<html><head><meta http-equiv="Content-Type" content="text/html charset=utf-8"><base href="x-msg://540/"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">ola, seu Elastix e o Mbilling esta com codec 729?<div><br></div><div>Se nao tiver, desative estes codec no tronco do Mbilling</div><div><br></div><div><br></div><div><br></div><div><br></div><div><br><div><div>On oct 1, 2013, at 10:42 a.m., Wilson Ritt Iglesias <<a href="mailto:wilson.ritt@hotmail.com">wilson.ritt@hotmail.com</a>> wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite"><div class="hmmessage" style="font-size: 12pt; font-family: Calibri; font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; line-height: normal; orphans: 2; text-align: -webkit-auto; text-indent: 0px; text-transform: none; white-space: normal; widows: 2; word-spacing: 0px; -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px; "><div dir="ltr">Perdão, peguei a saída do Elastix e não no MBilling.<div><br></div><div>Continuo tendo o mesmo problema:</div><div><br></div><div><div>localhost*CLI></div><div> == Using SIP RTP CoS mark 5</div><div> -- Executing [556492361315@billing:1] AGI("SIP/3000-00000058", "magnus")</div><div> -- Launched AGI Script /var/lib/asterisk/agi-bin/magnus</div><div> -- AGI Script Executing Application: (DIAL) Options: (sip/Brasiltel/556492361315,60,L(300000000:61000:30000))</div><div> == Using SIP RTP CoS mark 5</div><div>[Oct 1 09:41:54] WARNING[4032]: chan_sip.c:6031 sip_call: No audio format found to offer. Cancelling call to 556492361315</div><div> -- Couldn't call sip/Brasiltel/556492361315</div><div> == Everyone is busy/congested at this time (0:0/0/0)</div><div> -- <SIP/3000-00000058>AGI Script magnus completed, returning 0</div><div> -- Executing [556492361315@billing:2] Hangup("SIP/3000-00000058", "")</div><div> == Spawn extension (billing, 556492361315, 2) exited non-zero on 'SIP/3000-00000058'</div><div><br></div><br><div><hr id="stopSpelling">From:<span class="Apple-converted-space"> </span><a href="mailto:wilson.ritt@hotmail.com">wilson.ritt@hotmail.com</a><br>To:<span class="Apple-converted-space"> </span><a href="mailto:asteriskbrasil@listas.asteriskbrasil.org">asteriskbrasil@listas.asteriskbrasil.org</a><br>Date: Tue, 1 Oct 2013 10:17:40 -0300<br>Subject: Re: [AsteriskBrasil] Magnus Billing<br><br><div dir="ltr">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...<div>(<span style="font-size: 12pt; ">-- Got SIP response 603 "Declined" back from 189.38.32.8:5060)</span></div><div><span style="font-size: 12pt; "><br></span></div><div><div>pabx*CLI></div><div> == Using SIP RTP TOS bits 184</div><div> == Using SIP RTP CoS mark 5</div><div> -- Executing [556492361315@from-internal:1] Macro("SIP/5236-00006183", "user-callerid,SKIPTTL,") in new stack</div><div> -- Executing [s@macro-user-callerid:1] Set("SIP/5236-00006183", "AMPUSER=5236") in new stack</div><div> -- Executing [s@macro-user-callerid:2] GotoIf("SIP/5236-00006183", "0?report") in new stack</div><div> -- Executing [s@macro-user-callerid:3] ExecIf("SIP/5236-00006183", "1?Set(REALCALLERIDNUM=5236)") in new stack</div><div> -- Executing [s@macro-user-callerid:4] Set("SIP/5236-00006183", "AMPUSER=5236") in new stack</div><div> -- Executing [s@macro-user-callerid:5] Set("SIP/5236-00006183", "AMPUSERCIDNAME=Wilson") in new stack</div><div> -- Executing [s@macro-user-callerid:6] GotoIf("SIP/5236-00006183", "0?report") in new stack</div><div> -- Executing [s@macro-user-callerid:7] Set("SIP/5236-00006183", "AMPUSERCID=5236") in new stack</div><div> -- Executing [s@macro-user-callerid:8] Set("SIP/5236-00006183", "CALLERID(all)="Wilson" <5236>") in new stack</div><div> -- Executing [s@macro-user-callerid:9] ExecIf("SIP/5236-00006183", "0?Set(CHANNEL(language)=)") in new stack</div><div> -- Executing [s@macro-user-callerid:10] GotoIf("SIP/5236-00006183", "1?continue") in