2)Обратное выполняется успешно
3)Ошибки:
-на веб клиенте: "Bad media description"
-на веб клиенте в логе можно найти:cause=488; text="Not Acceptable Here"
-в Asterisk'e:
| Код: |
| [Feb 25 16:37:08] ERROR[1890][C-00000000]: netsock2.c:269 ast_sockaddr_resolve: getaddrinfo("d82157g3bs9f.invalid", "(null)", ...): Name or service not known [Feb 25 16:37:08] WARNING[1890][C-00000000]: chan_sip.c:15763 __set_address_from_contact: Invalid host name in Contact: (can't resolve in DNS) : 'd82157g3bs9f.invalid' -- SIP/1060-00000001 answered SIP/1059-00000000 [Feb 25 16:37:08] WARNING[1897][C-00000000]: res_srtp.c:397 ast_srtp_unprotect: SRTP unprotect failed with: authentication failure 10 == Spawn extension (disp-112, 1060, 1) exited non-zero on 'SIP/1059-00000000' == Using SIP RTP CoS mark 5 |
4) 1060 это веб юзер
1002 cisco ip phone
1061 x-lite
Остальные юзеры не используются.
sip.conf:
| Код: |
| [general] context = public ; Default context for incoming calls. Defaults to 'default' allowoverlap = no ; Disable overlap dialing support. (Default is yes) realm = telecor.ru udpbindaddr = 0.0.0.0:5060 tcpenable = no ; Enable server for incoming TCP connections (default is no) tcpbindaddr = 0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces) transport = udp,ws,wss srvlookup = yes ; Enable DNS SRV lookups on outbound calls avpf = yes ; Enable inter-operability with media streams using the AVPF RTP profile. subscribecontext = default [authentication] [basic-options](!); a template dtmfmode = rfc2833 context = from-office type = friend [natted-phone](!,basic-options); another template inheriting basic-options directmedia = no host = dynamic [public-phone](!,basic-options); another template inheriting basic-options directmedia = yes [my-codecs](!); a template for my preferred codecs disallow = all allow = ilbc allow = g729 allow = gsm allow = g723 allow = ulaw [ulaw-phone](!); and another one for ulaw-only disallow = all allow = ulaw [1060] type=friend host=dynamic secret=1060 context=disp-112 hasiax = no hassip = yes encryption = yes avpf = yes icesupport = yes videosupport=no directmedia=no disallow=all allow=alaw allow=ulaw [1061] type=friend host=dynamic secret=1061 context=disp-112 hasiax = no hassip = yes icesupport = yes [1002] type=friend host=dynamic secret=1111 context=disp-112 [1059] type=friend host=dynamic secret=1059 context=disp-112 hasiax = no hassip = yes encryption = yes avpf = yes icesupport = yes videosupport=no directmedia=no disallow=all allow=alaw allow=ulaw |
extension.conf:
| Код: |
| [default] [disp-112] exten => _XXXX,1,Dial(SIP/${EXTEN}) [general] static = yes writeprotect = no clearglobalvars = yes [globals] FEATURES = DIALOPTIONS = RINGTIME = 20 FOLLOWMEOPTIONS = PAGING_HEADER = Intercom |
Asterisk развёрнут на Ubuntu 12.04 LTS
добавите в /etc/host соответствующую запись
| awsswa @ Пн Фев 25, 2013 14:46 писал(а): |
| d82157g3bs9f.invalid - имя машины не резольвится по DNS добавите в /etc/host соответствующую запись |
Это я и сам понял по сообщению, но вам не кажется, что тут проблема не в этом?
Потому что у меня нет машины с таким именем, да и в сообщении об ошибке это имя каждый раз разное
У вас похоже web-клиент подставляет в SIP-поле "Contact:" не корректное значение, и астериску это не нравится.
выяснил, что эта ошибка не оказывает влияния. и возникает при звонке с sip phone на web, а этот звонок проходит нормально.
проблема возникает когда звонишь с веб клиента на ip phone(или на что-либо другое, например x-lite).
