On Mon, Sep 8, 2014 at 10:19 AM, Olli Heiskanen < [email protected]> wrote:
> Hi Matthew, > > Here's the debug output: > > > > > > <--- SIP read from UDP:PU.BL.IC.IP:5060 ---> > INVITE sip:[email protected] SIP/2.0 > Record-Route: <sip:PU.BL.IC.IP;r2=on;lr=on;ftag=856i7ei98p;nat=yes> > Record-Route: > <sip:PU.BL.IC.IP;transport=ws;r2=on;lr=on;ftag=856i7ei98p;nat=yes> > Via: SIP/2.0/UDP > PU.BL.IC.IP;branch=z9hG4bK3cb9.e83bde7c484eec5374f679e7232435af.0 > Via: SIP/2.0/WS > 8d833dsg4asm.invalid;rport=47184;received=CL.IE.NT.IP;branch=z9hG4bK3627044 > Max-Forwards: 69 > To: <sip:[email protected]> > From: "660" <sip:[email protected]>;tag=856i7ei98p > Call-ID: oc0ppijresm05k2emsgt > CSeq: 3394 INVITE > Contact: <sip:[email protected] > ;gr=urn:uuid:81780308-9304-4e37-984f-a2e864b17bd3;alias=CL.IE.NT.IP~47184~5;alias=CL.IE.NT.IP~47184~5> > Allow: ACK,CANCEL,BYE,OPTIONS,INFO,NOTIFY,INVITE,MESSAGE > Content-Type: application/sdp > Supported: gruu,outbound > User-Agent: SIP.js/0.6.2 > Content-Length: 1862 > > v=0 > o=- 9082254022026432015 2 IN IP4 PU.BL.IC.IP > s=- > t=0 0 > a=group:BUNDLE audio > a=msid-semantic: WMS JtsXs3B8wXmgyUYVkUGWHBYVYgX0aSC38IWx > m=audio 10862 RTP/SAVPF 111 103 104 0 8 106 105 13 126 > c=IN IP4 PU.BL.IC.IP > a=candidate:3350409123 1 udp 2122194687 192.168.0.101 65339 typ host > generation 0 > a=candidate:3350409123 2 udp 2122194687 192.168.0.101 65339 typ host > generation 0 > a=candidate:2301678419 1 tcp 1518214911 192.168.0.101 0 typ host > generation 0 > a=candidate:2301678419 2 tcp 1518214911 192.168.0.101 0 typ host > generation 0 > a=candidate:1190865175 1 udp 1685987071 CL.IE.NT.IP 65339 typ srflx raddr > 192.168.0.101 rport 65339 generation 0 > a=candidate:1190865175 2 udp 1685987071 CL.IE.NT.IP 65339 typ srflx raddr > 192.168.0.101 rport 65339 generation 0 > a=ice-ufrag:7N23UxBo9XUgx9pJ > a=ice-pwd:jL7AIeiJD5byGDSapfSftPRl > a=ice-options:google-ice > 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 > a=setup:actpass > a=mid:audio > a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level > a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time > a=rtpmap:111 opus/48000/2 > a=fmtp:111 minptime=10 > a=rtpmap:103 ISAC/16000 > a=rtpmap:104 ISAC/32000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:106 CN/32000 > a=rtpmap:105 CN/16000 > a=rtpmap:13 CN/8000 > a=rtpmap:126 telephone-event/8000 > a=maxptime:60 > a=ssrc:1842567493 cname:aJCyVX5iKPNU6Gf8 > a=ssrc:1842567493 msid:JtsXs3B8wXmgyUYVkUGWHBYVYgX0aSC38IWx > 01a46fec-8a85-412d-9905-dcbefb8952b6 > a=ssrc:1842567493 mslabel:JtsXs3B8wXmgyUYVkUGWHBYVYgX0aSC38IWx > a=ssrc:1842567493 label:01a46fec-8a85-412d-9905-dcbefb8952b6 > a=sendrecv > a=rtcp:10863 > a=rtcp-mux > a=candidate:GjpvUWJxlvHL7PZ6 1 UDP 1518214655 PU.BL.IC.IP 10862 typ host > a=candidate:GjpvUWJxlvHL7PZ6 2 UDP 1518214654 PU.BL.IC.IP 10863 typ host > <-------------> > --- (16 headers 42 lines) --- > Sending to PU.BL.IC.IP:5060 (no NAT) > Sending to PU.BL.IC.IP:5060 (no NAT) > Using INVITE request as basis request - oc0ppijresm05k2emsgt > Found peer '660' for '660' from PU.BL.IC.IP:5060 > == Using SIP RTP TOS bits 184 > == Using SIP RTP CoS mark 5 > Found RTP audio format 111 > Found RTP audio format 103 > Found RTP audio format 104 > Found RTP audio format 0 > Found RTP audio format 8 > Found RTP audio format 106 > Found RTP audio format 105 > Found RTP audio format 13 > Found RTP audio format 126 > Found unknown media description format opus for ID 111 > Found unknown media