Galera, bom dia. Há um tempo atras eu postei uma dúvida quanto a interligar dois PBXIP via SIP. O Mestre ASterisk me respondeu com um link para dois tutorias do seu site. Um deles, esse aqui: http://mestreasterisk.com.br/configuracao/interligar-servidores-asterisk-via-sip/ eu o segui, mas quanto tento fazer ligação do PBXIP1 para o 2, no log do 1 aparecem umas mensagens estranhas, do tipo: 489 Bad Event, SIP/2.0 403 Forbidden. No 2 não aparece nada.... Eu joguei o log num arquivo .txt... e anexei o mesmo aqui no email... Se alguém puder me ajudar, agradeço desde já.. Valeu
-- Otávio
Call-ID: OThkZGUyNGJmZTRhMDM0NGNjMzZlYjFhMzg2NGZmOGU. CSeq: 1 ACK Content-Length: 0 <-------------> --- (8 headers 0 lines) --- elastix*CLI> <--- SIP read from 192.168.1.10:5060 ---> INVITE sip:2...@192.168.1.11;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK-d8754z-5f7d3f5944ff5ef8-1---d8754z- Max-Forwards: 70 Contact: <sip:1...@192.168.1.10:5060;transport=UDP> To: <sip:2...@192.168.1.11;transport=UDP> From: "1000"<sip:1...@192.168.1.11;transport=UDP>;tag=f33fc456 Call-ID: OThkZGUyNGJmZTRhMDM0NGNjMzZlYjFhMzg2NGZmOGU. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp Proxy-Authorization: Digest username="1000",realm="asterisk",nonce="651a0cf6",uri="sip:2...@192.168.1.11;transport=UDP",response="9f6e298f3ab4d7c344dace2d8fc45f47",algorithm=MD5 User-Agent: Zoiper rev.5324 Content-Length: 327 v=0 o=Zoiper_user 0 0 IN IP4 192.168.1.10 s=Zoiper_session c=IN IP4 192.168.1.10 t=0 0 m=audio 8000 RTP/AVP 3 0 8 110 98 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:110 speex/8000 a=rtpmap:98 iLBC/8000 a=fmtp:98 mode=30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv <-------------> --- (13 headers 15 lines) --- Sending to 192.168.1.10 : 5060 (NAT) Using INVITE request as basis request - OThkZGUyNGJmZTRhMDM0NGNjMzZlYjFhMzg2NGZmOGU. Found user '1000' Found RTP audio format 3 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 110 Found RTP audio format 98 Found RTP audio format 101 Peer audio RTP is at port 192.168.1.10:8000 Found audio description format GSM for ID 3 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format speex for ID 110 Found audio description format iLBC for ID 98 Found audio description format telephone-event for ID 101 Capabilities: us - 0x4 (ulaw), peer - audio=0x60e (gsm|ulaw|alaw|speex|ilbc)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.1.10:8000 Looking for 2000 in contexto1000 (domain 192.168.1.11) list_route: hop: <sip:1...@192.168.1.10:5060;transport=UDP> <--- Transmitting (NAT) to 192.168.1.10:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK-d8754z-5f7d3f5944ff5ef8-1---d8754z-;received=192.168.1.10 From: "1000"<sip:1...@192.168.1.11;transport=UDP>;tag=f33fc456 To: <sip:2...@192.168.1.11;transport=UDP> Call-ID: OThkZGUyNGJmZTRhMDM0NGNjMzZlYjFhMzg2NGZmOGU. CSeq: 2 INVITE ser-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:2...@192.168.1.11> Content-Length: 0 <------------> -- Executing [2...@contexto1000:1] Dial("SIP/1000-0a0f9b78", "SIP/1001/2000") in new stack Audio is at 192.168.1.11 port 10140 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 192.168.1.12:5060: INVITE sip:2...@192.168.1.12 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK63684dce;rport From: "device" <sip:1...@192.168.1.11>;tag=as3f0010ce To: <sip:2...@192.168.1.12> Contact: <sip:1...@192.168.1.11> Call-ID: 5237928d4a0ebce93235ca1379ff3...@192.168.1.11 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 03 Feb 2010 07:47:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 238 elastix*CLI> v=0 o=root 7992 7992 IN IP4 192.168.1.11 s=session c=IN IP4 192.168.1.11 t=0 0 m=audio 10140 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- <--- SIP read from 192.168.1.12:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK63684dce;received=192.168.1.11;rport=5060 From: "device" <sip:1...