new stack</div><div> -- Goto (macro-user-callerid,s,19)</div><div> -- Executing [s@macro-user-callerid:19] Set("SIP/5236-00006183", "CALLERID(number)=5236") in new stack</div><div> -- Executing [s@macro-user-callerid:20] Set("SIP/5236-00006183", "CALLERID(name)=Wilson") in new stack</div><div> -- Executing [s@macro-user-callerid:21] NoOp("SIP/5236-00006183", "Using CallerID "Wilson" <5236>") in new stack</div><div> -- Executing [556492361315@from-internal:2] NoOp("SIP/5236-00006183", "Calling Out Route: Magnus_Teste") in new stack</div><div> -- Executing [556492361315@from-internal:3] Set("SIP/5236-00006183", "MOHCLASS=default") in new stack</div><div> -- Executing [556492361315@from-internal:4] Set("SIP/5236-00006183", "_NODEST=") in new stack</div><div> -- Executing [556492361315@from-internal:5] Macro("SIP/5236-00006183", "record-enable,5236,OUT,") in new stack</div><div> -- Executing [s@macro-record-enable:1] GotoIf("SIP/5236-00006183", "1?check") in new stack</div><div> -- Goto (macro-record-enable,s,4)</div><div> -- Executing [s@macro-record-enable:4] ExecIf("SIP/5236-00006183", "0?MacroExit()") in new stack</div><div> -- Executing [s@macro-record-enable:5] GotoIf("SIP/5236-00006183", "0?Group:OUT") in new stack</div><div> -- Goto (macro-record-enable,s,15)</div><div> -- Executing [s@macro-record-enable:15] GotoIf("SIP/5236-00006183", "0?IN") in new stack</div><div> -- Executing [s@macro-record-enable:16] ExecIf("SIP/5236-00006183", "0?MacroExit()") in new stack</div><div> -- Executing [s@macro-record-enable:17] NoOp("SIP/5236-00006183", "Recording enable for 5236") in new stack</div><div> -- Executing [s@macro-record-enable:18] Set("SIP/5236-00006183", "CALLFILENAME=OUT5236-20131001-131356-1380644036.77600") in new stack</div><div> -- Executing [s@macro-record-enable:19] Goto("SIP/5236-00006183", "record") in new stack</div><div> -- Goto (macro-record-enable,s,23)</div><div> -- Executing [s@macro-record-enable:23] MixMonitor("SIP/5236-00006183", "OUT5236-20131001-131356-1380644036.77600.gsm,,") in new stack</div><div> -- Executing [s@macro-record-enable:24] Set("SIP/5236-00006183", "CDR(userfield)=audio:OUT5236-20131001-131356-1380644036.77600.gsm") in new stack</div><div> -- Executing [s@macro-record-enable:25] MacroExit("SIP/5236-00006183", "") in new stack</div><div> -- Executing [556492361315@from-internal:6] Macro("SIP/5236-00006183", "dialout-trunk,1,556492361315,") in new stack</div><div> -- Executing [s@macro-dialout-trunk:1] Set("SIP/5236-00006183", "DIAL_TRUNK=1") in new stack</div><div> -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/5236-00006183", "0?sub-pincheck,s,1") in new stack</div><div> -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/5236-00006183", "0?disabletrunk,1") in new stack</div><div> -- Executing [s@macro-dialout-trunk:4] Set("SIP/5236-00006183", "DIAL_NUMBER=556492361315") in new stack</div><div> -- Executing [s@macro-dialout-trunk:5] Set("SIP/5236-00006183", "DIAL_TRUNK_OPTIONS=tr") in new stack</div><div> -- Executing [s@macro-dialout-trunk:6] Set("SIP/5236-00006183", "OUTBOUND_GROUP=OUT_1") in new stack</div><div> -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/5236-00006183", "1?nomax") in new stack</div><div> -- Goto (macro-dialout-trunk,s,9)</div><div> -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/5236-00006183", "0?skipoutcid") in new stack</div><div> -- Executing [s@macro-dialout-trunk:10] Set("SIP/5236-00006183", "DIAL_TRUNK_OPTIONS=") in new stack</div><div> -- Executing [s@macro-dialout-trunk:11] Macro("SIP/5236-00006183", "outbound-callerid,1") in new stack</div><div> -- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/5236-00006183", "0?Set(CALLERPRES()=)") in new stack</div><div> -- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/5236-00006183", "0?Set(REALCALLERIDNUM=5236)") in