<html><body><div style="font-family: times new roman, new york, times, serif; font-size: 12pt; color: #000000"><div>O Alcatel está fazendo SIP/302 ou SIP/301, devolvendo a chamada de volta pra você.<br></div><div>Como eu queria que o asterisk pudesse fazer isso..rs</div><div><br></div><div><br></div><div><span name="x"></span><div></div><p><span style="font-size: medium;"><span style="font-family: 'courier new', courier, monaco, monospace, sans-serif;" data-mce-style="font-family: 'courier new', courier, monaco, monospace, sans-serif;"><b>Atenciosamente,</b></span><span style="font-family: 'courier new', courier, monaco, monospace, sans-serif;" data-mce-style="font-family: 'courier new', courier, monaco, monospace, sans-serif;"><b><br></b></span></span></p><div><span style="font-size: small; font-family: 'courier new', courier, monaco, monospace, sans-serif;"><span style="font-size: x-small;"><b style="font-size: 12pt;">Neimar Lima de Ávila | Desenvolvimento/Soluções | </b></span><span style="color: rgb(51, 102, 255);" data-mce-style="color: #3366ff;"><b>Virtual Sistemas Ltda</b></span></span></div><div><span style="font-size: small; font-family: 'courier new', courier, monaco, monospace, sans-serif;">Rua Gonçalves Dias, 142 SL 704 - Funcionários - CEP:30.140-090 - Bhte/MG </span><br><span style="font-size: small; font-family: 'courier new', courier, monaco, monospace, sans-serif;">Tel: (031)32456213 - Ramal 2016 | Cel: (031)84952402(CLARO)</span><br><span style="font-size: small; font-family: 'courier new', courier, monaco, monospace, sans-serif;"><a href="http://www.virtualsistemas.com.br/" target="_blank" data-mce-href="http://www.virtualsistemas.com.br/"><span color="#00008b" style="color: rgb(0, 0, 139);" data-mce-style="color: #00008b;">www.virtualsistemas.com.br</span></a> | <b><a href="mailto:neimar@virtualsistemas.com.br" target="_blank" data-mce-href="mailto:neimar@virtualsistemas.com.br"><span color="#00008b" style="color: rgb(0, 0, 139);" data-mce-style="color: #00008b;">neimar@virtualsistemas.com.br</span></a></b></span></div><p></p><p><span style="font-size: small; font-family: 'courier new', courier, monaco, monospace, sans-serif;"><b>Preserve o Meio Ambiente! Pense Antes de Imprimir</b> </span><br><span style="font-size: small; font-family: 'courier new', courier, monaco, monospace, sans-serif;">Os dados transmitidos nesta mensagem destinam-se exclusivamente a(s) pessoa(s) mencionada(s) e contém informações confidenciais,</span><br><span style="font-size: small; font-family: 'courier new', courier, monaco, monospace, sans-serif;">legalmente protegidas, para conhecimento exclusivo do(s) destinatário(s).O exame, retransmissão, divulgação, leitura, cópia ou outro uso </span><br><span style="font-size: small; font-family: 'courier new', courier, monaco, monospace, sans-serif;">desta correspondencia, por pessoas, físicas ou jurídicas, que não o(s) destinatário(s), constituirá obtenção de dados por meio ilícito, </span><br><span style="font-size: small; font-family: 'courier new', courier, monaco, monospace, sans-serif;">configurando ofensa ao Art. 5°, inciso XII, da CF/88.