Vinícius, obrigado pela resposta, mas no meu caso a opcao "r" já esta no Dial, abaixo segue o log gerado na CLI do asterisk, no teste que fiz, eu liguei do ramal 2004 para o ramal 2010, atendi no ramal 2010, e transferi a ligacao para o ramal 2001, ai no momento que eu desliguei o ramal 2010, o ramal 2001 continua chamando, mas para o ramal 2004 fica mudo. Fiz um teste tb colocando a opcao "m" no Dial, ai desta maneira fica tocando a musica de espera, mas ai qdo eu disco para um ramal interno tb toca a musica de espera, e o pessoal da empresa nao quer isso.
abaixo o log da CLI -- Executing Dial("SIP/2004-b153b6b8", "SIP/2010|120|tTwWr") in new stack -- Called 2010 -- SIP/2010-0950f120 is ringing -- SIP/2010-0950f120 answered SIP/2004-b153b6b8 -- Started music on hold, class 'default', on SIP/2004-b153b6b8 -- Playing 'pbx-transfer' (language 'pt_BR') -- Executing Macro("Local/2...@from-internal-xfer-33d6,2", "exten-vm|2001|2001") in new stack -- Executing Macro("Local/2...@from-internal-xfer-33d6,2", "user-callerid") in new stack -- Executing NoOp("Local/2...@from-internal-xfer-33d6,2", "user-callerid: 2010") in new stack -- Executing Set("Local/2...@from-internal-xfer-33d6,2", "AMPUSER=2010") in new stack -- Executing GotoIf("Local/2...@from-internal-xfer-33d6,2", "1?report") in new stack -- Goto (macro-user-callerid,s,13) -- Executing NoOp("Local/2...@from-internal-xfer-33d6,2", "TTL: 64 ARG1: 2001") in new stack -- Executing GotoIf("Local/2...@from-internal-xfer-33d6,2", "0?continue") in new stack -- Executing Set("Local/2...@from-internal-xfer-33d6,2", "__TTL=63") in new stack -- Executing GotoIf("Local/2...@from-internal-xfer-33d6,2", "1?continue") in new stack -- Goto (macro-user-callerid,s,23) -- Executing NoOp("Local/2...@from-internal-xfer-33d6,2", "Using CallerID "" <2010>") in new stack -- Executing Set("Local/2...@from-internal-xfer-33d6,2", "FROMCONTEXT=exten-vm") in new stack -- Executing Set("Local/2...@from-internal-xfer-33d6,2", "VMBOX=2001") in new stack -- Executing Set("Local/2...@from-internal-xfer-33d6,2", "EXTTOCALL=2001") in new stack -- Executing Set("Local/2...@from-internal-xfer-33d6,2", "CFUEXT=") in new stack -- Executing Set("Local/2...@from-internal-xfer-33d6,2", "CFBEXT=") in new stack -- Executing Set("Local/2...@from-internal-xfer-33d6,2", "RT=120") in new stack -- Executing Macro("Local/2...@from-internal-xfer-33d6,2", "record-enable|2001|IN") in new stack -- Executing GotoIf("Local/2...@from-internal-xfer-33d6,2", "0?2:4") in new stack -- Goto (macro-record-enable,s,4) -- Executing AGI("Local/2...@from-internal-xfer-33d6,2", "recordingcheck|20090630-135700|1246381020.7224") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck recordingcheck|20090630-135700|1246381020.7224: Inbound recording enabled. recordingcheck|20090630-135700|1246381020.7224: CALLFILENAME=20090630-135700-1246381020.7224 -- AGI Script recordingcheck completed, returning 0 -- Executing MixMonitor("Local/2...@from-internal-xfer-33d6,2", "20090630-135700-1246381020.7224.wav") in new stack -- Executing Macro("Local/2...@from-internal-xfer-33d6,2", "dial|120|tTwWr|2001") in new stack -- Executing GotoIf("Local/2...@from-internal-xfer-33d6,2", "1?dial") in new stack -- Goto (macro-dial,s,3) -- Executing AGI("Local/2...@from-internal-xfer-33d6,2", "dialparties.agi") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi == Begin MixMonitor Recording Local/2...@from-internal-xfer-33d6,2 dialparties.agi: Starting New Dialparties.agi == Parsing '/etc/asterisk/manager.conf': Found == Parsing '/etc/asterisk/manager_custom.conf': Found == Manager 'admin' logged on from 127.0.0.1 dialparties.agi: Caller ID name is 'unknown' number is '2010' dialparties.agi: Methodology of ring is 'none' -- dialparties.agi: Added extension 2001 to extension map -- dialparties.agi: Extension 2001 cf is disabled -- dialparties.agi: Extension 2001 do not disturb is disabled dialparties.agi: Extension 2001 has ExtensionState: 0 -- dialparties.agi: Checking CW and CFB status for extension 2001 -- dialparties.agi: dbset CALLTRACE/2001 to 2010 == Manager 'admin' logged off from 127.0.0.1 -- AGI Script dialparties.agi completed, returning 0 -- Executing Dial("Local/2...@from-internal-xfer-33d6,2", "SIP/2001|120|tTwWr") in new stack -- Called 2001 -- Local/2...