[AsteriskBrasil] RES: Magnus Billing
info em magnussolution.com
info em magnussolution.com
Terça Outubro 1 16:32:49 BRT 2013
ola, use o firebug e veja que erro que da
Att. magnus
On oct 1, 2013, at 4:05 p.m., Eduardo <dudutecinf em bol.com.br> wrote:
> Magnus boa tarde,fiz a instalação no meu elastix de acordo com o procedimento de instalação do mesmo no site,deu tudo ok,porém quando entro na tela inicial aparece o globo no centro da pagina e só fica o nome abaixo iniciando,fiz algo errado ou a necessidade de instalar mais alguma coisa?
>
> Grato
> Eduardo
>
> De: asteriskbrasil-bounces em listas.asteriskbrasil.org [mailto:asteriskbrasil-bounces em listas.asteriskbrasil.org] Em nome de info em magnussolution.com
> Enviada em: terça-feira, 1 de outubro de 2013 11:13
> Para: asteriskbrasil em listas.asteriskbrasil.org
> Assunto: Re: [AsteriskBrasil] Magnus Billing
>
> na verdade seu problema nao é entre o elastix e o MBilling, é entre o Mbilling eo seu tronco.
>
> Esta mensagem "[Oct 1 09:57:22] WARNING[4243]: chan_sip.c:6031 sip_call: No audio format found to offer. Cancelling call to 556492361315"
> é recebida quando a ligação é enviada para a Brasiltel. Possivelmente eles nao aceitam codec 711.
>
> Tente instalar o codec 729 no MBilling e ative ele no tronco Brasiltel.
>
>
> At.. Magnus
>
>
>
>
> On oct 1, 2013, at 11:01 a.m., Wilson Ritt Iglesias <wilson.ritt em hotmail.com> wrote:
>
>
> Não possuo g729 no Elastix.
>
> Desabilitei conforme imagem abaixo;
>
>
>
> <image001.jpg>
>
>
>
> <Capture2.PNG>
>
>
>
> Ainda com o mesmo problema =\
>
> localhost*CLI>
> == Using SIP RTP CoS mark 5
> -- Executing [556492361315 em billing:1] AGI("SIP/3000-00000064", "magnus")
> -- Launched AGI Script /var/lib/asterisk/agi-bin/magnus
> -- AGI Script Executing Application: (DIAL) Options: (sip/Brasiltel/556492361315,60,L(300000000:61000:30000))
> > Limit Data for this call:
> > timelimit = 300000000 ms (300000.000 s)
> > play_warning = 61000 ms (61.000 s)
> > play_to_caller = yes
> > play_to_callee = no
> > warning_freq = 30000 ms (30.000 s)
> > start_sound =
> > warning_sound = timeleft
> > end_sound =
> == Using SIP RTP CoS mark 5
> [Oct 1 09:57:22] WARNING[4243]: chan_sip.c:6031 sip_call: No audio format found to offer. Cancelling call to 556492361315
> -- Couldn't call sip/Brasiltel/556492361315
> == Everyone is busy/congested at this time (0:0/0/0)
> -- <SIP/3000-00000064>AGI Script magnus completed, returning 0
> -- Executing [556492361315 em billing:2] Hangup("SIP/3000-00000064", "")
> == Spawn extension (billing, 556492361315, 2) exited non-zero on 'SIP/3000-00000064'
>
>
> From: info em magnussolution.com
> Date: Tue, 1 Oct 2013 10:52:29 -0300
> To: asteriskbrasil em listas.asteriskbrasil.org
> Subject: Re: [AsteriskBrasil] Magnus Billing
>
> ola, seu Elastix e o Mbilling esta com codec 729?
>
> Se nao tiver, desative estes codec no tronco do Mbilling
>
>
>
>
>
> On oct 1, 2013, at 10:42 a.m., Wilson Ritt Iglesias <wilson.ritt em hotmail.com> wrote:
>
> Perdão, peguei a saÃda do Elastix e não no MBilling.