JsSIP | TRANSPORT | sending WebSocket message:
BYE sip:1002@192.168.0.5:5060;transport=ws SIP/2.0
Via: SIP/2.0/WS qrka9d5monhg.invalid;branch=z9hG4bK7057579
Max-Forwards: 69
To: ;tag=as125b7421
From: "1060" ;tag=npaimck7m5
Call-ID: mt9dhrdhfh0eo80doboi
CSeq: 8387 BYE
Reason: SIP ;cause=488; text="Not Acceptable Here"
Supported: path, outbound, gruu
User-Agent: JsSIP 0.3.0-devel
Content-Length: 0
| Цитата: |
| Via: SIP/2.0/WS qrka9d5monhg.invalid;branch=z9hG4bK7057579 |
Должен быть вместо "qrka9d5monhg.invalid" быть нормальный ip или нормальное dns.
Выложили бы весь диалог.
это лог начиная с того момента как я начал звонить с веба на айпифон
| Код: |
| JsSIP | EVENT EMITTER | adding event newDTMF jssip-0.3.0-devel.js:61 JsSIP | EVENT EMITTER | adding event ended jssip-0.3.0-devel.js:61 JsSIP | EVENT EMITTER | adding event started jssip-0.3.0-devel.js:61 JsSIP | EVENT EMITTER | adding event failed jssip-0.3.0-devel.js:61 JsSIP | EVENT EMITTER | adding event progress jssip-0.3.0-devel.js:61 JsSIP | EVENT EMITTER | adding event connecting jssip-0.3.0-devel.js:61 JsSIP | EVENT EMITTER | emitting event newSession jssip-0.3.0-devel.js:181 JsSIP | EVENT EMITTER | new listener added to event progress jssip-0.3.0-devel.js:57 JsSIP | EVENT EMITTER | new listener added to event started jssip-0.3.0-devel.js:57 JsSIP | EVENT EMITTER | new listener added to event failed jssip-0.3.0-devel.js:57 JsSIP | EVENT EMITTER | new listener added to event newDTMF jssip-0.3.0-devel.js:57 JsSIP | EVENT EMITTER | new listener added to event ended jssip-0.3.0-devel.js:57 JsSIP | EVENT EMITTER | emitting event connecting jssip-0.3.0-devel.js:181 JsSIP | MEDIA SESSION | requesting access to local media jssip-0.3.0-devel.js:4669 JsSIP | MEDIA SESSION | got stream: [object LocalMediaStream] jssip-0.3.0-devel.js:4651 JsSIP | MEDIA SESSION | PeerConnection state changed to opening | ICE state: undefined jssip-0.3.0-devel.js:4627 JsSIP | MEDIA SESSION | ICE candidate received: a=candidate:2458163306 1 udp 2113937151 192.168.0.156 51717 typ host generation 0 jssip-0.3.0-devel.js:4604 JsSIP | MEDIA SESSION | ICE candidate received: a=candidate:2458163306 2 udp 2113937151 192.168.0.156 51717 typ host generation 0 jssip-0.3.0-devel.js:4604 JsSIP | MEDIA SESSION | ICE candidate received: a=candidate:332177118 1 udp 1845501695 188.162.221.228 63941 typ srflx raddr 192.168.0.156 rport 51717 generation 0 jssip-0.3.0-devel.js:4604 JsSIP | MEDIA SESSION | ICE candidate received: a=candidate:332177118 2 udp 1845501695 188.162.221.228 63941 typ srflx raddr 192.168.0.156 rport 51717 generation 0 jssip-0.3.0-devel.js:4604 JsSIP | MEDIA SESSION | ICE candidate received: a=candidate:3691472026 1 tcp 1509957375 192.168.0.156 56022 typ host generation 0 jssip-0.3.0-devel.js:4604 JsSIP | MEDIA SESSION | ICE candidate received: a=candidate:3691472026 2 tcp 1509957375 192.168.0.156 56022 typ host generation 0 jssip-0.3.0-devel.js:4604 JsSIP | MEDIA