@from-internal-xfer-33d6,1 is ringing -- SIP/2001-0952a830 is ringing -- Stopped music on hold on SIP/2004-b153b6b8 -- Playing 'beep' (language 'pt_BR') == Spawn extension (macro-dial, s, 10) exited non-zero on 'Transfered/SIP/2004-b153b6b8<ZOMBIE>' in macro 'dial' == Spawn extension (macro-dial, s, 10) exited non-zero on 'Transfered/SIP/2004-b153b6b8<ZOMBIE>' in macro 'exten-vm' == Spawn extension (macro-dial, s, 10) exited non-zero on 'Transfered/SIP/2004-b153b6b8<ZOMBIE>' -- Executing Macro("Transfered/SIP/2004-b153b6b8<ZOMBIE>", "hangupcall") in new stack -- Executing ResetCDR("Transfered/SIP/2004-b153b6b8<ZOMBIE>", "w") in new stack -- Executing NoCDR("Transfered/SIP/2004-b153b6b8<ZOMBIE>", "") in new stack Jun 30 13:57:08 NOTICE[22488]: cdr.c:443 ast_cdr_free: CDR on channel 'SIP/2004-b153b6b8' not posted Jun 30 13:57:08 NOTICE[22488]: cdr.c:445 ast_cdr_free: CDR on channel 'SIP/2004-b153b6b8' lacks end -- Executing GotoIf("Transfered/SIP/2004-b153b6b8<ZOMBIE>", "1?skiprg") in new stack -- Goto (macro-hangupcall,s,6) -- Executing GotoIf("Transfered/SIP/2004-b153b6b8<ZOMBIE>", "1?skipblkvm") in new stack -- Goto (macro-hangupcall,s,9) -- Executing GotoIf("Transfered/SIP/2004-b153b6b8<ZOMBIE>", "1?theend") in new stack -- Goto (macro-hangupcall,s,11) -- Executing Hangup("Transfered/SIP/2004-b153b6b8<ZOMBIE>", "") in new stack == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'Transfered/SIP/2004-b153b6b8<ZOMBIE>' in macro 'hangupcall' == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'Transfered/SIP/2004-b153b6b8<ZOMBIE>' == Parsing '/etc/asterisk/manager.conf': Found == Parsing '/etc/asterisk/manager_custom.conf': Found == Manager 'admin' logged on from 127.0.0.1 == Manager 'admin' logged off from 127.0.0.1 == Parsing '/etc/asterisk/manager.conf': Found == Parsing '/etc/asterisk/manager_custom.conf': Found Atenciosamente, Rosauro Baretta Vinícius Fontes escreveu: > A música de espera só é reproduzida durante a consulta ao ramal. Após ouve-se > o ringback. > > Como no seu caso o ringback não está sendo reproduzido, utilize a opção r na > application Dial(). > > > > > Vinícius Fontes > www.asteriskforum.com.br - Informações e discussão sobre Asterisk e telefonia > IP > > ----- "Rosauro - RLINE" <rosa...@rline.com.br> escreveu: > > >> Pessoal, >> >> estou tendo problemas com a transferencia assistida, o que me acontece >> é >> o seguinte. Quando inicio a transferência assistida com o *2 (uso o >> meucci) a pessoal com a qual eu estou conversando fica ouvindo a >> música >> de espera, mas ai qdo o ramal de destino começa a chamar, e eu desligo >> o >> meu telefone, ai para de tocar a musica de espera, para a pessoa que >> eu >> estava falando e nem da o tom de chamada, e o telefone fica mudo para >> >> ela, mas o ramal de destino (o qual eu transferi a ligacao) continua >> chamando, ai se a pessoa atender a ligacao, a ligacao é transferida >> normalmente, mas ocorre que a pessoa que ligou acaba achando que a >> ligacao caiu pq ficou muda e desliga o telefone. >> >> alguém poderia me dar uma dica de como eu poderia resolver este meu >> problema ? >> >> Atenciosamente, >> >> Rosauro Baretta >> _______________________________________________ >> Openmoko Freerunner, primeiro telefone open source, disponível no >> Brasil rodando o Android da Google. >> http://www.neodroid.com >> >> Compre uma camiseta da AsteriskBrasil.org! >> http://www.voipmania.com.br >> >> Acesse o canal IRC de discussão sobre Asterisk em Português Brasileiro >> na rede Freenode.net: #asterisk-br >> _______________________________________________ >> Lista de discussões AsteriskBrasil.org >> AsteriskBrasil@listas.asteriskbrasil.org >> http://listas.asteriskbrasil.org/mailman/listinfo/asteriskbrasil >> > _______________________________________________ > Openmoko Freerunner, primeiro telefone open source, disponível no Brasil > rodando o Android da Google. > http://www.neodroid.com > > Compre uma camiseta da AsteriskBrasil.org! > http://www.voipmania.com.br > > Acesse o canal IRC de discussão sobre Asterisk em Português Brasileiro na > rede Freenode.net: #asterisk-br > _______________________________________________ > Lista de discussões AsteriskBrasil.org > AsteriskBrasil@listas.asteriskbrasil.org > http://listas.asteriskbrasil.org/mailman/listinfo/asteriskbrasil > > _______________________________________________ Openmoko Freerunner, primeiro telefone open source, disponível no Brasil rodando o Android da Google. http://www.neodroid.com Compre uma camiseta da AsteriskBrasil.org! http://www.voipmania.com.br Acesse o canal IRC de discussão sobre Asterisk em Português Brasileiro na rede Freenode.net: #asterisk-br _______________________________________________ Lista de discussões AsteriskBrasil.org AsteriskBrasil@listas.asteriskbrasil.org http://listas.asteriskbrasil.org/mailman/listinfo/asteriskbrasil