@192.168.1.11>;tag=as3f0010ce To: <sip:2...@192.168.1.12>;tag=as17dc6a33 Call-ID: 5237928d4a0ebce93235ca1379ff3...@192.168.1.11 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4cc328e6" Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Transmitting (NAT) to 192.168.1.12:5060: ACK sip:2...@192.168.1.12 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK63684dce;rport From: "device" <sip:1...@192.168.1.11>;tag=as3f0010ce To: <sip:2...@192.168.1.12>;tag=as17dc6a33 Contact: <sip:1...@192.168.1.11> Call-ID: 5237928d4a0ebce93235ca1379ff3...@192.168.1.11 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Audio is at 192.168.1.11 port 10140 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 192.168.1.12:5060: INVITE sip:2...@192.168.1.12 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK73b01775;rport From: "device" <sip:1...@192.168.1.11>;tag=as3f0010ce To: <sip:2...@192.168.1.12> Contact: <sip:1...@192.168.1.11> Call-ID: 5237928d4a0ebce93235ca1379ff3...@192.168.1.11 CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Proxy-Authorization: Digest username="1001", realm="asterisk", algorithm=MD5, uri="sip:2...@192.168.1.12", nonce="4cc328e6", response="6b312b28dc6ff85285f167902b988b19" Date: Wed, 03 Feb 2010 07:47:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 238 v=0 o=root 7992 7993 IN IP4 192.168.1.11 s=session c=IN IP4 192.168.1.11 t=0 0 m=audio 10140 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called 1001/2000 <--- SIP read from 192.168.1.12:5060 ---> SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK73b01775;received=192.168.1.11;rport=5060 From: "device" <sip:1...@192.168.1.11>;tag=as3f0010ce To: <sip:2...@192.168.1.12>;tag=as17dc6a33 Call-ID: 5237928d4a0ebce93235ca1379ff3...@192.168.1.11 CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Transmitting (NAT) to 192.168.1.12:5060: ACK sip:2...@192.168.1.12 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK73b01775;rport From: "device" <sip:1...@192.168.1.11>;tag=as3f0010ce To: <sip:2...@192.168.1.12>;tag=as17dc6a33 Contact: <sip:1...@192.168.1.11> Call-ID: 5237928d4a0ebce93235ca1379ff3...@192.168.1.11 CSeq: 103 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/1001-0a1533c0 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing [2...@contexto1000:2] Hangup("SIP/1000-0a0f9b78", "") in new stack == Spawn extension (contexto1000, 2000, 2) exited non-zero on 'SIP/1000-0a0f9b78' Scheduling destruction of SIP dialog 'OThkZGUyNGJmZTRhMDM0NGNjMzZlYjFhMzg2NGZmOGU.' in 32000 ms (Method: INVITE) <--- Reliably Transmitting (NAT) to 192.168.1.10:5060 ---> SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK-d8754z-5f7d3f5944ff5ef8-1---d8754z-;received=192.168.1.10 From: "1000"<sip:1...@192.168.1.11;transport=UDP>;tag=f33fc456 To: <sip:2...@192.168.1.11;transport=UDP>;tag=as39b0c200 Call-ID: OThkZGUyNGJmZTRhMDM0NGNjMzZlYjFhMzg2NGZmOGU. CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> elastix*CLI> <--- SIP read from 192.168.1.10:5060 ---> ACK sip:2...@192.168.1.11;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK-d8754z-5f7d3f5944ff5ef8-1---d8754z- Max-Forwards: 70 To: <sip:2...@192.168.1.11;transport=UDP>;tag=as39b0c200 From: "1000"<sip:1...@192.168.1.11;transport=UDP>;tag=f33fc456 Call-ID: OThkZGUyNGJmZTRhMDM0NGNjMzZlYjFhMzg2NGZmOGU. CSeq: 2 ACK Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog '5237928d4a0ebce93235ca1379ff3...@192.168.1.11' Method: INVITE elastix*CLI>
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