>
> Continuo tendo o mesmo problema:
>
> localhost*CLI>
> == Using SIP RTP CoS mark 5
> -- Executing [556492361315 em billing:1] AGI("SIP/3000-00000058", "magnus")
> -- Launched AGI Script /var/lib/asterisk/agi-bin/magnus
> -- AGI Script Executing Application: (DIAL) Options: (sip/Brasiltel/556492361315,60,L(300000000:61000:30000))
> == Using SIP RTP CoS mark 5
> [Oct 1 09:41:54] WARNING[4032]: chan_sip.c:6031 sip_call: No audio format found to offer. Cancelling call to 556492361315
> -- Couldn't call sip/Brasiltel/556492361315
> == Everyone is busy/congested at this time (0:0/0/0)
> -- <SIP/3000-00000058>AGI Script magnus completed, returning 0
> -- Executing [556492361315 em billing:2] Hangup("SIP/3000-00000058", "")
> == Spawn extension (billing, 556492361315, 2) exited non-zero on 'SIP/3000-00000058'
>
>
> From: wilson.ritt em hotmail.com
> To: asteriskbrasil em listas.asteriskbrasil.org
> Date: Tue, 1 Oct 2013 10:17:40 -0300
> Subject: Re: [AsteriskBrasil] Magnus Billing
>
> Liberei todos os codecs no MBilling, e alterou a saÃda quando tento realizar a ligação, porém ainda recebo tom de ocupado... Pelo visto agora deu algum declined...
> (-- Got SIP response 603 "Declined" back from 189.38.32.8:5060)
>
> pabx*CLI>
> == Using SIP RTP TOS bits 184
> == Using SIP RTP CoS mark 5
> -- Executing [556492361315 em from-internal:1] Macro("SIP/5236-00006183", "user-callerid,SKIPTTL,") in new stack
> -- Executing [s em macro-user-callerid:1] Set("SIP/5236-00006183", "AMPUSER=5236") in new stack
> -- Executing [s em macro-user-callerid:2] GotoIf("SIP/5236-00006183", "0?report") in new stack
> -- Executing [s em macro-user-callerid:3] ExecIf("SIP/5236-00006183", "1?Set(REALCALLERIDNUM=5236)") in new stack
> -- Executing [s em macro-user-callerid:4] Set("SIP/5236-00006183", "AMPUSER=5236") in new stack
> -- Executing [s em macro-user-callerid:5] Set("SIP/5236-00006183", "AMPUSERCIDNAME=Wilson") in new stack
> -- Executing [s em macro-user-callerid:6] GotoIf("SIP/5236-00006183", "0?report") in new stack
> -- Executing [s em macro-user-callerid:7] Set("SIP/5236-00006183", "AMPUSERCID=5236") in new stack
> -- Executing [s em macro-user-callerid:8] Set("SIP/5236-00006183", "CALLERID(all)="Wilson" <5236>") in new stack
> -- Executing [s em macro-user-callerid:9] ExecIf("SIP/5236-00006183", "0?Set(CHANNEL(language)=)") in new stack
> -- Executing [s em macro-user-callerid:10] GotoIf("SIP/5236-00006183", "1?continue") in new stack
> -- Goto (macro-user-callerid,s,19)
> -- Executing [s em macro-user-callerid:19] Set("SIP/5236-00006183", "CALLERID(number)=5236") in new stack
> -- Executing [s em macro-user-callerid:20] Set("SIP/5236-00006183", "CALLERID(name)=Wilson") in new stack
> -- Executing [s em macro-user-callerid:21] NoOp("SIP/5236-00006183", "Using CallerID "Wilson" <5236>") in new stack
> -- Executing [556492361315 em from-internal:2] NoOp("SIP/5236-00006183", "Calling Out Route: Magnus_Teste") in new stack
> -- Executing [556492361315 em from-internal:3] Set("SIP/5236-00006183", "MOHCLASS=default") in new stack