new stack</div><div> -- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/5236-00006183", "1?normcid") in new stack</div><div> -- Goto (macro-outbound-callerid,s,6)</div><div> -- Executing [s@macro-outbound-callerid:6] Set("SIP/5236-00006183", "USEROUTCID=") in new stack</div><div> -- Executing [s@macro-outbound-callerid:7] Set("SIP/5236-00006183", "EMERGENCYCID=") in new stack</div><div> -- Executing [s@macro-outbound-callerid:8] Set("SIP/5236-00006183", "TRUNKOUTCID=") in new stack</div><div> -- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/5236-00006183", "1?trunkcid") in new stack</div><div> -- Goto (macro-outbound-callerid,s,12)</div><div> -- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/5236-00006183", "0?Set(CALLERID(all)=)") in new stack</div><div> -- Executing [s@macro-outbound-callerid:13] ExecIf("SIP/5236-00006183", "0?Set(CALLERID(all)=)") in new stack</div><div> -- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/5236-00006183", "0?Set(CALLERID(all)=)") in new stack</div><div> -- Executing [s@macro-outbound-callerid:15] ExecIf("SIP/5236-00006183", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack</div><div> -- Executing [s@macro-dialout-trunk:12] GosubIf("SIP/5236-00006183", "0?sub-flp-1,s,1") in new stack</div><div> -- Executing [s@macro-dialout-trunk:13] Set("SIP/5236-00006183", "OUTNUM=556492361315") in new stack</div><div> -- Executing [s@macro-dialout-trunk:14] Set("SIP/5236-00006183", "custom=SIP/Magnus_Billing") in new stack</div><div> -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/5236-00006183", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default))") in new stack</div><div> -- Executing [s@macro-dialout-trunk:16] Macro("SIP/5236-00006183", "dialout-trunk-predial-hook,") in new stack</div><div> -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/5236-00006183", "") in new stack</div><div> -- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/5236-00006183", "0?bypass,1") in new stack</div><div> -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/5236-00006183", "0?customtrunk") in new stack</div><div> -- Executing [s@macro-dialout-trunk:19] Dial("SIP/5236-00006183", "SIP/Magnus_Billing/556492361315,300,") in new stack</div><div> == Using SIP RTP TOS bits 184</div><div> == Using SIP RTP CoS mark 5</div><div> -- Called SIP/Magnus_Billing/556492361315</div><div> == Begin MixMonitor Recording SIP/5236-00006183</div><div> -- Got SIP response 603 "Declined" back from 189.38.32.8:5060</div><div> -- SIP/Magnus_Billing-00006184 is busy</div><div> == Everyone is busy/congested at this time (1:1/0/0)</div><div> -- Executing [s@macro-dialout-trunk:20] NoOp("SIP/5236-00006183", "Dial failed for some reason with DIALSTATUS = BUSY and HANGUPCAUSE = 21") in new stack</div><div> -- Executing [s@macro-dialout-trunk:21] Goto("SIP/5236-00006183", "s-BUSY,1") in new stack</div><div> -- Goto (macro-dialout-trunk,s-BUSY,1)</div><div> -- Executing [s-BUSY@macro-dialout-trunk:1] NoOp("SIP/5236-00006183", "Dial failed due to trunk reporting BUSY - giving up") in new stack</div><div> -- Executing [s-BUSY@macro-dialout-trunk:2] PlayTones("SIP/5236-00006183", "busy") in new stack</div><div> -- Executing [s-BUSY@macro-dialout-trunk:3] Busy("SIP/5236-00006183", "20") in new stack</div><div> == Spawn extension (macro-dialout-trunk, s-BUSY, 3) exited non-zero on 'SIP/5236-00006183' in macro 'dialout-trunk'</div><div> == Spawn extension (from-internal, 556492361315, 6) exited non-zero on 'SIP/5236-00006183'</div><div> -- Executing [h@from-internal:1] Macro("SIP/5236-00006183", "hangupcall") in new stack</div><div> -- Executing [s@macro-hangupcall:1] GotoIf("SIP/5236-00006183", "0?endmixmoncheck") in new stack</div><div> -- Executing [s@macro-hangupcall:2] Set("SIP/5236-00006183", "MIXMON_CALLFILENAME=/var/spool/asterisk/monitor/OUT5236-20131001-131356-1380644036.77600.gsm") in