[AsteriskBrasil] Magnus Billing

Claudiomiro Larssen biro0702 em gmail.com
Terça Outubro 1 11:11:16 BRT 2013 

Não esta faltando um "00" antes do número "556492361315", ou então retirar
o "55", e enviar apenas o "0" antes do "6492361315" ?


On Tue, Oct 1, 2013 at 11:01 AM, Wilson Ritt Iglesias <
wilson.ritt em hotmail.com> wrote:

> Não possuo g729 no Elastix.
>
> Desabilitei conforme imagem abaixo;
>
>
>
>
>
>
>
>
> 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 saída 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 saída 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 saídas 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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