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.r...@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@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@billing:2] Hangup("SIP/3000-00000058", "")
> == Spawn extension (billing, 556492361315, 2) exited non-zero on
> 'SIP/3000-00000058'
>
>
> From: wilson.r...@hotmail.com
> To: asteriskbrasil@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@from-internal:1] Macro("SIP/5236-00006183",
> "user-callerid,SKIPTTL,") in new stack
> -- Executing [s@macro-user-callerid:1] Set("SIP/5236-00006183",
> "AMPUSER=5236") in new stack
> -- Executing [s@macro-user-callerid:2] GotoIf("SIP/5236-00006183",
> "0?report") in new stack
> -- Executing [s@macro-user-callerid:3] ExecIf("SIP/5236-00006183",
> "1?Set(REALCALLERIDNUM=5236)") in new stack
> -- Executing [s@macro-user-callerid:4] Set("SIP/5236-00006183",
> "AMPUSER=5236") in new stack
> -- Executing [s@macro-user-callerid:5] Set("SIP/5236-00006183",
> "AMPUSERCIDNAME=Wilson") in new stack
> -- Executing [s@macro-user-callerid:6] GotoIf("SIP/5236-00006183",
> "0?report") in new stack
> -- Executing [s@macro-user-callerid:7] Set("SIP/5236-00006183",
> "AMPUSERCID=5236") in new stack
> -- Executing [s@macro-user-callerid:8] Set("SIP/5236-00006183",
> "CALLERID(all)="Wilson" <5236>") in new stack
> -- Executing [s@macro-user-callerid:9] ExecIf("SIP/5236-00006183",
> "0?Set(CHANNEL(language)=)") in new stack
> -- Executing [s@macro-user-callerid:10] GotoIf("SIP/5236-00006183",
> "1?continue") in new stack
> -- Goto (macro-user-callerid,s,19)
> -- Executing [s@macro-user-callerid:19] Set("SIP/5236-00006183",
> "CALLERID(number)=5236") in new stack
> -- Executing [s@macro-user-callerid:20] Set("SIP/5236-00006183",
> "CALLERID(name)=Wilson") in new stack
> -- Executing [s@macro-user-callerid:21] NoOp("SIP/5236-00006183", "Using
> CallerID "Wilson" <5236>") in new stack
> -- Executing [556492361315@from-internal:2] NoOp("SIP/5236-00006183",
> "Calling Out Route: Magnus_Teste") in new stack
> -- Executing [556492361315@from-internal:3] Set("SIP/5236-00006183",
> "MOHCLASS=default") in new stack
> -- Executing [556492361315@from-internal:4] Set("SIP/5236-00006183",
> "_NODEST=") in new stack
> -- Executing [556492361315@from-internal:5] Macro("SIP/5236-00006183",
> "record-enable,5236,OUT,") in new stack
> -- Executing [s@macro-record-enable:1] GotoIf("SIP/5236-00006183",
> "1?check") in new stack
> -- Goto (macro-record-enable,s,4)
> -- Executing [s@macro-record-enable:4] ExecIf("SIP/5236-00006183",
> "0?MacroExit()") in new stack
> -- Executing [s@macro-record-enable:5] GotoIf("SIP/5236-00006183",
> "0?Group:OUT") in new stack
> -- Goto (macro-record-enable,s,15)
> -- Executing [s@macro-record-enable:15] GotoIf("SIP/5236-00006183",
> "0?IN") in new stack
> -- Executing [s@macro-record-enable:16] ExecIf("SIP/5236-00006183",
> "0?MacroExit()") in new stack
> -- Executing [s@macro-record-enable:17] NoOp("SIP/5236-00006183",
> "Recording enable for 5236") in new stack
> -- Executing [s@macro-record-enable:18] Set("SIP/5236-00006183",
> "CALLFILENAME=OUT5236-20131001-131356-1380644036.77600") in new stack
> -- Executing [s@macro-record-enable:19] Goto("SIP/5236-00006183",
> "record") in new stack
> -- Goto (macro-record-enable,s,23)
> -- Executing [s@macro-record-enable:23] MixMonitor("SIP/5236-00006183",
> "OUT5236-20131001-131356-1380644036.77600.gsm,,") in new stack