new stack</div><div> -- Executing [s@macro-hangupcall:3] GotoIf("SIP/5236-00006183", "1?defaultmixmondir") in new stack</div><div> -- Goto (macro-hangupcall,s,5)</div><div> -- Executing [s@macro-hangupcall:5] System("SIP/5236-00006183", "test -e /var/spool/asterisk/monitor/OUT5236-20131001-131356-1380644036.77600.gsm") in new stack</div><div> -- Executing [s@macro-hangupcall:6] NoOp("SIP/5236-00006183", "SYSTEMSTATUS = APPERROR") in new stack</div><div> -- Executing [s@macro-hangupcall:7] GotoIf("SIP/5236-00006183", "0?endmixmoncheck") in new stack</div><div> -- Executing [s@macro-hangupcall:8] Set("SIP/5236-00006183", "CDR(userfield)=") in new stack</div><div> -- Executing [s@macro-hangupcall:9] NoOp("SIP/5236-00006183", "End of MIXMON check") in new stack</div><div> -- Executing [s@macro-hangupcall:10] GotoIf("SIP/5236-00006183", "1?nomeetmemon") in new stack</div><div> -- Goto (macro-hangupcall,s,28)</div><div> -- Executing [s@macro-hangupcall:28] NoOp("SIP/5236-00006183", "End of MEETME check") in new stack</div><div> -- Executing [s@macro-hangupcall:29] GotoIf("SIP/5236-00006183", "1?noautomon") in new stack</div><div> -- Goto (macro-hangupcall,s,34)</div><div> -- Executing [s@macro-hangupcall:34] NoOp("SIP/5236-00006183", "TOUCH_MONITOR_OUTPUT=") in new stack</div><div> -- Executing [s@macro-hangupcall:35] GotoIf("SIP/5236-00006183", "1?noautomon2") in new stack</div><div> -- Goto (macro-hangupcall,s,41)</div><div> -- Executing [s@macro-hangupcall:41] NoOp("SIP/5236-00006183", "MONITOR_FILENAME=") in new stack</div><div> -- Executing [s@macro-hangupcall:42] GotoIf("SIP/5236-00006183", "1?skiprg") in new stack</div><div> -- Goto (macro-hangupcall,s,45)</div><div> -- Executing [s@macro-hangupcall:45] GotoIf("SIP/5236-00006183", "1?skipblkvm") in new stack</div><div> -- Goto (macro-hangupcall,s,48)</div><div> -- Executing [s@macro-hangupcall:48] GotoIf("SIP/5236-00006183", "1?theend") in new stack</div><div> -- Goto (macro-hangupcall,s,50)</div><div> -- Executing [s@macro-hangupcall:50] Hangup("SIP/5236-00006183", "") in new stack</div><div> == Spawn extension (macro-hangupcall, s, 50) exited non-zero on 'SIP/5236-00006183' in macro 'hangupcall'</div><div> == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/5236-00006183'</div><div> == End MixMonitor Recording SIP/5236-00006183</div><div><br></div><div><br></div><br><div><hr id="ecxstopSpelling">Date: Tue, 1 Oct 2013 10:13:05 -0300<br>From: <a href="mailto:giovbs@gmail.com">giovbs@gmail.com</a><br>To: <a href="mailto:asteriskbrasil@listas.asteriskbrasil.org">asteriskbrasil@listas.asteriskbrasil.org</a><br>Subject: Re: [AsteriskBrasil] Magnus Billing<br><br><div dir="ltr">Wilson,<div><br></div><div>verifique os codecs pois está retornando este erro:<br><br><font color="#191919" face="Arial, Helvetica, sans-serif"><span style="font-size: 14px; line-height: 22.171875px; ">No audio format found to offer. Cancelling call to </span></font><a target="_blank" style="font-family: Arial, Helvetica, sans-serif; font-size: 14px; line-height: 22.171875px; ">556492361315</a><br><div class="ecxgmail_extra"><br>Abraço.<br><br><br><div class="ecxgmail_quote">Em 1 de outubro de 2013 10:07, Wilson Ritt Iglesias<span class="Apple-converted-space"> </span><span dir="ltr"><<a href="mailto:wilson.ritt@hotmail.com" target="_blank">wilson.ritt@hotmail.com</a>></span><span class="Apple-converted-space"> </span>escreveu:<br><blockquote class="ecxgmail_quote" style="border-left-width: 1px; border-left-color: rgb(204, 204, 204); border-left-style: solid; padding-left: 1ex; "><div dir="ltr"><div style="line-height: 21px; color: rgb(68, 68, 68); font-size: 15px; padding: 10px 0px 0px; display: table; table-layout: fixed; width: 1235px; min-height: 180px; "><div style="line-height: 22.171875px; color: rgb(25, 25, 25); font-family: Arial, Helvetica, sans-serif; font-size: 0.95em; padding: 0px; word-wrap: break-word; overflow: hidden; ">Ao tentar ligar, tenho essas saídas no painel do asterisk:</div><div style="line-height: 22.171875px; color: rgb(25, 25, 25); font-family: Arial, Helvetica, sans-serif; font-size: 0.95em; padding: 0px; word-wrap: break-word; overflow: hidden; "><br style="line-height: 19px; "></div><div style="line-height: 22.171875px; color: rgb(25, 25, 25); font-family: Arial, Helvetica, sans-serif; font-size: 0.95em; padding: 0px; word-wrap: break-word; overflow: hidden; ">Ligando pelo tronco no Elastix que criei:<br style="line-height: 19px; padding: 0px; "><br><br><br></div></div></div></blockquote><blockquote class="ecxgmail_quote" style="border-left-width: 1px; border-left-color: rgb(204, 204, 204); border-left-style: solid; padding-left: 1ex; "><div dir="ltr"><div style="line-height: 21px; color: rgb(68, 68, 68); font-size: 15px; padding: 10px 0px 0px; display: table; table-layout: fixed; width: 1235px; min-height: 180px; "><div style="line-height: 22.171875px; color: rgb(25, 25, 25); font-family: Arial, Helvetica, sans-serif; font-size: 0.95em; padding: 0px; word-wrap: break-word; overflow: hidden; "><span style="font-size: 0.95em; ">[Oct 1 08:46:58] WARNING[3307]: chan_sip.c:6031 sip_call: No audio format found to offer. Cancelling call to<span class="Apple-converted-space"> </span></span><a target="_blank" style="font-size: 0.95em; ">556492361315</a><br>-- Couldn't call sip/Brasiltel/<a target="_blank">556492361315</a><br style="line-height: 19px; padding: 0px; "></div></div></div></blockquote></div></div></div></div><br>_______________________________________________ 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 <a href="http://www.Khomp.com">www.Khomp.com</a>. _______________________________________________ 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 <a href="http://www.aligera.com.br">www.aligera.com.br</a>. _______________________________________________ Para remover seu email desta lista, basta enviar um email em branco para <a href="mailto:asteriskbrasil-unsubscribe@listas.asteriskbrasil.org">asteriskbrasil-unsubscribe@listas.asteriskbrasil.org</a></div></div></div><br>_______________________________________________ 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<span class="Apple-converted-space"> </span><a href="http://www.Khomp.com">www.Khomp.com</a>. _______________________________________________ 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<span class="Apple-converted-space"> </span><a href="http://www.aligera.com.br">www.aligera.com.br</a>. _______________________________________________ Para remover seu email desta lista, basta enviar um email em branco para<span class="Apple-converted-space"> </span><a href="mailto:asteriskbrasil-unsubscribe@listas.asteriskbrasil.org">asteriskbrasil-unsubscribe@listas.asteriskbrasil.org</a></div></div></div>_______________________________________________<br>KHOMP: completa linha de placas externas FXO, FXS, GSM e E1;<br>Media Gateways de 1 a 64 E1s para SIP com R2, ISDN e SS7;<br>Intercomunicadores para acesso remoto via rede IP. Conheça em <a href="http://www.Khomp.com">www.Khomp.com</a>.<br>_______________________________________________<br>ALIGERA – Fabricante nacional de Gateways SIP-E1 para R2, ISDN e SS7.<br>Placas de 1E1, 2E1, 4E1 e 8E1 para PCI ou PCI Express.<br>Channel Bank – Appliance Asterisk - Acesse <a href="http://www.aligera.com.br">www.aligera.com.br</a>.<br>_______________________________________________<br>Para remover seu email desta lista, basta enviar um email em branco para <a href="mailto:asteriskbrasil-unsubscribe@listas.asteriskbrasil.org">asteriskbrasil-unsubscribe@listas.asteriskbrasil.org</a></div></blockquote></div><br></div></body></html>