Hi, My bad, below is the output for peer 661 and the log output, which does not tell me much. I guess Asterisk assumes it's working correctly as there are no errors etc, however for some reason the INVITE leaves Asterisk without the ice definitions in the sdp. I must assume it's a configuration issue as I've been modifying the sip peer table recently, and installed Asterisk 12, then moved back to 11. I compiled the current 11.11.0 version from source, starting from ./configure --with-crypto --with-ssl --with-srtp, make menuselect etc.
*CLI> realtime load sippeers name 661 Column Name Column Value -------------------- -------------------- id 6 type friend name 661 host dynamic secret encryption yes avpf yes icesupport yes ipaddr PU.BL.IC.IP port 5060 regseconds 1410190721 defaultuser 661 fullcontact sip:6...@pu.bl.ic.ip:5060 lastms 0 useragent context default directmedia no deny 0.0.0.0/0.0.0.0 permit PU.BL.IC.IP nat force_rport,comedia language disallow allow force_avp yes callerid amaflags mailbox regexten regserver fromdomain testers.com videosupport no contactpermit contactdeny fullname 661 win8 minipc hasvoicemail subscribemwi dtlsenable yes dtlsverify no dtlscertfile /etc/asterisk/keys/asterisk.pem dtlsprivatekey /etc/asterisk/keys/asterisk.pem dtlssetup actpass sippasswd f825851fe6805899cb141bd469457829 rpid domain testers.com sippasswd2 [Sep 8 21:18:12] VERBOSE[9315][C-00000002] netsock2.c: == Using SIP RTP TOS bits 184 [Sep 8 21:18:12] VERBOSE[9315][C-00000002] netsock2.c: == Using SIP RTP CoS mark 5 [Sep 8 21:18:12] VERBOSE[9406][C-00000002] pbx.c: -- Executing [661@default:1] NoOp("SIP/660-00000002", "general : Dialed 661") in new stack [Sep 8 21:18:12] VERBOSE[9406][C-00000002] pbx.c: -- Executing [661@default:2] Dial("SIP/660-00000002", "SIP/661,3600,rt") in new stack [Sep 8 21:18:12] VERBOSE[9406][C-00000002] netsock2.c: == Using SIP RTP TOS bits 184 [Sep 8 21:18:12] VERBOSE[9406][C-00000002] netsock2.c: == Using SIP RTP CoS mark 5 [Sep 8 21:18:12] VERBOSE[9406][C-00000002] app_dial.c: -- Called SIP/661 [Sep 8 21:18:13] VERBOSE[9406][C-00000002] app_dial.c: == Everyone is busy/congested at this time (1:0/0/1) [Sep 8 21:18:13] VERBOSE[9406][C-00000002] pbx.c: -- Executing [661@default:3] Hangup("SIP/660-00000002", "") in new stack [Sep 8 21:18:13] VERBOSE[9406][C-00000002] pbx.c: == Spawn extension (default, 661, 3) exited non-zero on 'SIP/660-00000002' Here is the log output of my 661 sip.js client: There the client receives an INVITE without ice data in the sdp, complains about it and responds with a 488. Mon Sep 08 2014 20:21:41 GMT+0300 (FLE Daylight Time) | sip.transport | received WebSocket text message: INVITE sip:lgvt0hci@lqne1q8dttn3.invalid;transport=ws SIP/2.0 Record-Route: <sip:PU.BL.IC.IP;transport=ws;r2=on;lr=on;ftag=as40c12073> Record-Route: <sip:PU.BL.IC.IP;r2=on;lr=on;ftag=as40c12073> Via: SIP/2.0/WS PU.BL.IC.IP;branch=z9hG4bKff35.a9b596f609aadeccdf58183a9fe4fcba.1 Via: SIP/2.0/UDP PU.BL.IC.IP:5070;branch=z9hG4bK39a2221b;rport=5070 Max-Forwards: 69 From: "660 win8" <sip:6...@testers.com>;tag=as40c12073 To: <sip:6...@pu.bl.ic.ip:5060> Contact: <sip:6...@pu.bl.ic.ip:5070> Call-ID: 597260a76cb0cb9155392f3a3c0be...