</span></p><div></div><span name="x"></span><br></div><hr id="zwchr"><div style="color:#000;font-weight:normal;font-style:normal;text-decoration:none;font-family:Helvetica,Arial,sans-serif;font-size:12pt;"><b>De: </b>"Guilherme Rezende" <asterisk@guilherme.eti.br><br><b>Para: </b>asteriskbrasil@listas.asteriskbrasil.org<br><b>Enviadas: </b>Sexta-feira, 22 de Fevereiro de 2013 13:02:22<br><b>Assunto: </b>Re: [AsteriskBrasil] SIP Trunk Asterisk x Alcatel<br><div><br></div>Estou achando estranho que não existe parâmetro algum de <br>autenticação... Talvez dentro da conf da Alcatel seja necessário <br>habilitar o forward de sua rede, que no caso deve ser 172.16.1.0/24. <br>Acredito ser algo de permissão.<br><div><br></div>Em 22/02/2013 09:49, Jefferson B. Limeira escreveu:<br>> Bom dia,<br>><br>> Estamos participando da integração de um asterisk 1.6.2.11 com uma<br>> Alcatel via SIP. Quando ligo do asterisk para a Alcatel recebo um<br>> forwarding da chamada de volta para o asterisk. Segue maiores<br>> informações:<br>><br>> sip.conf:<br>><br>> [alcatel]<br>> host=172.16.1.3<br>> context=from-Alcatel<br>> type=friend<br>> nat=no<br>> disallow=all<br>> allow=alaw<br>><br>> extensions.conf<br>><br>> exten => _6X.,1,Dial(SIP/${EXTEN:1}@alcatel)<br>> same => n,HangUp<br>><br>> no console do asterisk durante a chamada<br>><br>> -- Executing [69202@saida:1] Dial("SIP/jefferson-00001a43",<br>> "SIP/9202@alcatel") in new stack<br>> == Using SIP RTP CoS mark 5<br>> -- Called 9202@alcatel<br>><br>> -- Now forwarding SIP/jefferson-00001a43 to<br>> 'Local/9202@from-Alcatel' (thanks to SIP/alcatel-00001a44)<br>><br>> [Feb 22 09:36:24] NOTICE[19305]: chan_local.c:538 local_call: No such<br>> extension/context 9202@from-Alcatel while calling Local channel<br>> [Feb 22 09:36:24] NOTICE[19305]: app_dial.c:789 do_forward: Failed to<br>> dial on local channel for call forward to '9202@from-Alcatel'<br>> == Everyone is busy/congested at this time (1:0/0/1)<br>> -- Executing [69202@saida:2] Hangup("SIP/jefferson-00001a43", "")<br>> in new stack<br>> == Spawn extension (TI, 69202, 2) exited non-zero on<br>> 'SIP/jefferson-00001a43'<br>><br>><br>> Segue sip debug deste peer<br>><br>> asterisk*CLI> sip set debug peer alcatel<br>> SIP Debugging Enabled for IP: 172.16.1.3:5060<br>> == Using SIP RTP CoS mark 5<br>> -- Executing [69202@TI:1] Dial("SIP/jefferson-00001a46",<br>> "SIP/9202@alcatel") in new stack<br>> == Using SIP RTP CoS mark 5<br>> Audio is at 172.16.200.92 port 5404<br>> Adding codec 0x8 (alaw) to SDP<br>> Adding non-codec 0x1 (telephone-event) to SDP<br>> Reliably Transmitting (no NAT) to 172.16.1.3:5060:<br>> INVITE sip:9202@172.16.1.3 SIP/2.0<br>> Via: SIP/2.0/UDP 172.16.200.92:5060;branch=z9hG4bK7b79bc0e;rport<br>> Max-Forwards: 70<br>> From: "jefferson"<sip:jefferson@172.16.200.92>;tag=as1b46e101<br>> To:<sip:9202@172.16.1.3><br>> Contact:<sip:jefferson@172.16.200.92><br>> Call-ID: 7f135fc326d306e80dc2bf4d6af36a20@172.16.200.92<br>> CSeq: 102 INVITE<br>> User-Agent: