Sugiro verificarem quais os codecs estão permitidos para os respectivos ramais e se os os mesmos estão habilitados nos clientes, ative o debug do sip na interface CLI, que ele irá informar o motivo da não conexão.
SDS José Leitão ----- Original Message ----- From: Frederico Simões To: asteriskbrasil em listas.asteriskbrasil.org Sent: Friday, December 23, 2005 1:54 PM Subject: Re: [AsteriskBrasil] Sip´s não falam entre si no asterisk em home Estou com um problema parecido... O meu fala de X-lite para X-lite... so que não fala de PAP para PAP!!! Se alguem poder solucionar [ext-local] include => ext-local-custom exten => 10001,1,Macro(exten-vm,novm,10001) exten => 10002,1,Macro(exten-vm,novm,10002) exten => 1001,1,Macro(exten-vm,1001 em default,1001) exten => ${VM_PREFIX}1001,1,Macro(vm,1001) exten => 1002,1,Macro(exten-vm,1002 em default,1002) exten => ${VM_PREFIX}1002,1,Macro(vm,1002) exten => 10021,1,Macro(exten-vm,10021 em default,10021) exten => ${VM_PREFIX}10021,1,Macro(vm,10021) exten => 10022,1,Macro(exten-vm,10022 em default,10022) exten => ${VM_PREFIX}10022,1,Macro(vm,10022) exten => 1003,1,Macro(exten-vm,1003 em default,1003) exten => ${VM_PREFIX}1003,1,Macro(vm,1003) exten => 1004,1,Macro(exten-vm,1004 em default,1004) exten => ${VM_PREFIX}1004,1,Macro(vm,1004) exten => 1005,1,Macro(exten-vm,1005 em default,1005) exten => ${VM_PREFIX}1005,1,Macro(vm,1005) exten => 11001,1,Macro(exten-vm,novm,11001) exten => 11002,1,Macro(exten-vm,novm,11002) exten => 12001,1,Macro(exten-vm,novm,12001) exten => 3001,1,Macro(exten-vm,novm,3001) exten => 3002,1,Macro(exten-vm,novm,3002) exten => 4001,1,Macro(exten-vm,novm,4001) exten => 4002,1,Macro(exten-vm,novm,4002) exten => 4011,1,Macro(exten-vm,novm,4011) exten => 4012,1,Macro(exten-vm,novm,4012) exten => 4021,1,Macro(exten-vm,novm,4021) [macro-exten-vm] exten => s,1,Macro(user-callerid) exten => s,2,Setvar(FROMCONTEXT=exten-vm) exten => s,3,Macro(record-enable,${ARG2},IN) exten => s,4,Macro(dial,${RINGTIMER},${DIAL_OPTIONS},${ARG2}) exten => s,5,GotoIf($[${CHANNEL:0:5} = Local]?s-${DIALSTATUS},1) ; if the channel is Local, then do not go to voicemail. This is primarily to avoid vm for call-forwarded extensions in ring groups exten => s,6,GotoIf($[${ARG1} = novm]?s-${DIALSTATUS},1) ; no voicemail in use for this extension exten => s,7,NoOp(Sending to Voicemail box ${ARG2}) exten => s,8,Macro(vm,${ARG1},${DIALSTATUS}) exten => s-BUSY,1,NoOp(Extension is reporting BUSY and has no Voicemail) exten => s-BUSY,2,Busy() exten => s-BUSY,3,Wait(60) exten => s-BUSY,4,NoOp() exten => _s-.,1,Congestion() Bom dia Asteriskers; Sempre trabalhei com o Asterisk instalado ´no braço´. Porém resolvi testar o asterisk em home e estou com um probleminha básico. Configurei dois ramais sip com o x-lite. Eles estão logando e o ecotest está ok. Porém quando tento falar de um sip (1234) para outro (2000) aparece a menságem no x-lite:( Call failed 486 Busy Here ). O ast em home está em uma rede interna juntamente com os outros softphones, exluindo-se a possibilidade de ser uma dificuldade em passar pelo NAT. Existe algo a mais que tem que ser feito, além de se configurar os ramais no ast em home para funcionar? Abaixo o log do que acontece no momento que ligo para outro sip: Desde já agradeço a atenção de todos. Willians Dias Vitória E.S. login as: root root em 192.168.30.125's password: Last login: Thu Jan 6 19:27:39 2005 from 192.168.30.161 Welcome to Asterisk em Home ------------------------------------------------- For access to the Asterisk em Home web GUI use this URL http://192.168.30.125 For help on Asterisk em Home commands you can use from this command shell type help-aah. [root em asterisk1 ~]# asterisk -r Asterisk 1.2.1, Copyright (C) 1999 - 2005 Digium. Written by Mark Spencer <markster em