@testers.com CSeq: 102 INVITE User-Agent: I Am the Devil Date: Mon, 08 Sep 2014 17:21:34 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 439 v=0 o=root 1283889088 1283889088 IN IP4 PU.BL.IC.IP s=Asterisk PBX 11.11.0 c=IN IP4 PU.BL.IC.IP t=0 0 m=audio 16822 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 ... Mon Sep 08 2014 20:21:41 GMT+0300 (FLE Daylight Time) | sip.transport | sending WebSocket message: SIP/2.0 100 Trying Via: SIP/2.0/WS PU.BL.IC.IP;branch=z9hG4bKff35.a9b596f609aadeccdf58183a9fe4fcba.1 Via: SIP/2.0/UDP PU.BL.IC.IP:5070;branch=z9hG4bK39a2221b;rport=5070 To: <sip:6...@pu.bl.ic.ip:5060> From: "660 win8" <sip:6...@testers.com>;tag=as40c12073 Call-ID: 597260a76cb0cb9155392f3a3c0be...@testers.com CSeq: 102 INVITE Supported: gruu,outbound Content-Length: 0 ... Mon Sep 08 2014 20:21:41 GMT+0300 (FLE Daylight Time) | sip.inviteservercontext | invalid SDP sip-0.6.2.js:2655 Mon Sep 08 2014 20:21:41 GMT+0300 (FLE Daylight Time) | sip.inviteservercontext | Failed to set remote offer sdp: Called with SDP without ice-ufrag and ice-pwd. sip-0.6.2.js:2655 Mon Sep 08 2014 20:21:41 GMT+0300 (FLE Daylight Time) | sip.transport | sending WebSocket message: SIP/2.0 488 Not Acceptable Here Via: SIP/2.0/WS PU.BL.IC.IP;branch=z9hG4bKff35.a9b596f609aadeccdf58183a9fe4fcba.0 Via: SIP/2.0/UDP PU.BL.IC.IP:5070;branch=z9hG4bK39a2221b;rport=5070 To: <sip:6...@pu.bl.ic.ip:5060>;tag=cl8lmb52gl From: "660 win8" <sip:6...@testers.com>;tag=as40c12073 Call-ID: 597260a76cb0cb9155392f3a3c0be...@testers.com CSeq: 102 INVITE Supported: gruu,outbound Content-Length: 0 ... Mon Sep 08 2014 20:21:41 GMT+0300 (FLE Daylight Time) | sip.transport | received WebSocket text message: ACK sip:v1vbuq35@0i03dp4lli27.invalid;transport=ws SIP/2.0 Via: SIP/2.0/WS PU.BL.IC.IP;branch=z9hG4bKff35.a9b596f609aadeccdf58183a9fe4fcba.0 Max-Forwards: 69 From: "660 win8" <sip:6...@testers.com>;tag=as40c12073 To: <sip:6...@pu.bl.ic.ip:5060>;tag=cl8lmb52gl Call-ID: 597260a76cb0cb9155392f3a3c0be...@testers.com CSeq: 102 ACK Content-Length: 0 Thanks, Olli 2014-09-08 18:50 GMT+03:00 Matthew Jordan <mjor...@digium.com>: > > > On Mon, Sep 8, 2014 at 10:19 AM, Olli Heiskanen < > ohjelmistoarkkite...@gmail.com> wrote: > >> Hi Matthew, >> >> Here's the debug output: >> >> >> >> >> >> <--- SIP read from UDP:PU.BL.IC.IP:5060 ---> >> INVITE sip:6...@testers.com 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:6...@testers.com> >> From: "660" <sip:6...@testers.com>;tag=856i7ei98p >> Call-ID: oc0ppijresm05k2emsgt >> CSeq: 3394 INVITE >> Contact: <sip:6...@testers.com >> ;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:6...@testers.com>;tag=856i7ei98p >> To: <sip:6...@testers.com> >> 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:6...@pu.bl.ic.ip: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:6...@pu.bl.ic.ip:5060 SIP/2.0 >> Via: SIP/2.0/UDP PU.BL.IC.IP:5070;branch=z9hG4bK4e738afa;rport >> Max-Forwards: 70 >> From: "660 win8" <sip:6...@testers.com>;tag=as73376885 >> To: <sip:6...@pu.bl.ic.ip:5060> >> Contact: <sip:6...@pu.bl.ic.ip:5070> >> Call-ID: 2f70cc9567be50a46ba2879d4391a...@testers.com >> 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 > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users