Oi Jefferson.<br><br>Já resolveu? O prefixo que está sendo utilizado para discar é "6" ou "69" ? <br><br>Dê uma olhada no padrão, um dígito está sendo ignorado:<br><br>_6X.,1,Dial(SIP/${EXTEN:<b><font size="4">1</font></b>}@alcatel)<br>
<br>Se discou 69202, o "6" será desconsiderado e a extensão considerada será 9202.<br><br><div class="gmail_quote">2013/2/22 Jefferson B. Limeira <span dir="ltr"><<a href="mailto:jbl@internexxus.com.br" target="_blank">jbl@internexxus.com.br</a>></span><br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Não há necessidade de autenticação, ambos os lados estão com IPs<br>
cadastrados e pertencem a uma rede considerada interna.<br>
O parametro host sendo diferente dynamic não exigem username/password<br>
no asterisk.<br>
<br>
O grande lance é no pacote com a informação abaixo:<br>
<div class="im"><br>
INVITE <a href="http://sip:9202@172.16.200.92:5060" target="_blank">sip:9202@172.16.200.92:5060</a> SIP/2.0<br>
<br>
</div>Minha dúvida é de onde surgiu isso... Alguma sugestão?<br>
<br>
Em 2013-02-22 13:02, Guilherme Rezende escreveu:<br>
<div class="HOEnZb"><div class="h5">> Estou achando estranho que não existe parâmetro algum de<br>
> autenticação... Talvez dentro da conf da Alcatel seja necessário<br>
> habilitar o forward de sua rede, que no caso deve ser <a href="http://172.16.1.0/24" target="_blank">172.16.1.0/24</a>.<br>
> Acredito ser algo de permissão.<br>
><br>
> Em 22/02/2013 09:49, Jefferson B. Limeira escreveu:<br>
>> Bom dia,<br>
>><br>
>> Estamos participando da integração de um asterisk 1.6.2.11 com<br>
>> uma<br>
>> Alcatel via SIP. Quando ligo do asterisk para a Alcatel recebo um<br>
>> forwarding da chamada de volta para o asterisk. Segue maiores<br>
>> informações:<br>
>><br>
>> sip.conf:<br>
>><br>
>> [alcatel]<br>
>> host=172.16.1.3<br>
>> context=from-Alcatel<br>
>> type=friend<br>
>> nat=no<br>
>> disallow=all<br>
>> allow=alaw<br>
>><br>
>> extensions.conf<br>
>><br>
>> exten => _6X.,1,Dial(SIP/${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<br>
>> 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<br>
>> 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>
>> "")<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: <a href="http://172.16.1.3:5060" target="_blank">172.16.1.3:5060</a><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 <a href="tel:172.16.200.92" value="+551721620092">172.16.200.92</a> port 5404<br>
>> Adding codec 0x8 (alaw) to SDP<br>
>> Adding non-codec 0x1 (telephone-event) to SDP<br>
>> Reliably Transmitting (no NAT) to <a href="http://172.16.1.3:5060" target="_blank">172.16.1.3:5060</a>:<br>
>> INVITE <a href="mailto:sip%3A9202@172.16.1.3">sip:9202@172.16.1.3</a> SIP/2.0<br>
>> Via: SIP/2.0/UDP <a href="tel:172.16.200.92" value="+551721620092">172.16.200.92</a>:5060;branch=z9hG4bK7b79bc0e;rport<br>
>> Max-Forwards: 70<br>
>> From: "jefferson"<<a href="mailto:sip%3Ajefferson@172.16.200.92">sip:jefferson@172.16.200.92</a>>;tag=as1b46e101<br>
>> To:<<a href="mailto:sip%3A9202@172.16.1.3">sip:9202@172.16.1.3</a>><br>
>> Contact:<<a href="mailto:sip%3Ajefferson@172.16.200.92">sip:jefferson@172.16.200.92</a>><br>
>> Call-ID: <a href="mailto:7f135fc326d306e80dc2bf4d6af36a20@172.16.200.92">7f135fc326d306e80dc2bf4d6af36a20@172.16.200.92</a><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:<a href="http://172.16.1.3:5060" target="_blank">172.16.1.3:5060</a> ---><br>