Asterisk PBX 1.6.2.11<br>> Date: Fri, 22 Feb 2013 12:38:28 GMT<br>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,<br>> INFO<br>> Supported: replaces, timer<br>> Content-Type: application/sdp<br>> Content-Length: 238<br>><br>> v=0<br>> o=root 303517300 303517300 IN IP4 172.16.200.92<br>> s=Asterisk PBX 1.6.2.11<br>> c=IN IP4 172.16.200.92<br>> t=0 0<br>> m=audio 5404 RTP/AVP 8 101<br>> a=rtpmap:8 PCMA/8000<br>> a=rtpmap:101 telephone-event/8000<br>> a=fmtp:101 0-16<br>> a=ptime:20<br>> a=sendrecv<br>> ---<br>> -- Called 9202@alcatel<br>><br>> <--- SIP read from UDP:172.16.1.3:5060 ---><br>> SIP/2.0 100 Trying<br>> To:<sip:9202@172.16.1.3><br>> From: "jefferson"<sip:jefferson@172.16.200.92>;tag=as1b46e101<br>> Call-ID: 7f135fc326d306e80dc2bf4d6af36a20@172.16.200.92<br>> CSeq: 102 INVITE<br>> Via: SIP/2.0/UDP<br>> 172.16.200.92:5060;received=172.16.200.92;branch=z9hG4bK7b79bc0e;rport=5060<br>> Content-Length: 0<br>> <-------------><br>> --- (7 headers 0 lines) ---<br>><br>> <--- SIP read from UDP:172.16.1.3:5060 ---><br>> SIP/2.0 100 Trying<br>> To:<sip:9202@172.16.1.3><br>> From: "jefferson"<sip:jefferson@172.16.200.92>;tag=as1b46e101<br>> Call-ID: 7f135fc326d306e80dc2bf4d6af36a20@172.16.200.92<br>> CSeq: 102 INVITE<br>> Via: SIP/2.0/UDP<br>> 172.16.200.92:5060;received=172.16.200.92;branch=z9hG4bK7b79bc0e;rport=5060<br>> Content-Length: 0<br>> <-------------><br>> --- (7 headers 0 lines) ---<br>><br>> <--- SIP read from UDP:172.16.1.3:5060 ---><br>> INVITE sip:9202@172.16.200.92:5060 SIP/2.0<br>> Record-Route:<sip:172.16.1.3;lr;transport=UDP><br>> Via: SIP/2.0/UDP<br>> 172.16.1.3:5060;branch=z9hG4bK467b772a9f9ef266e7d514eede2bf348cc48719d6f59fafe0690167af90cbb1d<br>> Via: SIP/2.0/UDP<br>> 172.16.200.92:5060;received=172.16.200.92;branch=z9hG4bK7b79bc0e;rport=5060<br>> Max-Forwards: 69<br>> From: "jefferson"<sip:jefferson@172.16.200.92>;tag=as1b46e101<br>> To:<sip:9202@172.16.1.3><br>> Contact:<sip:jefferson@172.16.1.3:5060><br>> Call-ID: 7f135fc326d306e80dc2bf4d6af36a20@172.16.200.92<br>> CSeq: 102 INVITE<br>> User-Agent: Asterisk PBX 1.6.2.11<br>> Date: Fri, 22 Feb 2013 12:38:28<br>> Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO<br>> Supported: replaces,timer<br>> Content-Type: application/sdp<br>> Content-Length: 236<br>> Session-Expires: 1800<br>><br>> v=0<br>> o=root 303517300 303517300 IN IP4 172.16.1.3<br>> s=Asterisk PBX 1.6.2.11<br>> c=IN IP4 172.16.1.3<br>> t=0 0<br>> m=audio 5404 RTP/AVP 8 101<br>> a=rtpmap:8 PCMA/8000<br>> a=rtpmap:101 telephone-event/8000<br>> a=fmtp:101 0-16<br>> a=ptime:20<br>> a=sendrecv<br>><br>> <-------------><br>> --- (17 headers 11 lines) ---<br>><br>> <--- Transmitting (no NAT) to 172.16.1.3:5060 ---><br>> SIP/2.0 100 Trying<br>> Via: SIP/2.0/UDP<br>> 172.16.1.3:5060;branch=z9hG4bK467b772a9f9ef266e7d514eede2bf348cc48719d6f59fafe0690167af90cbb1d;received=172.16.1.3<br>> Via: SIP/2.0/UDP<br>> 172.16.200.92:5060;received=172.16.200.92;branch=z9hG4bK7b79bc0e;rport=5060<br>> Record-Route:<sip:172.16.1.3;lr;transport=UDP><br>> From: "jefferson"<sip:jefferson@172.16.200.92>;tag=as1b46e101<br>> To:<sip:9202@172.16.1.3><br>> Call-ID: 7f135fc326d306e80dc2bf4d6af36a20@172.16.200.92<br>> CSeq: 102 INVITE<br>> Server: Asterisk PBX 1.6.2.11<br>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,<br>> INFO<br>> Supported: replaces, timer<br>> Contact:<sip:jefferson@172.16.200.92><br>> Content-Length: 0<br>><br>> <------------><br>> -- Now forwarding SIP/jefferson-00001a46 to<br>> 'Local/9202@from-Alcatel' (thanks to SIP/alcatel-00001a47)<br>> [Feb 22 09:38:28] NOTICE[19309]: chan_local.c:538 local_call: No such<br>> extension/context 9202@from-Alcatel while calling Local channel<br>> [Feb 22 09:38:28] NOTICE[19309]: app_dial.c:789 do_forward: Failed to<br>> dial on local channel for call forward to '9202@from-Alcatel'<br>> Scheduling destruction of SIP dialog<br>> '7f135fc326d306e80dc2bf4d6af36a20@172.16.200.92' in 32000 ms (Method:<br>> INVITE)<br>> Reliably Transmitting (no NAT) to 172.16.1.3:5060:<br>> CANCEL sip:9202@172.16.1.3 SIP/2.0<br>> Via: SIP/2.0/UDP 172.16.200.92:5060;branch=z9hG4bK7b79bc0e;rport<br>> Max-Forwards: 70<br>> From: "jefferson"<sip:jefferson@172.16.200.92>;tag=as1b46e101<br>> To:<sip:9202@172.16.1.3><br>> Call-ID: 7f135fc326d306e80dc2bf4d6af36a20@172.16.200.92<br>> CSeq: 102 CANCEL<br>> User-Agent: Asterisk PBX 1.6.2.11<br>> Content-Length: 0<br>><br>> ---<br>> Scheduling destruction of SIP dialog<br>> '7f135fc326d306e80dc2bf4d6af36a20@172.16.200.92' in 32000 ms (Method:<br>> INVITE)<br>> == Everyone is busy/congested at this time (1:0/0/1)<br>> -- Executing [69202@TI:2] Hangup("SIP/jefferson-00001a46", "") in<br>> new stack<br>> == Spawn extension (TI, 69202, 2) exited non-zero on<br>> 'SIP/jefferson-00001a46'<br>><br>> <--- SIP read from UDP:172.16.1.3:5060 ---><br>> SIP/2.0 200 OK<br>> To:<sip:9202@172.16.1.3><br>> From: "jefferson"<sip:jefferson@172.16.200.92>;tag=as1b46e101<br>> Call-ID: 7f135fc326d306e80dc2bf4d6af36a20@172.16.200.92<br>> CSeq: 102 CANCEL<br>> Via: SIP/2.0/UDP<br>> 172.16.200.92:5060;received=172.16.200.92;branch=z9hG4bK7b79bc0e;rport=5060<br>> Content-Length: 0<br>> <-------------><br>><br><div><br></div>_______________________________________________<br>EBS MODULAR: 3 slots para combinação entre E1, GSM, FXS ou FXO;<br>Linha de PORTEIROS IP, abrem até 2 dispositivos com acesso IP remoto;<br>Conheça esses e outros LANÇAMENTOS KHOMP em www.Khomp.com <br>_______________________________________________<br>DIGIVOICE Fabricante de Placas de Voz e Channel Bank<br>20 anos de experiência com E1(R2/ISDN), FXS, FXO e GSM<br>Centro Treinamento - Curso de PABX IP - Asterisk - Site www.digivoice.com.br<br>_______________________________________________<br>ALIGERA – Fabricante nacional de Gateways SIP-E1 para R2, ISDN e SS7.<br>Placas de 1E1, 2E1, 4E1 e 8E1 para PCI ou PCI Express.<br>Channel Bank – Appliance Asterisk - Acesse www.aligera.com.br.<br>_______________________________________________<br>Para remover seu email desta lista, basta enviar um email em branco para asteriskbrasil-unsubscribe@listas.asteriskbrasil.org<br></div><div><br></div></div></body></html>