SESSION | no more ICE candidates | PeerConnection state: opening | ICE state: undefined jssip-0.3.0-devel.js:4606 JsSIP | TRANSPORT | sending WebSocket message: INVITE sip:1002@192.168.0.5 SIP/2.0 Via: SIP/2.0/WS qrka9d5monhg.invalid;branch=z9hG4bK5011153 Max-Forwards: 69 To: From: "1060" ;tag=0l3ogdr7os Call-ID: mt9dhlb272ut4ui1pktn CSeq: 2590 INVITE Contact: Allow: ACK,CANCEL,BYE,OPTIONS,INVITE,MESSAGE Content-Type: application/sdp Supported: path, outbound, gruu User-Agent: JsSIP 0.3.0-devel Content-Length: 1599 v=0 o=- 2936281083 2 IN IP4 127.0.0.1 s=- t=0 0 a=group:BUNDLE audio a=msid-semantic: WMS qhEZJhzHX43PnhUEOqT6jU6MqlO7HYioqN5L m=audio 63941 RTP/SAVPF 103 104 111 0 8 107 106 105 13 126 c=IN IP4 188.162.221.228 a=rtcp:63941 IN IP4 188.162.221.228 a=candidate:2458163306 1 udp 2113937151 192.168.0.156 51717 typ host generation 0 a=candidate:2458163306 2 udp 2113937151 192.168.0.156 51717 typ host generation 0 a=candidate:332177118 1 udp 1845501695 188.162.221.228 63941 typ srflx raddr 192.168.0.156 rport 51717 generation 0 a=candidate:332177118 2 udp 1845501695 188.162.221.228 63941 typ srflx raddr 192.168.0.156 rport 51717 generation 0 a=candidate:3691472026 1 tcp 1509957375 192.168.0.156 56022 typ host generation 0 a=candidate:3691472026 2 tcp 1509957375 192.168.0.156 56022 typ host generation 0 a=ice-ufrag:tt8zTXhuHN82Ycqt a=ice-pwd:EzSstInsvuRsPHJ5eEm7Zi1T a=ice-options:google-ice a=sendrecv a=mid:audio a=rtcp-mux a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:7xbCvhQwUpfdCPIAk4nzDauvMK3l9sWIGlyLh+/l a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:R5CPl/gPmcEQzmjOVUZnhupuuzNf/VNVILwfpKtf a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:111 opus/48000/2 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:107 CN/48000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 a=ssrc:2753573187 cname:YmG+5wXB+c7y0yNW a=ssrc:2753573187 msid:qhEZJhzHX43PnhUEOqT6jU6MqlO7HYioqN5L a0 a=ssrc:2753573187 mslabel:qhEZJhzHX43PnhUEOqT6jU6MqlO7HYioqN5L a=ssrc:2753573187 label:qhEZJhzHX43PnhUEOqT6jU6MqlO7HYioqN5La0 jssip-0.3.0-devel.js:502 JsSIP | TRANSPORT | received WebSocket text message: SIP/2.0 401 Unauthorized Via: SIP/2.0/WS qrka9d5monhg.invalid;branch=z9hG4bK5011153;received=192.168.0.156 From: "1060" ;tag=0l3ogdr7os To: ;tag=as22f7357d Call-ID: mt9dhlb272ut4ui1pktn CSeq: 2590 INVITE Server: Asterisk PBX 11.2.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="telecor.ru", nonce="09045986" Content-Length: 0 jssip-0.3.0-devel.js:653 JsSIP | TRANSPORT | sending WebSocket message: ACK sip:1002@192.168.0.5 SIP/2.0 Via: SIP/2.0/WS qrka9d5monhg.invalid;branch=z9hG4bK5011153 To: ;tag=as22f7357d From: "1060" ;tag=0l3ogdr7os Call-ID: mt9dhlb272ut4ui1pktn CSeq: 2590 ACK jssip-0.3.0-devel.js:502 JsSIP | TRANSPORT | sending WebSocket message: INVITE sip:1002@192.168.0.5 SIP/2.0 Via: SIP/2.0/WS qrka9d5monhg.invalid;branch=z9hG4bK4949216 Max-Forwards: 69 To: From: "1060" ;tag=0l3ogdr7os