почему did это логин а не номер? DJs3000 ( 2011-09-01 16:48:50 +0400 )редактировать
Через вебморду эластикса создаю транк и настраиваю инбоунд при звонке вместо номера вижу логин учетной записи!
== Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Executing [HellKlUsov1@from-trunk-sip-arctel:1] Set("SIP/arctel-000000ba", "GROUP()=OUT_4") in new stack -- Executing [HellKlUsov1@from-trunk-sip-arctel:2] Goto("SIP/arctel-000000ba", "from-trunk,HellKlUsov1,1") in new stack -- Goto (from-trunk,HellKlUsov1,1) -- Executing [HellKlUsov1@from-trunk:1] Dial("SIP/arctel-000000ba", "SIP/305") in new stack == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Couldn't call 305 == Everyone is busy/congested at this time (0:0/0/0) -- Executing [HellKlUsov1@from-trunk:2] NoOp("SIP/arctel-000000ba", "Received an unknown call with DID set to HellKlUsov1") in new stack -- Executing [HellKlUsov1@from-trunk:3] Goto("SIP/arctel-000000ba", "s,a2") in new stack -- Goto (from-trunk,s,2) -- Executing [s@from-trunk:2] Answer("SIP/arctel-000000ba", "") in new stack -- Executing [s@from-trunk:3] Wait("SIP/arctel-000000ba", "2") in new stack -- Executing [s@from-trunk:4] Playback("SIP/arctel-000000ba", "ss-noservice") in new stack -- Executing [s@from-trunk:5] SayAlpha("SIP/arctel-000000ba", "") in new stack -- Executing [s@from-trunk:6] Hangup("SIP/arctel-000000ba", "") in new stack == Spawn extension (from-trunk, s, 6) exited non-zero on 'SIP/arctel-000000ba' -- Executing [h@from-trunk:1] Hangup("SIP/arctel-000000ba", "") in new stack Транк:
host=86.110.4.хх fromdomain=86.110.4.хх fromuser=HellKlUsov1 username=HellKlUsov1 secret=**** dtmfmode=rfc2833 type=peer disallow=all allow=alaw&ulaw&g729 nat=yes canreinvite=no insecure=invite secret=**** type=friend context=from-trunk HellKlUsov1:****@86.110.4.хх/HellKlUsov1 как задано в транке, на то и приходит вызов
register => [peer?][transport://]user[@domain][:secret[:authuser]]@host[:port][/extension][~expiry] после косой черты должен быть прописан did
ну вот дебага ток хз что тут понять можно
v=0 o=HuaweiSoftX3000 46524232 46524232 IN IP4 86.110.4.148 s=Sip Call c=IN IP4 86.110.4.148 t=0 0 m=audio 46340 RTP/AVP 18 4 97 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-15 a=fmtp:18 annexb=yes <-------------> --- (13 headers 11 lines) --- == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 Sending to 86.110.4.148 : 5060 (NAT) Using INVITE request as basis request - SBCbdeac770d39dcea08d78d2e05f0570da@172.21.0.2 Found peer 'arctel' for 'Anonymous' from 86.110.4.148:5060 <--- Reliably Transmitting (NAT) to 86.110.4.148:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 86.110.4.148:5060;branch=z9hG4bK80c761367163c159404141562;received=86.110.4.148 From: Anonymous <sip:Anonymous@86.110.4.148;user=phone>;tag=8013836e To: <sip:4952253038@213.247.249.2;user=phone>;tag=as176b56c3 Call-ID: SBCbdeac770d39dcea08d78d2e05f0570da@172.21.0.2 CSeq: 1 INVITE Server: Asterisk PBX 1.6.2.13 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7258ac33" Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'SBCbdeac770d39dcea08d78d2e05f0570da@172.21.0.2' in 32000 ms (Method: INVITE) <--- SIP read from UDP:86.110.4.148:5060 ---> ACK sip:HellKlUsov1@213.247.249.2:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 86.110.4.148:5060;branch=z9hG4bK80c761367163c159404141562 Call-ID: SBCbdeac770d39dcea08d78d2e05f0570da@172.21.0.2 From: Anonymous <sip:Anonymous@86.110.4.148;user=phone>;tag=8013836e To: <sip:4952253038@213.247.249.2;user=phone>;tag=as176b56c3 CSeq: 1 ACK Max-Forwards: 70 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from UDP:86.110.4.148:5060 ---> INVITE sip:HellKlUsov1@213.247.249.2:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 86.110.4.148:5060;branch=z9hG4bKa28d90ca664e5723cadcd2c29 Call-ID: SBC54877ac3dba80e99813bf2d8b2c6787f@172.21.0.2 From: Anonymous <sip:Anonymous@86.110.4.148;user=phone>;tag=c58eca08 To: <sip:4952253038@213.247.249.2;user=phone> CSeq: 1 INVITE Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REGISTER,PRACK,INFO,UPDATE,SUBSCRIBE,NOTIFY,MESSAGE,REFER Max-Forwards: 70 Supported: 100rel User-Agent: Huawei SoftX3000 V300R006 Contact: <sip:Anonymous@86.110.4.148:5060;user=phone> Content-Length: 252 Content-Type: application/sdp v=0 o=HuaweiSoftX3000 46521956 46521956 IN IP4 86.110.4.148 s=Sip Call c=IN IP4 86.110.4.148 t=0 0 m=audio 46354 RTP/AVP 18 4 97 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-15 a=fmtp:18 annexb=yes <-------------> --- (13 headers 11 lines) --- == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 Sending to 86.110.4.148 : 5060 (NAT) Using INVITE request as basis request - SBC54877ac3dba80e99813bf2d8b2c6787f@172.21.0.2 Found peer 'arctel' for 'Anonymous' from 86.110.4.148:5060 <--- Reliably Transmitting (NAT) to 86.110.4.148:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 86.110.4.148:5060;branch=z9hG4bKa28d90ca664e5723cadcd2c29;received=86.110.4.148 From: Anonymous <sip:Anonymous@86.110.4.148;user=phone>;tag=c58eca08 To: <sip:4952253038@213.247.249.2;user=phone>;tag=as445945a6 Call-ID: SBC54877ac3dba80e99813bf2d8b2c6787f@172.21.0.2 CSeq: 1 INVITE Server: Asterisk PBX 1.6.2.13 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="030b4b68" Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'SBC54877ac3dba80e99813bf2d8b2c6787f@172.21.0.2' in 32000 ms (Method: INVITE) <--- SIP read from UDP:86.110.4.148:5060 ---> ACK sip:HellKlUsov1@213.247.249.2:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 86.110.4.148:5060;branch=z9hG4bKa28d90ca664e5723cadcd2c29 Call-ID: SBC54877ac3dba80e99813bf2d8b2c6787f@172.21.0.2 From: Anonymous <sip:Anonymous@86.110.4.148;user=phone>;tag=c58eca08 To: <sip:4952253038@213.247.249.2;user=phone>;tag=as445945a6 CSeq: 1 ACK Max-Forwards: 70 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from UDP:86.110.4.148:5060 ---> INVITE sip:HellKlUsov1@213.247.249.2:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 86.110.4.148:5060;branch=z9hG4bK9731a67203af000e185554203 Call-ID: SBC34a2ce634f96346e74b67164c5bb3d2f@172.21.0.2 From: Anonymous <sip:Anonymous@86.110.4.148;user=phone>;tag=f8677964 To: <sip:4952253038@213.247.249.2;user=phone> CSeq: 1 INVITE Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REGISTER,PRACK,INFO,UPDATE,SUBSCRIBE,NOTIFY,MESSAGE,REFER Max-Forwards: 70 Supported: 100rel User-Agent: Huawei SoftX3000 V300R006 Contact: <sip:Anonymous@86.110.4.148:5060;user=phone> Content-Length: 252 Content-Type: application/sdp v=0 o=HuaweiSoftX3000 57252407 57252407 IN IP4 86.110.4.148 s=Sip Call c=IN IP4 86.110.4.148 t=0 0 m=audio 46370 RTP/AVP 18 4 97 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-15 a=fmtp:18 annexb=yes <-------------> --- (13 headers 11 lines) --- == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 Sending to 86.110.4.148 : 5060 (NAT) Using INVITE request as basis request - SBC34a2ce634f96346e74b67164c5bb3d2f@172.21.0.2 Found peer 'arctel' for 'Anonymous' from 86.110.4.148:5060 <--- Reliably Transmitting (NAT) to 86.110.4.148:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 86.110.4.148:5060;branch=z9hG4bK9731a67203af000e185554203;received=86.110.4.148 From: Anonymous <sip:Anonymous@86.110.4.148;user=phone>;tag=f8677964 To: <sip:4952253038@213.247.249.2;user=phone>;tag=as0dcf82f0 Call-ID: SBC34a2ce634f96346e74b67164c5bb3d2f@172.21.0.2 CSeq: 1 INVITE Server: Asterisk PBX 1.6.2.13 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="46fe7060" Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'SBC34a2ce634f96346e74b67164c5bb3d2f@172.21.0.2' in 32000 ms (Method: INVITE) <--- SIP read from UDP:86.110.4.148:5060 ---> ACK sip:HellKlUsov1@213.247.249.2:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 86.110.4.148:5060;branch=z9hG4bK9731a67203af000e185554203 Call-ID: SBC34a2ce634f96346e74b67164c5bb3d2f@172.21.0.2 From: Anonymous <sip:Anonymous@86.110.4.148;user=phone>;tag=f8677964 To: <sip:4952253038@213.247.249.2;user=phone>;tag=as0dcf82f0 CSeq: 1 ACK Max-Forwards: 70 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from UDP:86.110.4.148:5060 ---> Задан: 2011-09-01 16:05:50 +0400
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Обновлен: Sep 01 '11
Проект компании "АТС Дизайн"
Asterisk® и Digium® являются зарегистрированными торговыми марками компании Digium, Inc., США.
IP АТС Asterisk распространяется под лицензией GNU GPL.