> -- Executing [556492361315 em from-internal:4] Set("SIP/5236-00006183", "_NODEST=") in new stack
> -- Executing [556492361315 em from-internal:5] Macro("SIP/5236-00006183", "record-enable,5236,OUT,") in new stack
> -- Executing [s em macro-record-enable:1] GotoIf("SIP/5236-00006183", "1?check") in new stack
> -- Goto (macro-record-enable,s,4)
> -- Executing [s em macro-record-enable:4] ExecIf("SIP/5236-00006183", "0?MacroExit()") in new stack
> -- Executing [s em macro-record-enable:5] GotoIf("SIP/5236-00006183", "0?Group:OUT") in new stack
> -- Goto (macro-record-enable,s,15)
> -- Executing [s em macro-record-enable:15] GotoIf("SIP/5236-00006183", "0?IN") in new stack
> -- Executing [s em macro-record-enable:16] ExecIf("SIP/5236-00006183", "0?MacroExit()") in new stack
> -- Executing [s em macro-record-enable:17] NoOp("SIP/5236-00006183", "Recording enable for 5236") in new stack
> -- Executing [s em macro-record-enable:18] Set("SIP/5236-00006183", "CALLFILENAME=OUT5236-20131001-131356-1380644036.77600") in new stack
> -- Executing [s em macro-record-enable:19] Goto("SIP/5236-00006183", "record") in new stack
> -- Goto (macro-record-enable,s,23)
> -- Executing [s em macro-record-enable:23] MixMonitor("SIP/5236-00006183", "OUT5236-20131001-131356-1380644036.77600.gsm,,") in new stack
> -- Executing [s em macro-record-enable:24] Set("SIP/5236-00006183", "CDR(userfield)=audio:OUT5236-20131001-131356-1380644036.77600.gsm") in new stack
> -- Executing [s em macro-record-enable:25] MacroExit("SIP/5236-00006183", "") in new stack
> -- Executing [556492361315 em from-internal:6] Macro("SIP/5236-00006183", "dialout-trunk,1,556492361315,") in new stack
> -- Executing [s em macro-dialout-trunk:1] Set("SIP/5236-00006183", "DIAL_TRUNK=1") in new stack
> -- Executing [s em macro-dialout-trunk:2] GosubIf("SIP/5236-00006183", "0?sub-pincheck,s,1") in new stack
> -- Executing [s em macro-dialout-trunk:3] GotoIf("SIP/5236-00006183", "0?disabletrunk,1") in new stack
> -- Executing [s em macro-dialout-trunk:4] Set("SIP/5236-00006183", "DIAL_NUMBER=556492361315") in new stack
> -- Executing [s em macro-dialout-trunk:5] Set("SIP/5236-00006183", "DIAL_TRUNK_OPTIONS=tr") in new stack
> -- Executing [s em macro-dialout-trunk:6] Set("SIP/5236-00006183", "OUTBOUND_GROUP=OUT_1") in new stack
> -- Executing [s em macro-dialout-trunk:7] GotoIf("SIP/5236-00006183", "1?nomax") in new stack
> -- Goto (macro-dialout-trunk,s,9)
> -- Executing [s em macro-dialout-trunk:9] GotoIf("SIP/5236-00006183", "0?skipoutcid") in new stack
> -- Executing [s em macro-dialout-trunk:10] Set("SIP/5236-00006183", "DIAL_TRUNK_OPTIONS=") in new stack
> -- Executing [s em macro-dialout-trunk:11] Macro("SIP/5236-00006183", "outbound-callerid,1") in new stack
> -- Executing [s em macro-outbound-callerid:1] ExecIf("SIP/5236-00006183", "0?Set(CALLERPRES()=)") in new stack
> -- Executing [s em macro-outbound-callerid:2] ExecIf("SIP/5236-00006183", "0?Set(REALCALLERIDNUM=5236)") in new stack
