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:9202@172.16.1.3 SIP/2.0 > Via: SIP/2.0/UDP 172.16.200.92:5060;branch=z9hG4bK7b79bc0e;rport > Max-Forwards: 70 > From: "jefferson"<sip:jefferson@172.16.200.92>;tag=as1b46e101 > To:<sip:9202@172.16.1.3> > Contact:<sip:jefferson@172.16.200.92> > Call-ID: 7f135fc326d306e80dc2bf4d6af36a20@172.16.200.92 > 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:9202@172.16.1.3> > From: "jefferson"<sip:jefferson@172.16.200.92>;tag=as1b46e101 > Call-ID: 7f135fc326d306e80dc2bf4d6af36a20@172.16.200.92 > 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:9202@172.16.1.3> > From: "jefferson"<sip:jefferson@172.16.200.92>;tag=as1b46e101 > Call-ID: 7f135fc326d306e80dc2bf4d6af36a20@172.16.200.92 > 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:9202@172.16.200.92:5060 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:jefferson@172.16.200.92>;tag=as1b46e101 > To:<sip:9202@172.16.1.3> > Contact:<sip:jefferson@172.16.1.3:5060> > Call-ID: 7f135fc326d306e80dc2bf4d6af36a20@172.16.200.92 > 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:jefferson@172.16.200.92>;tag=as1b46e101 > To:<sip:9202@172.16.1.3> > Call-ID: 7f135fc326d306e80dc2bf4d6af36a20@172.16.200.92 > 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:jefferson@172.16.200.92> > 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 > '7f135fc326d306e80dc2bf4d6af36a20@172.16.200.92' in 32000 ms (Method: > INVITE) > Reliably Transmitting (no NAT) to 172.16.1.3:5060: > CANCEL sip:9202@172.16.1.3 SIP/2.0 > Via: SIP/2.0/UDP 172.16.200.92:5060;branch=z9hG4bK7b79bc0e;rport > Max-Forwards: 70 > From: "jefferson"<sip:jefferson@172.16.200.92>;tag=as1b46e101 > To:<sip:9202@172.16.1.3> > Call-ID: 7f135fc326d306e80dc2bf4d6af36a20@172.16.200.92 > CSeq: 102 CANCEL > User-Agent: Asterisk PBX 1.6.2.11 > Content-Length: 0 > > --- > Scheduling destruction of SIP dialog > '7f135fc326d306e80dc2bf4d6af36a20@172.16.200.92' 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:9202@172.16.1.3> > From: "jefferson"<sip:jefferson@172.16.200.92>;tag=as1b46e101 > Call-ID: 7f135fc326d306e80dc2bf4d6af36a20@172.16.200.92 > 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 > <-------------> > _______________________________________________ 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 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