При попытке позвонить в консоле вываливается
[Nov 7 18:23:05] NOTICE[18439]: chan_sip.c:19661 handle_response_invite: Failed to authenticate on INVITE to '"20104" ;tag=as2e9c8365'
-- SIP/377xxxx-0000101d is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
Входящие звонки работают
sip show peers и sip show registry показывают OK и registered
sip.conf
| Код: |
| [general] register => 377xxxx:pass@77.88.204.3/377xxxx tlsenable=yes tlsbindaddr=192.168.1.72 tlscafile=/etc/asterisk/certificate/ca.pem ; путь к доверенному центру сертификации tlscertfile=/etc/asterisk/certificate/asterisk.pem ; путь к локальному сертификату сервера tlscipher=DES-CBC3-SHA tlsclientmethod=tlsv1 tlsdontverifyserver=no registertimeout = 60 registerattempts = 0 t38pt_udptl=yes,redundancy,maxdatagram=400 faxdetect=yes echocancel=yes allowguest=no allowoverlap=no allowtransfer=yes disallow=all allow=alaw language=ru faxdetect = no defaultexpiry = 2400 context=default nat=yes insecure = port,invite qualify=yes [377xxxx] nat = yes type = friend host = 77.88.204.3 port = 5060 canreinvite = no qualify = yes context = restaurant disallow = all allow = alaw fromuser = 377xxxx fromdomain = 77.88.204.3 secret = pass usereqphone=yes insecure = port [rest](!) type=friend secret=passw host=dynamic directmedia=no nat=yes call-limit=1 context=restaurant [20104](rest) |
extensions.conf
| Код: |
| [restaurant] exten => _044XXXXXXX,1,Set(FILENAME=${UNIQUEID}) exten => _044XXXXXXX,n,MixMonitor(/var/spool/asterisk/monitor/${FILENAME}.wav,b) ;exten => _044XXXXXXX,n,Dial(SIP/${EXTEN}@377xxxx,,rTt) exten => _044XXXXXXX,n,Dial(SIP/377xxxx/${EXTEN},,rTt) exten => _044XXXXXXX,n,StopMixMonitor() |
Дебаг при исходящем звонке с номера 20104 на 499-88-89
| Код: |
| Audio is at 5060 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 77.88.204.3:5060: INVITE sip:4998889@77.88.204.3:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP MyIP:5060;branch=z9hG4bK0978231d;rport Max-Forwards: 70 From: "20104" ;tag=as2e9c8365 To: Contact: Call-ID: 0f63cfc34962005a2e949b757ddb7dd6@MyIP:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.7.1 Date: Wed, 07 Nov 2012 16:23:05 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 261 v=0 o=root 1787154781 1787154781 IN IP4 MyIP s=Asterisk PBX 1.8.7.1 c=IN IP4 MyIP t=0 0 m=audio 12900 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP MyIP:5060;branch=z9hG4bK0978231d;received=MyIP;rport=5060 From: "20104" ;tag=as2e9c8365 To: ;tag=as01c42149 Call-ID: 0f63cfc34962005a2e949b757ddb7dd6@MyIP:5060 CSeq: 102 INVITE Server: Asterisk PBX 1.6.2.24 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1044ae41" Content-Length: 0 --- (11 headers 0 lines) --- Transmitting (NAT) to 77.88.204.3:5060: ACK sip:4998889@MyIP:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP MyIP:5060;branch=z9hG4bK0978231d;rport Max-Forwards: 70 From: "20104" ;tag=as2e9c8365 To: ;tag=as01c42149 Contact: Call-ID: 0f63cfc34962005a2e949b757ddb7dd6@MyIP:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 1.8.7.1 Content-Length: 0 --- Really destroying SIP dialog '0f63cfc34962005a2e949b757ddb7dd6@MyIP:5060' Method: INVITE |
Последний раз редактировалось: rrv (Чт Ноя 08, 2012 13:53)
В первом посте забыл указать правило диалплана на 7значные номера,
| Код: |
| [restaurant] exten => _XXXXXXX,1,Set(FILENAME=${UNIQUEID}) exten => _XXXXXXX,n,MixMonitor(/var/spool/asterisk/monitor/${FILENAME}.wav,b) exten => _XXXXXXX,n,Dial(SIP/377xxxx/044${EXTEN},,rTt) exten => _XXXXXXX,n,StopMixMonitor() |
Но суть проблемы это не меняет, провайдер говорит, что у них все в порядке.