digium.com> ========================================================================= Connected to Asterisk 1.2.1 currently running on asterisk1 (pid = 2381) Verbosity is at least 3 -- Executing Macro("SIP/2000-7e61", "exten-vm|novm|1234") in new stack -- Executing Macro("SIP/2000-7e61", "user-callerid") in new stack -- Executing DBget("SIP/2000-7e61", "AMPUSER=DEVICE/2000/user") in new stack -- DBget: varname=AMPUSER, family=DEVICE, key=2000/user -- DBget: set variable AMPUSER to 2000 -- Executing DBget("SIP/2000-7e61", "AMPUSERCIDNAME=AMPUSER/2000/cidname") in new stack -- DBget: varname=AMPUSERCIDNAME, family=AMPUSER, key=2000/cidname -- DBget: set variable AMPUSERCIDNAME to Willians -- Executing GotoIf("SIP/2000-7e61", "0?5") in new stack -- Executing SetCallerID("SIP/2000-7e61", ""Willians" <2000>") in new stack -- Executing NoOp("SIP/2000-7e61", "Using CallerID "Willians" <2000>") in new stack -- Executing SetVar("SIP/2000-7e61", "FROMCONTEXT=exten-vm") in new stack -- Executing Macro("SIP/2000-7e61", "record-enable|1234|IN") in new stack -- Executing GotoIf("SIP/2000-7e61", "0 > 0?2:4") in new stack -- Goto (macro-record-enable,s,4) -- Executing AGI("SIP/2000-7e61", "recordingcheck|20050106-193009|1105057809.8") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck recordingcheck|20050106-193009|1105057809.8: Inbound recording not enabled -- AGI Script recordingcheck completed, returning 0 -- Executing NoOp("SIP/2000-7e61", "No recording needed") in new stack -- Executing Macro("SIP/2000-7e61", "dial|49|tr|1234") in new stack -- Executing GotoIf("SIP/2000-7e61", "0?4:2") in new stack -- Goto (macro-dial,s,2) -- Executing GotoIf("SIP/2000-7e61", "0?5:4") in new stack -- Goto (macro-dial,s,4) -- Executing AGI("SIP/2000-7e61", "dialparties.agi") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi -- AGI Script dialparties.agi completed, returning 0 -- Executing NoOp("SIP/2000-7e61", "Returned from dialparties with no extensions to call") in new stack -- Executing SetVar("SIP/2000-7e61", "DIALSTATUS=BUSY") in new stack -- Executing GotoIf("SIP/2000-7e61", "0?s-BUSY|1") in new stack -- Executing GotoIf("SIP/2000-7e61", "1?s-BUSY|1") in new stack -- Goto (macro-exten-vm,s-BUSY,1) -- Executing NoOp("SIP/2000-7e61", "Extension is reporting BUSY and has no Voicemail") in new stack -- Executing Busy("SIP/2000-7e61", "") in new stack == Spawn extension (macro-exten-vm, s-BUSY, 2) exited non-zero on 'SIP/2000-7e61' in macro 'exten-vm' == Spawn extension (from-internal, 1234, 1) exited non-zero on 'SIP/2000-7e61' -- Executing Macro("SIP/2000-7e61", "hangupcall") in new stack -- Executing ResetCDR("SIP/2000-7e61", "w") in new stack -- Executing NoCDR("SIP/2000-7e61", "") in new stack -- Executing Wait("SIP/2000-7e61", "5") in new stack == Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'SIP/2000-7e61' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/2000-7e61' asterisk1*CLI> ------------------------------------------------------------------------------ Yahoo! Shopping Find Great Deals on Holiday Gifts at Yahoo! Shopping ------------------------------------------------------------------------------ Yahoo! Shopping Find Great Deals on Holiday Gifts at Yahoo! Shopping ------------------------------------------------------------------------------ _______________________________________________ LIsta de discussões AsteriskBrasil.org AsteriskBrasil em listas.asteriskbrasil.org http://listas.asteriskbrasil.org/mailman/listinfo/asteriskbrasil _______________________________________________ Acesse o wiki AsteriskBrasil.org: http://www.asteriskbrasil.org -------------- Próxima Parte ---------- Um anexo em HTML foi limpo... URL: http://listas.asteriskbrasil.org/pipermail/asteriskbrasil/attachments/20051223/85e53959/attachment.html