description format ISAC for ID 103 > Found unknown media description format ISAC for ID 104 > Found audio description format PCMU for ID 0 > Found audio description format PCMA for ID 8 > Found unknown media description format CN for ID 106 > Found unknown media description format CN for ID 105 > Found audio description format CN for ID 13 > Found audio description format telephone-event for ID 126 > Capabilities: us - (gsm|ulaw|alaw|h263|testlaw), peer - > audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw) > Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x3 > (telephone-event|CN|), combined - 0x1 (telephone-event|) > Peer audio RTP is at port PU.BL.IC.IP:10862 > Looking for 661 in default (domain testers.com) > list_route: hop: <sip:PU.BL.IC.IP;r2=on;lr=on;ftag=856i7ei98p;nat=yes> > list_route: hop: > <sip:PU.BL.IC.IP;transport=ws;r2=on;lr=on;ftag=856i7ei98p;nat=yes> > > <--- Transmitting (NAT) to PU.BL.IC.IP:5060 ---> > SIP/2.0 100 Trying > Via: SIP/2.0/UDP > PU.BL.IC.IP;branch=z9hG4bK3cb9.e83bde7c484eec5374f679e7232435af.0;received=PU.BL.IC.IP;rport=5060 > Via: SIP/2.0/WS > 8d833dsg4asm.invalid;rport=47184;received=CL.IE.NT.IP;branch=z9hG4bK3627044 > Record-Route: <sip:PU.BL.IC.IP;r2=on;lr=on;ftag=856i7ei98p;nat=yes> > Record-Route: > <sip:PU.BL.IC.IP;transport=ws;r2=on;lr=on;ftag=856i7ei98p;nat=yes> > From: "660" <sip:[email protected]>;tag=856i7ei98p > To: <sip:[email protected]> > Call-ID: oc0ppijresm05k2emsgt > CSeq: 3394 INVITE > Server: I Am the Devil > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, > PUBLISH, MESSAGE > Supported: replaces, timer > Contact: <sip:[email protected]:5070> > Content-Length: 0 > > > <------------> > -- Executing [661@default:1] NoOp("SIP/660-00000007", "general : > Dialed 661") in new stack > -- Executing [661@default:2] Dial("SIP/660-00000007", > "SIP/661,3600,rt") in new stack > == Using SIP RTP TOS bits 184 > == Using SIP RTP CoS mark 5 > Audio is at 18366 > Adding codec 100003 (ulaw) to SDP > Adding codec 100002 (gsm) to SDP > Adding codec 100004 (alaw) to SDP > Adding codec 100017 (testlaw) to SDP > Adding non-codec 0x1 (telephone-event) to SDP > Reliably Transmitting (NAT) to PU.BL.IC.IP:5060: > INVITE sip:[email protected]:5060 SIP/2.0 > Via: SIP/2.0/UDP PU.BL.IC.IP:5070;branch=z9hG4bK4e738afa;rport > Max-Forwards: 70 > From: "660 win8" <sip:[email protected]>;tag=as73376885 > To: <sip:[email protected]:5060> > Contact: <sip:[email protected]:5070> > Call-ID: [email protected] > CSeq: 102 INVITE > User-Agent: I Am the Devil > Date: Mon, 08 Sep 2014 15:15:37 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, > PUBLISH, MESSAGE > Supported: replaces, timer > Content-Type: application/sdp > Content-Length: 437 > > v=0 > o=root 630896079 630896079 IN IP4 PU.BL.IC.IP > s=Asterisk PBX 11.11.0 > c=IN IP4 PU.BL.IC.IP > t=0 0 > m=audio 18366 RTP/SAVPF 0 3 8 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=ptime:20 > a=connection:new > a=setup:actpass > 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 > a=sendrecv > > --- > > That's not really DEBUG output - just VERBOSE output from the CLI with 'sip set debug on'. 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: -- Executing [661@default:2] Dial("SIP/660-00000007", "SIP/661,3600,rt") in new stack == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 What is their configuration? -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com & http://asterisk.org
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