Подключил сип напрямую через программный телефон
При попытке исходящего дозвониться тоже не получается
Ниже вывод tcpdump'a машины, на которой установлен программный телефон
| Код: |
| ruslan@ruslan-it:~$ sudo tcpdump -nnv host 77.88.204.3 tcpdump: listening on eth0, link-type EN10MB (Ethernet), capture size 65535 bytes 14:28:57.903969 IP (tos 0x0, ttl 64, id 4317, offset 0, flags [DF], proto UDP (17), length 1500) 192.168.11.44.34295 > 77.88.204.3.56384: UDP, length 1472 14:28:57.913618 IP (tos 0x0, ttl 59, id 2322, offset 0, flags [none], proto ICMP (1), length 576) 77.88.204.3 > 192.168.11.44: ICMP 77.88.204.3 udp port 56384 unreachable, length 556 IP (tos 0x0, ttl 59, id 4317, offset 0, flags [DF], proto UDP (17), length 1500) 192.168.11.44.34295 > 77.88.204.3.56384: UDP, length 1472 14:28:58.414505 IP (tos 0x68, ttl 64, id 0, offset 0, flags [DF], proto UDP (17), length 864) 192.168.11.44.5060 > 77.88.204.3.5060: SIP, length: 836 INVITE sip:4998889@77.88.204.3 SIP/2.0 Via: SIP/2.0/UDP 192.168.11.44:5060;rport;branch=z9hG4bK1559911004 From: ;tag=382982056 To: Call-ID: 996679304 CSeq: 20 INVITE Contact: Content-Type: application/sdp Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Max-Forwards: 70 User-Agent: Linphone/3.5.2 (eXosip2/3.6.0) Subject: Phone call Content-Length: 350 v=0 o=377xxxx 1478 1478 IN IP4 192.168.11.44 s=Talk c=IN IP4 192.168.11.44 t=0 0 m=audio 7078 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 m=video 9078 RTP/AVP 99 98 a=rtpmap:99 MP4V-ES/90000 a=fmtp:99 profile-level-id=3 a=rtpmap:98 H263-1998/90000 a=fmtp:98 CIF=1;QCIF=1 14:28:58.414696 IP (tos 0x68, ttl 64, id 0, offset 0, flags [DF], proto UDP (17), length 32) 192.168.11.44.5060 > 77.88.204.3.5060: SIP, length: 4 jaK[|sip] 14:28:58.424232 IP (tos 0x0, ttl 59, id 2324, offset 0, flags [none], proto UDP (17), length 508) 77.88.204.3.5060 > 192.168.11.44.5060: SIP, length: 480 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.11.44:5060;branch=z9hG4bK1559911004;received=MyIP;rport=1809 From: ;tag=382982056 To: ;tag=as6c3e4f10 Call-ID: 996679304 CSeq: 20 INVITE Server: Asterisk PBX 1.6.2.24 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3b04acdf" Content-Length: 0 14:28:58.424676 IP (tos 0x68, ttl 64, id 0, offset 0, flags [DF], proto UDP (17), length 281) 192.168.11.44.5060 > 77.88.204.3.5060: SIP, length: 253 ACK sip:4998889@77.88.204.3 SIP/2.0 Via: SIP/2.0/UDP 192.168.11.44:5060;rport;branch=z9hG4bK1559911004 From: ;tag=382982056 To: ;tag=as6c3e4f10 Call-ID: 996679304 CSeq: 20 ACK Content-Length: 0 14:28:58.453139 IP (tos 0x68, ttl 64, id 0, offset 0, flags [DF], proto UDP (17), length 1032) 192.168.11.44.5060 > 77.88.204.3.5060: SIP, length: 1004 INVITE sip:4998889@77.88.204.3 SIP/2.0 Via: SIP/2.0/UDP 192.168.11.44:5060;rport;branch=z9hG4bK565033930 From: ;tag=382982056 To: Call-ID: 996679304 CSeq: 21 INVITE Contact: Authorization: Digest username="377xxxx", realm="asterisk", nonce="3b04acdf", uri="sip:4998889@77.88.204.3", response="2a11de9c7180eff48e220cca50db61c8", algorithm=MD5 Content-Type: application/sdp Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Max-Forwards: 70 User-Agent: Linphone/3.5.2 (eXosip2/3.6.0) Subject: Phone call Content-Length: 350 v=0 o=377xxxx 1478 1478 IN IP4 192.168.11.44 s=Talk c=IN IP4 192.168.11.44 t=0 0 m=audio 7078 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 m=video 9078 RTP/AVP 99 98 a=rtpmap:99 MP4V-ES/90000 a=fmtp:99 profile-level-id=3 a=rtpmap:98 H263-1998/90000 a=fmtp:98 CIF=1;QCIF=1 14:28:58.462930 IP (tos 0x0, ttl 59, id 2325, offset 0, flags [none], proto UDP (17), length 446) 77.88.204.3.5060 > 192.168.11.44.5060: SIP, length: 418 SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.11.44:5060;branch=z9hG4bK565033930;received=MyIP;rport=1809 From: ;tag=382982056 To: Call-ID: 996679304 CSeq: 21 INVITE Server: Asterisk PBX 1.6.2.24 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Length: 0 14:28:59.479244 IP (tos 0x0, ttl 59, id 2328, offset 0, flags [none], proto UDP (17), length 509) 77.88.204.3.5060 > 192.168.11.44.5060: SIP, length: 481 SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP 192.168.11.44:5060;branch=z9hG4bK565033930;received=MyIP;rport=1809 From: ;tag=382982056 To: ;tag=as69de9bd4 Call-ID: 996679304 CSeq: 21 INVITE Server: Asterisk PBX 1.6.2.24 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer X-Asterisk-HangupCause: Call Rejected X-Asterisk-HangupCauseCode: 21 Content-Length: 0 14:28:59.479451 IP (tos 0x68, ttl 64, id 0, offset 0, flags [DF], proto UDP (17), length 280) 192.168.11.44.5060 > 77.88.204.3.5060: SIP, length: 252 ACK sip:4998889@77.88.204.3 SIP/2.0 Via: SIP/2.0/UDP 192.168.11.44:5060;rport;branch=z9hG4bK565033930 From: ;tag=382982056 To: ;tag=as69de9bd4 Call-ID: 996679304 CSeq: 21 ACK Content-Length: 0 |
Added after 3 hours 2 minutes:
Проблема решена, была на стороне провайдера.
Всем спасибо за участие