>> SIP/2.0 100 Trying<br>
>> To:<<a href="mailto:sip%3A9202@172.16.1.3">sip:9202@172.16.1.3</a>><br>
>> From: "jefferson"<<a href="mailto:sip%3Ajefferson@172.16.200.92">sip:jefferson@172.16.200.92</a>>;tag=as1b46e101<br>
>> Call-ID: <a href="mailto:7f135fc326d306e80dc2bf4d6af36a20@172.16.200.92">7f135fc326d306e80dc2bf4d6af36a20@172.16.200.92</a><br>
>> CSeq: 102 INVITE<br>
>> Via: SIP/2.0/UDP<br>
>><br>
>> <a href="tel:172.16.200.92" value="+551721620092">172.16.200.92</a>:5060;received=<a href="tel:172.16.200.92" value="+551721620092">172.16.200.92</a>;branch=z9hG4bK7b79bc0e;rport=5060<br>
>> Content-Length: 0<br>
>> <-------------><br>
>> --- (7 headers 0 lines) ---<br>
>><br>
>> <--- SIP read from UDP:<a href="http://172.16.1.3:5060" target="_blank">172.16.1.3:5060</a> ---><br>
>> SIP/2.0 100 Trying<br>
>> To:<<a href="mailto:sip%3A9202@172.16.1.3">sip:9202@172.16.1.3</a>><br>
>> From: "jefferson"<<a href="mailto:sip%3Ajefferson@172.16.200.92">sip:jefferson@172.16.200.92</a>>;tag=as1b46e101<br>
>> Call-ID: <a href="mailto:7f135fc326d306e80dc2bf4d6af36a20@172.16.200.92">7f135fc326d306e80dc2bf4d6af36a20@172.16.200.92</a><br>
>> CSeq: 102 INVITE<br>
>> Via: SIP/2.0/UDP<br>
>><br>
>> <a href="tel:172.16.200.92" value="+551721620092">172.16.200.92</a>:5060;received=<a href="tel:172.16.200.92" value="+551721620092">172.16.200.92</a>;branch=z9hG4bK7b79bc0e;rport=5060<br>
>> Content-Length: 0<br>
>> <-------------><br>
>> --- (7 headers 0 lines) ---<br>
>><br>
>> <--- SIP read from UDP:<a href="http://172.16.1.3:5060" target="_blank">172.16.1.3:5060</a> ---><br>
>> INVITE <a href="http://sip:9202@172.16.200.92:5060" target="_blank">sip:9202@172.16.200.92:5060</a> SIP/2.0<br>
>> Record-Route:<sip:172.16.1.3;lr;transport=UDP><br>
>> Via: SIP/2.0/UDP<br>
>><br>
>> 172.16.1.3:5060;branch=z9hG4bK467b772a9f9ef266e7d514eede2bf348cc48719d6f59fafe0690167af90cbb1d<br>
>> Via: SIP/2.0/UDP<br>
>><br>
>> 172.16.200.92:5060;received=<a href="tel:172.16.200.92" value="+551721620092">172.16.200.92</a>;branch=z9hG4bK7b79bc0e;rport=5060<br>
>> Max-Forwards: 69<br>
>> From: "jefferson"<<a href="mailto:sip%3Ajefferson@172.16.200.92">sip:jefferson@172.16.200.92</a>>;tag=as1b46e101<br>
>> To:<<a href="mailto:sip%3A9202@172.16.1.3">sip:9202@172.16.1.3</a>><br>
>> Contact:<<a href="http://sip:jefferson@172.16.1.3:5060" target="_blank">sip:jefferson@172.16.1.3:5060</a>><br>
>> Call-ID: <a href="mailto:7f135fc326d306e80dc2bf4d6af36a20@172.16.200.92">7f135fc326d306e80dc2bf4d6af36a20@172.16.200.92</a><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 <a href="http://172.16.1.3:5060" target="_blank">172.16.1.3:5060</a> ---><br>
>> SIP/2.0 100 Trying<br>
>> Via: SIP/2.0/UDP<br>
>><br>
>> 172.16.1.3:5060;branch=z9hG4bK467b772a9f9ef266e7d514eede2bf348cc48719d6f59fafe0690167af90cbb1d;received=172.16.1.3<br>
>> Via: SIP/2.0/UDP<br>
>><br>
>> 172.16.200.92:5060;received=<a href="tel:172.16.200.92" value="+551721620092">172.16.200.92</a>;branch=z9hG4bK7b79bc0e;rport=5060<br>
>> Record-Route:<sip:172.16.1.3;lr;transport=UDP><br>
>> From: "jefferson"<<a href="mailto:sip%3Ajefferson@172.16.200.92">sip:jefferson@172.16.200.92</a>>;tag=as1b46e101<br>
>> To:<<a href="mailto:sip%3A9202@172.16.1.3">sip:9202@172.16.1.3</a>><br>