Call-ID: mt9dhlb272ut4ui1pktn CSeq: 2591 INVITE Authorization: Digest username="1060",realm="telecor.ru",nonce="09045986",uri="sip:1002@192.168.0.5",response="4ec3b1f869d4e89f905ef18715a66288",algorithm=MD5 Contact: Allow: ACK,CANCEL,BYE,OPTIONS,INVITE,MESSAGE Content-Type: application/sdp Supported: path, outbound, gruu User-Agent: JsSIP 0.3.0-devel Content-Length: 1599 v=0 o=- 2936281083 2 IN IP4 127.0.0.1 s=- t=0 0 a=group:BUNDLE audio a=msid-semantic: WMS qhEZJhzHX43PnhUEOqT6jU6MqlO7HYioqN5L m=audio 63941 RTP/SAVPF 103 104 111 0 8 107 106 105 13 126 c=IN IP4 188.162.221.228 a=rtcp:63941 IN IP4 188.162.221.228 a=candidate:2458163306 1 udp 2113937151 192.168.0.156 51717 typ host generation 0 a=candidate:2458163306 2 udp 2113937151 192.168.0.156 51717 typ host generation 0 a=candidate:332177118 1 udp 1845501695 188.162.221.228 63941 typ srflx raddr 192.168.0.156 rport 51717 generation 0 a=candidate:332177118 2 udp 1845501695 188.162.221.228 63941 typ srflx raddr 192.168.0.156 rport 51717 generation 0 a=candidate:3691472026 1 tcp 1509957375 192.168.0.156 56022 typ host generation 0 a=candidate:3691472026 2 tcp 1509957375 192.168.0.156 56022 typ host generation 0 a=ice-ufrag:tt8zTXhuHN82Ycqt a=ice-pwd:EzSstInsvuRsPHJ5eEm7Zi1T a=ice-options:google-ice a=sendrecv a=mid:audio a=rtcp-mux a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:7xbCvhQwUpfdCPIAk4nzDauvMK3l9sWIGlyLh+/l a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:R5CPl/gPmcEQzmjOVUZnhupuuzNf/VNVILwfpKtf a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:111 opus/48000/2 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:107 CN/48000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 a=ssrc:2753573187 cname:YmG+5wXB+c7y0yNW a=ssrc:2753573187 msid:qhEZJhzHX43PnhUEOqT6jU6MqlO7HYioqN5L a0 a=ssrc:2753573187 mslabel:qhEZJhzHX43PnhUEOqT6jU6MqlO7HYioqN5L a=ssrc:2753573187 label:qhEZJhzHX43PnhUEOqT6jU6MqlO7HYioqN5La0 jssip-0.3.0-devel.js:502 JsSIP | TRANSACTION | Timer D expired for INVITE client transaction z9hG4bK5011153 jssip-0.3.0-devel.js:1913 JsSIP | TRANSPORT | received WebSocket text message: SIP/2.0 100 Trying Via: SIP/2.0/WS qrka9d5monhg.invalid;branch=z9hG4bK4949216;received=192.168.0.156 From: "1060" ;tag=0l3ogdr7os To: Call-ID: mt9dhlb272ut4ui1pktn CSeq: 2591 INVITE Server: Asterisk PBX 11.2.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 jssip-0.3.0-devel.js:653 JsSIP | TRANSPORT | received WebSocket text message: SIP/2.0 180 Ringing Via: SIP/2.0/WS qrka9d5monhg.invalid;branch=z9hG4bK4949216;received=192.168.0.156 From: "1060" ;tag=0l3ogdr7os To: ;tag=as7392a136 Call-ID: mt9dhlb272ut4ui1pktn CSeq: 2591 INVITE Server: Asterisk PBX 11.2.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 jssip-0.3.0-devel.js:653 JsSIP | DIALOG | new UAC dialog created with status EARLY jssip-0.3.0-devel.js:2493 JsSIP | EVENT EMITTER | emitting event progress |
здесь я снял трубку и получил ошибку, в нём можно найти 488 ошибку.