> -- Executing [s em macro-outbound-callerid:3] GotoIf("SIP/5236-00006183", "1?normcid") in new stack
> -- Goto (macro-outbound-callerid,s,6)
> -- Executing [s em macro-outbound-callerid:6] Set("SIP/5236-00006183", "USEROUTCID=") in new stack
> -- Executing [s em macro-outbound-callerid:7] Set("SIP/5236-00006183", "EMERGENCYCID=") in new stack
> -- Executing [s em macro-outbound-callerid:8] Set("SIP/5236-00006183", "TRUNKOUTCID=") in new stack
> -- Executing [s em macro-outbound-callerid:9] GotoIf("SIP/5236-00006183", "1?trunkcid") in new stack
> -- Goto (macro-outbound-callerid,s,12)
> -- Executing [s em macro-outbound-callerid:12] ExecIf("SIP/5236-00006183", "0?Set(CALLERID(all)=)") in new stack
> -- Executing [s em macro-outbound-callerid:13] ExecIf("SIP/5236-00006183", "0?Set(CALLERID(all)=)") in new stack
> -- Executing [s em macro-outbound-callerid:14] ExecIf("SIP/5236-00006183", "0?Set(CALLERID(all)=)") in new stack
> -- Executing [s em macro-outbound-callerid:15] ExecIf("SIP/5236-00006183", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack
> -- Executing [s em macro-dialout-trunk:12] GosubIf("SIP/5236-00006183", "0?sub-flp-1,s,1") in new stack
> -- Executing [s em macro-dialout-trunk:13] Set("SIP/5236-00006183", "OUTNUM=556492361315") in new stack
> -- Executing [s em macro-dialout-trunk:14] Set("SIP/5236-00006183", "custom=SIP/Magnus_Billing") in new stack
> -- Executing [s em macro-dialout-trunk:15] ExecIf("SIP/5236-00006183", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default))") in new stack
> -- Executing [s em macro-dialout-trunk:16] Macro("SIP/5236-00006183", "dialout-trunk-predial-hook,") in new stack
> -- Executing [s em macro-dialout-trunk-predial-hook:1] MacroExit("SIP/5236-00006183", "") in new stack
> -- Executing [s em macro-dialout-trunk:17] GotoIf("SIP/5236-00006183", "0?bypass,1") in new stack
> -- Executing [s em macro-dialout-trunk:18] GotoIf("SIP/5236-00006183", "0?customtrunk") in new stack
> -- Executing [s em macro-dialout-trunk:19] Dial("SIP/5236-00006183", "SIP/Magnus_Billing/556492361315,300,") in new stack
> == Using SIP RTP TOS bits 184
> == Using SIP RTP CoS mark 5
> -- Called SIP/Magnus_Billing/556492361315
> == Begin MixMonitor Recording SIP/5236-00006183
> -- Got SIP response 603 "Declined" back from 189.38.32.8:5060
> -- SIP/Magnus_Billing-00006184 is busy
> == Everyone is busy/congested at this time (1:1/0/0)
> -- Executing [s em macro-dialout-trunk:20] NoOp("SIP/5236-00006183", "Dial failed for some reason with DIALSTATUS = BUSY and HANGUPCAUSE = 21") in new stack
> -- Executing [s em macro-dialout-trunk:21] Goto("SIP/5236-00006183", "s-BUSY,1") in new stack
> -- Goto (macro-dialout-trunk,s-BUSY,1)
> -- Executing [s-BUSY em macro-dialout-trunk:1] NoOp("SIP/5236-00006183", "Dial failed due to trunk reporting BUSY - giving up") in new stack
> -- Executing [s-BUSY em macro-dialout-trunk:2] PlayTones("SIP/5236-00006183", "busy") in new stack
> -- Executing [s-BUSY em macro-dialout-trunk:3] Busy("SIP/5236-00006183", "20") in new stack
