com" target="_blank">http://digium.com</a> & <a href="http://asterisk.org" target="_blank">http://asterisk.org</a></div></div>
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