[AsteriskBrasil] Magnus Billing

Wilson Ritt Iglesias wilson.ritt em hotmail.com
Terça Outubro 1 11:01:07 BRT 2013 

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 		 	   		  
-------------- Próxima Parte ----------
Um anexo em HTML foi limpo...
URL: http://listas.asteriskbrasil.org/pipermail/asteriskbrasil/attachments/20131001/33d934ec/attachment-0001.htm 
-------------- Próxima Parte ----------
Um anexo não texto foi limpo...
Nome  : Capture2.PNG
Tipo  : image/png
Tam   : 19277 bytes
Descr.: não disponível
Url   : http://listas.asteriskbrasil.org/pipermail/asteriskbrasil/attachments/20131001/33d934ec/attachment-0001.png

Mais detalhes sobre a lista de discussão AsteriskBrasil