> == Spawn extension (macro-dialout-trunk, s-BUSY, 3) exited non-zero on 'SIP/5236-00006183' in macro 'dialout-trunk'
> == Spawn extension (from-internal, 556492361315, 6) exited non-zero on 'SIP/5236-00006183'
> -- Executing [h em from-internal:1] Macro("SIP/5236-00006183", "hangupcall") in new stack
> -- Executing [s em macro-hangupcall:1] GotoIf("SIP/5236-00006183", "0?endmixmoncheck") in new stack
> -- Executing [s em macro-hangupcall:2] Set("SIP/5236-00006183", "MIXMON_CALLFILENAME=/var/spool/asterisk/monitor/OUT5236-20131001-131356-1380644036.77600.gsm") in new stack
> -- Executing [s em macro-hangupcall:3] GotoIf("SIP/5236-00006183", "1?defaultmixmondir") in new stack
> -- Goto (macro-hangupcall,s,5)
> -- Executing [s em macro-hangupcall:5] System("SIP/5236-00006183", "test -e /var/spool/asterisk/monitor/OUT5236-20131001-131356-1380644036.77600.gsm") in new stack
> -- Executing [s em macro-hangupcall:6] NoOp("SIP/5236-00006183", "SYSTEMSTATUS = APPERROR") in new stack
> -- Executing [s em macro-hangupcall:7] GotoIf("SIP/5236-00006183", "0?endmixmoncheck") in new stack
> -- Executing [s em macro-hangupcall:8] Set("SIP/5236-00006183", "CDR(userfield)=") in new stack
> -- Executing [s em macro-hangupcall:9] NoOp("SIP/5236-00006183", "End of MIXMON check") in new stack
> -- Executing [s em macro-hangupcall:10] GotoIf("SIP/5236-00006183", "1?nomeetmemon") in new stack
> -- Goto (macro-hangupcall,s,28)
> -- Executing [s em macro-hangupcall:28] NoOp("SIP/5236-00006183", "End of MEETME check") in new stack
> -- Executing [s em macro-hangupcall:29] GotoIf("SIP/5236-00006183", "1?noautomon") in new stack
> -- Goto (macro-hangupcall,s,34)
> -- Executing [s em macro-hangupcall:34] NoOp("SIP/5236-00006183", "TOUCH_MONITOR_OUTPUT=") in new stack
> -- Executing [s em macro-hangupcall:35] GotoIf("SIP/5236-00006183", "1?noautomon2") in new stack
> -- Goto (macro-hangupcall,s,41)
> -- Executing [s em macro-hangupcall:41] NoOp("SIP/5236-00006183", "MONITOR_FILENAME=") in new stack
> -- Executing [s em macro-hangupcall:42] GotoIf("SIP/5236-00006183", "1?skiprg") in new stack
> -- Goto (macro-hangupcall,s,45)
> -- Executing [s em macro-hangupcall:45] GotoIf("SIP/5236-00006183", "1?skipblkvm") in new stack
> -- Goto (macro-hangupcall,s,48)
> -- Executing [s em macro-hangupcall:48] GotoIf("SIP/5236-00006183", "1?theend") in new stack
> -- Goto (macro-hangupcall,s,50)
> -- Executing [s em macro-hangupcall:50] Hangup("SIP/5236-00006183", "") in new stack
> == Spawn extension (macro-hangupcall, s, 50) exited non-zero on 'SIP/5236-00006183' in macro 'hangupcall'
> == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/5236-00006183'
> == End MixMonitor Recording SIP/5236-00006183
>
>
>
> Date: Tue, 1 Oct 2013 10:13:05 -0300
> From: giovbs em gmail.com
> To: asteriskbrasil em listas.asteriskbrasil.org
> Subject: Re: [AsteriskBrasil] Magnus Billing
>
> Wilson,
>
> verifique os codecs pois está retornando este erro:
>
> No audio format found to offer. Cancelling call to 556492361315
>
> Abraço.
