Não há necessidade de autenticação, ambos os lados estão com IPs cadastrados e pertencem a uma rede considerada interna. O parametro host sendo diferente dynamic não exigem username/password no asterisk.
O grande lance é no pacote com a informação abaixo: INVITE sip:[email protected]:5060 SIP/2.0 Minha dúvida é de onde surgiu isso... Alguma sugestão? Em 2013-02-22 13:02, Guilherme Rezende escreveu: > Estou achando estranho que não existe parâmetro algum de > autenticação... Talvez dentro da conf da Alcatel seja necessário > habilitar o forward de sua rede, que no caso deve ser 172.16.1.0/24. > Acredito ser algo de permissão. > > Em 22/02/2013 09:49, Jefferson B. Limeira escreveu: >> Bom dia, >> >> Estamos participando da integração de um asterisk 1.6.2.11 com >> uma >> Alcatel via SIP. Quando ligo do asterisk para a Alcatel recebo um >> forwarding da chamada de volta para o asterisk. Segue maiores >> informações: >> >> sip.conf: >> >> [alcatel] >> host=172.16.1.3 >> context=from-Alcatel >> type=friend >> nat=no >> disallow=all >> allow=alaw >> >> extensions.conf >> >> exten => _6X.,1,Dial(SIP/${EXTEN:1}@alcatel) >> same => n,HangUp >> >> no console do asterisk durante a chamada >> >> -- Executing [69202@saida:1] Dial("SIP/jefferson-00001a43", >> "SIP/9202@alcatel") in new stack >> == Using SIP RTP CoS mark 5 >> -- Called 9202@alcatel >> >> -- Now forwarding SIP/jefferson-00001a43 to >> 'Local/9202@from-Alcatel' (thanks to SIP/alcatel-00001a44) >> >> [Feb 22 09:36:24] NOTICE[19305]: chan_local.c:538 local_call: No >> such >> extension/context 9202@from-Alcatel while calling Local channel >> [Feb 22 09:36:24] NOTICE[19305]: app_dial.c:789 do_forward: Failed >> to >> dial on local channel for call forward to '9202@from-Alcatel' >> == Everyone is busy/congested at this time (1:0/0/1) >> -- Executing [69202@saida:2] Hangup("SIP/jefferson-00001a43", >> "") >> in new stack >> == Spawn extension (TI, 69202, 2) exited non-zero on >> 'SIP/jefferson-00001a43' >> >> >> Segue sip debug deste peer >> >> asterisk*CLI> sip set debug peer alcatel >> SIP Debugging Enabled for IP: 172.16.1.3:5060 >> == Using SIP RTP CoS mark 5 >> -- Executing [69202@TI:1] Dial("SIP/jefferson-00001a46", >> "SIP/9202@alcatel") in new stack >> == Using SIP RTP CoS mark 5 >> Audio is at 172.16.200.92 port 5404 >> Adding codec 0x8 (alaw) to SDP >> Adding non-codec 0x1 (telephone-event) to SDP >> Reliably Transmitting (no NAT) to 172.16.1.3:5060: >> INVITE sip:[email protected] SIP/2.0 >> Via: SIP/2.0/UDP 172.16.200.92:5060;branch=z9hG4bK7b79bc0e;rport >> Max-Forwards: 70 >> From: "jefferson"<sip:[email protected]>;tag=as1b46e101 >> To:<sip:[email protected]> >> Contact:<sip:[email protected]> >> Call-ID: [email protected] >> CSeq: 102 INVITE >> User-Agent: Asterisk PBX 1.6.2.11 >> Date: Fri, 22 Feb 2013 12:38:28 GMT >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, >> INFO >> Supported: replaces, timer >> Content-Type: application/sdp >> Content-Length: 238 >> >> v=0 >> o=root 303517300 303517300 IN IP4 172.16.200.92 >> s=Asterisk PBX 1.6.2.11 >> c=IN IP4 172.16.200.92 >> t=0 0 >> m=audio 5404 RTP/AVP 8 101 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> a=ptime:20 >> a=sendrecv >> --- >> -- Called 9202@alcatel >> >> <--- SIP read from UDP:172.16.1.3:5060 ---> >> SIP/2.0 100 Trying >> To:<sip:[email protected]> >> From: "jefferson"<sip:[email protected]>;tag=as1b46e101 >> Call-ID: [email protected] >> CSeq: 102 INVITE >> Via: SIP/2.0/UDP >> >> 172.16.200.92:5060;received=172.16.200.92;branch=z9hG4bK7b79bc0e;rport=5060 >> Content-Length: 0 >> <-------------> >> --- (7 headers 0 lines) --- >> >> <--- SIP read from UDP:172.16.1.3:5060 ---> >> SIP/2.0 100 Trying >> To:<sip:[email protected]> >> From: "jefferson"<sip:[email protected]>;tag=as1b46e101 >> Call-ID: [email protected] >> CSeq: 102 INVITE >> Via: SIP/2.0/UDP >> >> 172.16.200.92:5060;received=172.16.200.92;branch=z9hG4bK7b79bc0e;rport=5060 >> Content-Length: 0 >> <-------------> >> --- (7 headers 0 lines) --- >> >> <--- SIP read from UDP:172.16.1.3:5060 ---> >> INVITE sip:[email protected]:5060 SIP/2.0 >> Record-Route:<sip:172.16.1.3;lr;transport=UDP> >> Via: SIP/2.0/UDP >> >> 172.16.1.3:5060;branch=z9hG4bK467b772a9f9ef266e7d514eede2bf348cc48719d6f59fafe0690167af90cbb1d >> Via: SIP/2.0/UDP >> >> 172.16.200.92:5060;received=172.16.200.92;branch=z9hG4bK7b79bc0e;rport=5060 >> Max-Forwards: 69 >> From: "jefferson"<sip:[email protected]>;tag=as1b46e101 >> To:<sip:[email protected]> >> Contact:<sip:[email protected]:5060> >> Call-ID: [email protected] >> CSeq: 102 INVITE >> User-Agent: Asterisk PBX 1.6.2.11 >> Date: Fri, 22 Feb 2013 12:38:28 >> Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO >> Supported: replaces,timer >> Content-Type: application/sdp >> Content-Length: 236 >> Session-Expires: 1800 >> >> v=0 >> o=root 303517300 303517300 IN IP4 172.16.1.3 >> s=Asterisk PBX 1.6.2.11 >> c=IN IP4 172.16.1.3 >> t=0 0 >> m=audio 5404 RTP/AVP 8 101 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> a=ptime:20 >> a=sendrecv >> >> <-------------> >> --- (17 headers 11 lines) --- >> >> <--- Transmitting (no NAT) to 172.16.1.3:5060 ---> >> SIP/2.0 100 Trying >> Via: SIP/2.0/UDP >> >> 172.16.1.3:5060;branch=z9hG4bK467b772a9f9ef266e7d514eede2bf348cc48719d6f59fafe0690167af90cbb1d;received=172.16.1.3 >> Via: SIP/2.0/UDP >> >> 172.16.200.92:5060;received=172.16.200.92;branch=z9hG4bK7b79bc0e;rport=5060 >> Record-Route:<sip:172.16.1.3;lr;transport=UDP> >> From: "jefferson"<sip:[email protected]>;tag=as1b46e101 >> To:<sip:[email protected]> >> Call-ID: [email protected] >> CSeq: 102 INVITE >> Server: Asterisk PBX 1.6.2.11 >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, >> INFO >> Supported: replaces, timer >> Contact:<sip:[email protected]> >> Content-Length: 0 >> >> <------------> >> -- Now forwarding SIP/jefferson-00001a46 to >> 'Local/9202@from-Alcatel' (thanks to SIP/alcatel-00001a47) >> [Feb 22 09:38:28] NOTICE[19309]: chan_local.c:538 local_call: No >> such >> extension/context 9202@from-Alcatel while calling Local channel >> [Feb 22 09:38:28] NOTICE[19309]: app_dial.c:789 do_forward: Failed >> to >> dial on local channel for call forward to '9202@from-Alcatel' >> Scheduling destruction of SIP dialog >> '[email protected]' in 32000 ms >> (Method: >> INVITE) >> Reliably Transmitting (no NAT) to 172.16.1.3:5060: >> CANCEL sip:[email protected] SIP/2.0 >> Via: SIP/2.0/UDP 172.16.200.92:5060;branch=z9hG4bK7b79bc0e;rport >> Max-Forwards: 70 >> From: "jefferson"<sip:[email protected]>;tag=as1b46e101 >> To:<sip:[email protected]> >> Call-ID: [email protected] >> CSeq: 102 CANCEL >> User-Agent: Asterisk PBX 1.6.2.11 >> Content-Length: 0 >> >> --- >> Scheduling destruction of SIP dialog >> '[email protected]' in 32000 ms >> (Method: >> INVITE) >> == Everyone is busy/congested at this time (1:0/0/1) >> -- Executing [69202@TI:2] Hangup("SIP/jefferson-00001a46", "") >> in >> new stack >> == Spawn extension (TI, 69202, 2) exited non-zero on >> 'SIP/jefferson-00001a46' >> >> <--- SIP read from UDP:172.16.1.3:5060 ---> >> SIP/2.0 200 OK >> To:<sip:[email protected]> >> From: "jefferson"<sip:[email protected]>;tag=as1b46e101 >> Call-ID: [email protected] >> CSeq: 102 CANCEL >> Via: SIP/2.0/UDP >> >> 172.16.200.92:5060;received=172.16.200.92;branch=z9hG4bK7b79bc0e;rport=5060 >> Content-Length: 0 >> <-------------> >> > -- []'s Jefferson B. Limeira [email protected] (41) 9928-8628 _______________________________________________ EBS MODULAR: 3 slots para combinação entre E1, GSM, FXS ou FXO; Linha de PORTEIROS IP, abrem até 2 dispositivos com acesso IP remoto; Conheça esses e outros LANÇAMENTOS KHOMP em www.Khomp.com _______________________________________________ DIGIVOICE Fabricante de Placas de Voz e Channel Bank 20 anos de experiência com E1(R2/ISDN), FXS, FXO e GSM Centro Treinamento - Curso de PABX IP - Asterisk - Site www.digivoice.com.br _______________________________________________ ALIGERA Fabricante nacional de Gateways SIP-E1 para R2, ISDN e SS7. Placas de 1E1, 2E1, 4E1 e 8E1 para PCI ou PCI Express. Channel Bank Appliance Asterisk - Acesse www.aligera.com.br. _______________________________________________ Para remover seu email desta lista, basta enviar um email em branco para [email protected]