> -- Executing [s@macro-record-enable:24] Set("SIP/5236-00006183",
> "CDR(userfield)=audio:OUT5236-20131001-131356-1380644036.77600.gsm") in new
> stack
> -- Executing [s@macro-record-enable:25] MacroExit("SIP/5236-00006183",
> "") in new stack
> -- Executing [556492361315@from-internal:6] Macro("SIP/5236-00006183",
> "dialout-trunk,1,556492361315,") in new stack
> -- Executing [s@macro-dialout-trunk:1] Set("SIP/5236-00006183",
> "DIAL_TRUNK=1") in new stack
> -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/5236-00006183",
> "0?sub-pincheck,s,1") in new stack
> -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/5236-00006183",
> "0?disabletrunk,1") in new stack
> -- Executing [s@macro-dialout-trunk:4] Set("SIP/5236-00006183",
> "DIAL_NUMBER=556492361315") in new stack
> -- Executing [s@macro-dialout-trunk:5] Set("SIP/5236-00006183",
> "DIAL_TRUNK_OPTIONS=tr") in new stack
> -- Executing [s@macro-dialout-trunk:6] Set("SIP/5236-00006183",
> "OUTBOUND_GROUP=OUT_1") in new stack
> -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/5236-00006183",
> "1?nomax") in new stack
> -- Goto (macro-dialout-trunk,s,9)
> -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/5236-00006183",
> "0?skipoutcid") in new stack
> -- Executing [s@macro-dialout-trunk:10] Set("SIP/5236-00006183",
> "DIAL_TRUNK_OPTIONS=") in new stack
> -- Executing [s@macro-dialout-trunk:11] Macro("SIP/5236-00006183",
> "outbound-callerid,1") in new stack
> -- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/5236-00006183",
> "0?Set(CALLERPRES()=)") in new stack
> -- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/5236-00006183",
> "0?Set(REALCALLERIDNUM=5236)") in new stack
> -- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/5236-00006183",
> "1?normcid") in new stack
> -- Goto (macro-outbound-callerid,s,6)
> -- Executing [s@macro-outbound-callerid:6] Set("SIP/5236-00006183",
> "USEROUTCID=") in new stack
> -- Executing [s@macro-outbound-callerid:7] Set("SIP/5236-00006183",
> "EMERGENCYCID=") in new stack
> -- Executing [s@macro-outbound-callerid:8] Set("SIP/5236-00006183",
> "TRUNKOUTCID=") in new stack
> -- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/5236-00006183",
> "1?trunkcid") in new stack
> -- Goto (macro-outbound-callerid,s,12)
> -- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/5236-00006183",
> "0?Set(CALLERID(all)=)") in new stack
> -- Executing [s@macro-outbound-callerid:13] ExecIf("SIP/5236-00006183",
> "0?Set(CALLERID(all)=)") in new stack
> -- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/5236-00006183",
> "0?Set(CALLERID(all)=)") in new stack
> -- Executing [s@macro-outbound-callerid:15] ExecIf("SIP/5236-00006183",
> "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack
> -- Executing [s@macro-dialout-trunk:12] GosubIf("SIP/5236-00006183",
> "0?sub-flp-1,s,1") in new stack
> -- Executing [s@macro-dialout-trunk:13] Set("SIP/5236-00006183",
> "OUTNUM=556492361315") in new stack
> -- Executing [s@macro-dialout-trunk:14] Set("SIP/5236-00006183",
> "custom=SIP/Magnus_Billing") in new stack
> -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/5236-00006183",
> "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default))") in new stack
> -- Executing [s@macro-dialout-trunk:16] Macro("SIP/5236-00006183",
> "dialout-trunk-predial-hook,") in new stack
> -- Executing [s@macro-dialout-trunk-predial-hook:1]
> MacroExit("SIP/5236-00006183", "") in new stack
> -- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/5236-00006183",