| Код: |
| JsSIP | TRANSPORT | received WebSocket text message: SIP/2.0 200 OK Via: SIP/2.0/WS qrka9d5monhg.invalid;branch=z9hG4bK4949216;received=192.168.0.156 From: "1060" ;tag=0l3ogdr7os To: ;tag=as7392a136 Call-ID: mt9dhlb272ut4ui1pktn CSeq: 2591 INVITE Server: Asterisk PBX 11.2.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 687 v=0 o=root 1634110579 1634110579 IN IP4 192.168.0.5 s=Asterisk PBX 11.2.1 c=IN IP4 192.168.0.5 t=0 0 m=audio 19372 RTP/SAVPF 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=ice-ufrag:56e3d69618d0795413bc2dad5fed3c83 a=ice-pwd:48f75d0b5471a197754f6fe77533be7f a=candidate:Hc0a80005 1 UDP 2130706431 192.168.0.5 19372 typ host a=candidate:Sbca2dde4 1 UDP 1694498815 188.162.221.228 63944 typ srflx a=candidate:Hc0a80005 2 UDP 2130706430 192.168.0.5 19373 typ host a=candidate:Sbca2dde4 2 UDP 1694498814 188.162.221.228 63944 typ srflx a=sendrecv a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:Fywlq1bEZfCxU/RJtnt1+dKSE4eqB2MJXvADcD88 jssip-0.3.0-devel.js:653 JsSIP | MEDIA SESSION | PeerConnection state changed to active | ICE state: undefined jssip-0.3.0-devel.js:4627 SetRemoteDescription failed. jssip-0.3.0-devel.js:3724 JsSIP | DIALOG | dialog mt9dhlb272ut4ui1pktn0l3ogdr7osas7392a136 changed to CONFIRMED state jssip-0.3.0-devel.js:2504 JsSIP | TRANSPORT | sending WebSocket message: ACK sip:1002@192.168.0.5:5060;transport=ws SIP/2.0 Via: SIP/2.0/WS qrka9d5monhg.invalid;branch=z9hG4bK3428942 Max-Forwards: 69 To: ;tag=as7392a136 From: "1060" ;tag=0l3ogdr7os Call-ID: mt9dhlb272ut4ui1pktn CSeq: 2591 ACK Supported: path, outbound, gruu User-Agent: JsSIP 0.3.0-devel Content-Length: 0 jssip-0.3.0-devel.js:502 JsSIP | TRANSPORT | sending WebSocket message: BYE sip:1002@192.168.0.5:5060;transport=ws SIP/2.0 Via: SIP/2.0/WS qrka9d5monhg.invalid;branch=z9hG4bK4326091 Max-Forwards: 69 To: ;tag=as7392a136 From: "1060" ;tag=0l3ogdr7os Call-ID: mt9dhlb272ut4ui1pktn CSeq: 2592 BYE Reason: SIP ;cause=488; text="Not Acceptable Here" Supported: path, outbound, gruu User-Agent: JsSIP 0.3.0-devel Content-Length: 0 jssip-0.3.0-devel.js:502 JsSIP | SESSION | closing INVITE session mt9dhlb272ut4ui1pktn0l3ogdr7os jssip-0.3.0-devel.js:3310 JsSIP | MEDIA SESSION | closing PeerConnection jssip-0.3.0-devel.js:4632 JsSIP | DIALOG | dialog mt9dhlb272ut4ui1pktn0l3ogdr7osas7392a136 deleted jssip-0.3.0-devel.js:2513 JsSIP | EVENT EMITTER | emitting event failed jssip-0.3.0-devel.js:181 JsSIP | MEDIA SESSION | PeerConnection state changed to closed | ICE state: undefined jssip-0.3.0-devel.js:4627 JsSIP | TRANSPORT | received WebSocket text message: SIP/2.0 200 OK Via: SIP/2.0/WS qrka9d5monhg.invalid;branch=z9hG4bK4326091;received=192.168.0.156 From: "1060" ;tag=0l3ogdr7os To: ;tag=as7392a136 Call-ID: mt9dhlb272ut4ui1pktn CSeq: 2592 BYE Server: Asterisk PBX 11.2.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 |