>> Call-ID: <a href="mailto:7f135fc326d306e80dc2bf4d6af36a20@172.16.200.92">7f135fc326d306e80dc2bf4d6af36a20@172.16.200.92</a><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:<<a href="mailto:sip%3Ajefferson@172.16.200.92">sip:jefferson@172.16.200.92</a>><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<br>
>> 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<br>
>> to<br>
>> dial on local channel for call forward to '9202@from-Alcatel'<br>
>> Scheduling destruction of SIP dialog<br>
>> '<a href="mailto:7f135fc326d306e80dc2bf4d6af36a20@172.16.200.92">7f135fc326d306e80dc2bf4d6af36a20@172.16.200.92</a>' in 32000 ms<br>
>> (Method:<br>
>> INVITE)<br>
>> Reliably Transmitting (no NAT) to <a href="http://172.16.1.3:5060" target="_blank">172.16.1.3:5060</a>:<br>
>> CANCEL <a href="mailto:sip%3A9202@172.16.1.3">sip:9202@172.16.1.3</a> SIP/2.0<br>
>> Via: SIP/2.0/UDP <a href="tel:172.16.200.92" value="+551721620092">172.16.200.92</a>:5060;branch=z9hG4bK7b79bc0e;rport<br>
>> Max-Forwards: 70<br>
>> From: "jefferson"<<a href="mailto:sip%3Ajefferson@172.16.200.92">sip:jefferson@172.16.200.92</a>>;tag=as1b46e101<br>
>> To:<<a href="mailto:sip%3A9202@172.16.1.3">sip:9202@172.16.1.3</a>><br>
>> Call-ID: <a href="mailto:7f135fc326d306e80dc2bf4d6af36a20@172.16.200.92">7f135fc326d306e80dc2bf4d6af36a20@172.16.200.92</a><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>
>> '<a href="mailto:7f135fc326d306e80dc2bf4d6af36a20@172.16.200.92">7f135fc326d306e80dc2bf4d6af36a20@172.16.200.92</a>' in 32000 ms<br>
>> (Method:<br>
>> INVITE)<br>
>> == Everyone is busy/congested at this time (1:0/0/1)<br>
>> -- Executing [69202@TI:2] Hangup("SIP/jefferson-00001a46", "")<br>
>> in<br>
>> new stack<br>
>> == Spawn extension (TI, 69202, 2) exited non-zero on<br>
>> 'SIP/jefferson-00001a46'<br>
>><br>
>> <--- SIP read from UDP:<a href="http://172.16.1.3:5060" target="_blank">172.16.1.3:5060</a> ---><br>
>> SIP/2.0 200 OK<br>
>> To:<<a href="mailto:sip%3A9202@172.16.1.3">sip:9202@172.16.1.3</a>><br>
>> From: "jefferson"<<a href="mailto:sip%3Ajefferson@172.16.200.92">sip:jefferson@172.16.200.92</a>>;tag=as1b46e101<br>
>> Call-ID: <a href="mailto:7f135fc326d306e80dc2bf4d6af36a20@172.16.200.92">7f135fc326d306e80dc2bf4d6af36a20@172.16.200.92</a><br>
>> CSeq: 102 CANCEL<br>
>> Via: SIP/2.0/UDP<br>
>><br>
>> <a href="tel:172.16.200.92" value="+551721620092">172.16.200.92</a>:5060;received=<a href="tel:172.16.200.92" value="+551721620092">172.16.200.92</a>;branch=z9hG4bK7b79bc0e;rport=5060<br>
>> Content-Length: 0<br>
>> <-------------><br>
>><br>
><br>
<br>
</div></div><div class="im HOEnZb">--<br>
[]'s Jefferson B. Limeira<br>
<a href="mailto:jbl@internexxus.com.br">jbl@internexxus.com.br</a><br>
<a href="tel:%2841%29%209928-8628" value="+554199288628">(41) 9928-8628</a><br>
</div><div class="HOEnZb"><div class="h5">_______________________________________________<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 <a href="http://www.Khomp.com" target="_blank">www.Khomp.com</a> <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 <a href="http://www.digivoice.com.br" target="_blank">www.digivoice.com.br</a><br>
_______________________________________________<br>
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</div></div></blockquote></div><br><br clear="all"><br>-- <br>Jairo Molina Jr∴<br>
<a href="http://www.intermol.com.br/" target="_blank">http://www.intermol.com.br</a><font face="arial, helvetica, sans-serif"><span style="border-collapse:collapse"><a value="+551183288940"><span dir="ltr"></span></a></span></font>