> "0?bypass,1") in new stack
> -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/5236-00006183",
> "0?customtrunk") in new stack
> -- Executing [s@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@macro-dialout-trunk:20] NoOp("SIP/5236-00006183", "Dial
> failed for some reason with DIALSTATUS = BUSY and HANGUPCAUSE = 21") in new
> stack
> -- Executing [s@macro-dialout-trunk:21] Goto("SIP/5236-00006183",
> "s-BUSY,1") in new stack
> -- Goto (macro-dialout-trunk,s-BUSY,1)
> -- Executing [s-BUSY@macro-dialout-trunk:1] NoOp("SIP/5236-00006183",
> "Dial failed due to trunk reporting BUSY - giving up") in new stack
> -- Executing [s-BUSY@macro-dialout-trunk:2]
> PlayTones("SIP/5236-00006183", "busy") in new stack
> -- Executing [s-BUSY@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@from-internal:1] Macro("SIP/5236-00006183", "hangupcall")
> in new stack
> -- Executing [s@macro-hangupcall:1] GotoIf("SIP/5236-00006183",
> "0?endmixmoncheck") in new stack
> -- Executing [s@macro-hangupcall:2] Set("SIP/5236-00006183",
> "MIXMON_CALLFILENAME=/var/spool/asterisk/monitor/OUT5236-20131001-131356-1380644036.77600.gsm")
> in new stack
> -- Executing [s@macro-hangupcall:3] GotoIf("SIP/5236-00006183",
> "1?defaultmixmondir") in new stack
> -- Goto (macro-hangupcall,s,5)
> -- Executing [s@macro-hangupcall:5] System("SIP/5236-00006183", "test -e
> /var/spool/asterisk/monitor/OUT5236-20131001-131356-1380644036.77600.gsm") in
> new stack
> -- Executing [s@macro-hangupcall:6] NoOp("SIP/5236-00006183",
> "SYSTEMSTATUS = APPERROR") in new stack
> -- Executing [s@macro-hangupcall:7] GotoIf("SIP/5236-00006183",
> "0?endmixmoncheck") in new stack
> -- Executing [s@macro-hangupcall:8] Set("SIP/5236-00006183",
> "CDR(userfield)=") in new stack
> -- Executing [s@macro-hangupcall:9] NoOp("SIP/5236-00006183", "End of
> MIXMON check") in new stack
> -- Executing [s@macro-hangupcall:10] GotoIf("SIP/5236-00006183",
> "1?nomeetmemon") in new stack
> -- Goto (macro-hangupcall,s,28)
> -- Executing [s@macro-hangupcall:28] NoOp("SIP/5236-00006183", "End of
> MEETME check") in new stack
> -- Executing [s@macro-hangupcall:29] GotoIf("SIP/5236-00006183",
> "1?noautomon") in new stack
> -- Goto (macro-hangupcall,s,34)
> -- Executing [s@macro-hangupcall:34] NoOp("SIP/5236-00006183",
> "TOUCH_MONITOR_OUTPUT=") in new stack
> -- Executing [s@macro-hangupcall:35] GotoIf("SIP/5236-00006183",
> "1?noautomon2") in new stack
> -- Goto (macro-hangupcall,s,41)
> -- Executing [s@macro-hangupcall:41] NoOp("SIP/5236-00006183",
> "MONITOR_FILENAME=") in new stack
> -- Executing [s@macro-hangupcall:42] GotoIf("SIP/5236-00006183",
> "1?skiprg") in new stack
> -- Goto (macro-hangupcall,s,45)
> -- Executing [s@macro-hangupcall:45] GotoIf("SIP/5236-00006183",
> "1?skipblkvm") in new stack
> -- Goto (macro-hangupcall,s,48)
> -- Executing [s@macro-hangupcall:48] GotoIf("SIP/5236-00006183",
> "1?theend") in new stack
> -- Goto (macro-hangupcall,s,50)
> -- Executing [s@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: gio...@gmail.com
> To: asteriskbrasil@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.r...@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-unsubscr...@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-unsubscr...@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-unsubscr...@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-unsubscr...@listas.asteriskbrasil.org