>
>
> Em 1 de outubro de 2013 10:07, Wilson Ritt Iglesias <wilson.ritt em hotmail.com> escreveu:
> Ao tentar ligar, tenho essas saÃdas no painel do asterisk:
>
> Ligando pelo tronco no Elastix que criei:
>
>
>
> [Oct 1 08:46:58] WARNING[3307]: chan_sip.c:6031 sip_call: No audio format found to offer. Cancelling call to 556492361315
> -- Couldn't call sip/Brasiltel/556492361315
>
> _______________________________________________ KHOMP: completa linha de placas externas FXO, FXS, GSM e E1; Media Gateways de 1 a 64 E1s para SIP com R2, ISDN e SS7; Intercomunicadores para acesso remoto via rede IP. Conhe�a em www.Khomp.com. _______________________________________________ ALIGERA � Fabricante nacional de Gateways SIP-E1 para R2, ISDN e SS7. Placas de 1E1, 2E1, 4E1 e 8E1 para PCI ou PCI Express. Channel Bank � Appliance Asterisk - Acesse www.aligera.com.br. _______________________________________________ Para remover seu email desta lista, basta enviar um email em branco para asteriskbrasil-unsubscribe em listas.asteriskbrasil.org
>
> _______________________________________________ KHOMP: completa linha de placas externas FXO, FXS, GSM e E1; Media Gateways de 1 a 64 E1s para SIP com R2, ISDN e SS7; Intercomunicadores para acesso remoto via rede IP. Conhe�a em www.Khomp.com. _______________________________________________ ALIGERA � Fabricante nacional de Gateways SIP-E1 para R2, ISDN e SS7. Placas de 1E1, 2E1, 4E1 e 8E1 para PCI ou PCI Express. Channel Bank � Appliance Asterisk - Acesse www.aligera.com.br. _______________________________________________ Para remover seu email desta lista, basta enviar um email em branco para asteriskbrasil-unsubscribe em listas.asteriskbrasil.org
> _______________________________________________
> KHOMP: completa linha de placas externas FXO, FXS, GSM e E1;
> Media Gateways de 1 a 64 E1s para SIP com R2, ISDN e SS7;
> Intercomunicadores para acesso remoto via rede IP. Conheça em www.Khomp.com.
> _______________________________________________
> ALIGERA – Fabricante nacional de Gateways SIP-E1 para R2, ISDN e SS7.
> Placas de 1E1, 2E1, 4E1 e 8E1 para PCI ou PCI Express.
> Channel Bank – Appliance Asterisk - Acesse www.aligera.com.br.
> _______________________________________________
> Para remover seu email desta lista, basta enviar um email em branco para asteriskbrasil-unsubscribe em listas.asteriskbrasil.org
>
>
> _______________________________________________ KHOMP: completa linha de placas externas FXO, FXS, GSM e E1; Media Gateways de 1 a 64 E1s para SIP com R2, ISDN e SS7; Intercomunicadores para acesso remoto via rede IP. Conhe�a em www.Khomp.com. _______________________________________________ ALIGERA � Fabricante nacional de Gateways SIP-E1 para R2, ISDN e SS7. Placas de 1E1, 2E1, 4E1 e 8E1 para PCI ou PCI Express. Channel Bank � Appliance Asterisk - Acesse www.aligera.com.br. _______________________________________________ Para remover seu email desta lista, basta enviar um email em branco para asteriskbrasil-unsubscribe em listas.asteriskbrasil.org
> _______________________________________________
> KHOMP: completa linha de placas externas FXO, FXS, GSM e E1;
> Media Gateways de 1 a 64 E1s para SIP com R2, ISDN e SS7;
> Intercomunicadores para acesso remoto via rede IP. Conheça em www.Khomp.com.
> _______________________________________________
> ALIGERA – Fabricante nacional de Gateways SIP-E1 para R2, ISDN e SS7.
> Placas de 1E1, 2E1, 4E1 e 8E1 para PCI ou PCI Express.
> Channel Bank – Appliance Asterisk - Acesse www.aligera.com.br.
> _______________________________________________
> Para remover seu email desta lista, basta enviar um email em branco para asteriskbrasil-unsubscribe em listas.asteriskbrasil.org
>
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