Jony, Segue o log abaixo
[2013-12-05 15:31:47] VERBOSE[18836][C-00000599] pbx.c: -- Executing [s@sub-flp-2:1] ExecIf("SIP/6500-00000b84", "0?Set(TARGET_FLP_2=05114973)") in new stack [2013-12-05 15:31:47] VERBOSE[18836][C-00000599] pbx.c: -- Executing [s@sub-flp-2:2] GotoIf("SIP/6500-00000b84", "0?match") in new stack [2013-12-05 15:31:47] VERBOSE[18836][C-00000599] pbx.c: -- Executing [s@sub-flp-2:3] ExecIf("SIP/6500-00000b84", "0?Set(TARGET_FLP_2=0025114973)") in new stack [2013-12-05 15:31:47] VERBOSE[18836][C-00000599] pbx.c: -- Executing [s@sub-flp-2:4] GotoIf("SIP/6500-00000b84", "0?match") in new stack [2013-12-05 15:31:47] VERBOSE[18836][C-00000599] pbx.c: -- Executing [s@sub-flp-2:5] ExecIf("SIP/6500-00000b84", "0?Set(TARGET_FLP_2=0253114973)") in new stack [2013-12-05 15:31:47] VERBOSE[18836][C-00000599] pbx.c: -- Executing [s@sub-flp-2:6] GotoIf("SIP/6500-00000b84", "0?match") in new stack [2013-12-05 15:31:47] VERBOSE[18836][C-00000599] pbx.c: -- Executing [s@sub-flp-2:7] ExecIf("SIP/6500-00000b84", " 0?Set(TARGET_FLP_2=02554304633114973)") in new stack [2013-12-05 15:31:47] VERBOSE[18836][C-00000599] pbx.c: -- Executing [s@sub-flp-2:8] GotoIf("SIP/6500-00000b84", "0?match") in new stack [2013-12-05 15:31:47] VERBOSE[18836][C-00000599] pbx.c: -- Executing [s@sub-flp-2:9] ExecIf("SIP/6500-00000b84", " 0?Set(TARGET_FLP_2=02554304687004973)") in new stack [2013-12-05 15:31:47] VERBOSE[18836][C-00000599] pbx.c: -- Executing [s@sub-flp-2:10] GotoIf("SIP/6500-00000b84", "0?match") in new stack [2013-12-05 15:31:47] VERBOSE[18836][C-00000599] pbx.c: -- Executing [s@sub-flp-2:11] Return("SIP/6500-00000b84", "") in new stack [2013-12-05 15:31:47] VERBOSE[18836][C-00000599] pbx.c: -- Executing [s@macro-dialout-trunk:13] Set("SIP/6500-00000b84", "OUTNUM=33114973") in new stack [2013-12-05 15:31:47] VERBOSE[18836][C-00000599] pbx.c: -- Executing [s@macro-dialout-trunk:14] Set("SIP/6500-00000b84", "custom=SIP/DO_VOXIP") in new stack [2013-12-05 15:31:47] VERBOSE[18836][C-00000599] pbx.c: -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/6500-00000b84", " 0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^))") in new stack [2013-12-05 15:31:47] VERBOSE[18836][C-00000599] pbx.c: -- Executing [s@macro-dialout-trunk:16] ExecIf("SIP/6500-00000b84", " 0?Set(DIAL_TRUNK_OPTIONS=M(confirm))") in new stack [2013-12-05 15:31:47] VERBOSE[18836][C-00000599] pbx.c: -- Executing [s@macro-dialout-trunk:17] Macro("SIP/6500-00000b84", " dialout-trunk-predial-hook,") in new stack [2013-12-05 15:31:47] VERBOSE[18836][C-00000599] pbx.c: -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/6500-00000b84", "") in new stack [2013-12-05 15:31:47] VERBOSE[18836][C-00000599] pbx.c: -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/6500-00000b84", "0?bypass,1") in new stack [2013-12-05 15:31:47] VERBOSE[18836][C-00000599] pbx.c: -- Executing [s@macro-dialout-trunk:19] ExecIf("SIP/6500-00000b84", " 1?Set(CONNECTEDLINE(num,i)=33114973)") in new stack [2013-12-05 15:31:47] VERBOSE[18836][C-00000599] pbx.c: -- Executing [s@macro-dialout-trunk:20] ExecIf("SIP/6500-00000b84", " 1?Set(CONNECTEDLINE(name,i)=CID:5121086689)") in new stack [2013-12-05 15:31:47] VERBOSE[18836][C-00000599] pbx.c: -- Executing [s@macro-dialout-trunk:21] GotoIf("SIP/6500-00000b84", "0?customtrunk") in new stack [2013-12-05 15:31:47] VERBOSE[18836][C-00000599] pbx.c: -- Executing [s@macro-dialout-trunk:22] Dial("SIP/6500-00000b84", " SIP/DO_VOXIP/33114973,300,") in new stack [2013-12-05 15:31:47] VERBOSE[18836][C-00000599] netsock2.c: == Using SIP RTP TOS bits 184 [2013-12-05 15:31:47] VERBOSE[18836][C-00000599] netsock2.c: == Using SIP RTP CoS mark 5 [2013-12-05 15:31:47] VERBOSE[18836][C-00000599] app_dial.c: -- Called SIP/DO_VOXIP/33114973 [2013-12-05 15:31:48] VERBOSE[18836][C-00000599] app_dial.c: -- SIP/DO_VOXIP-00000b85 requested media update control 26, passing it to SIP/6500-00000b84 [2013-12-05 15:31:48] VERBOSE[18836][C-00000599] app_dial.c: -- Call on SIP/DO_VOXIP-00000b85 placed on hold [2013-12-05 15:31:48] VERBOSE[18836][C-00000599] res_musiconhold.c: -- Started music on hold, class 'default', on SIP/6500-00000b84 [2013-12-05 15:31:48] VERBOSE[18836][C-00000599] app_dial.c: -- SIP/DO_VOXIP-00000b85 is making progress passing it to SIP/6500-00000b84 [2013-12-05 15:31:51] VERBOSE[18836][C-00000599] app_dial.c: -- SIP/DO_VOXIP-00000b85 is making progress passing it to SIP/6500-00000b84 [2013-12-05 15:32:02] VERBOSE[18836][C-00000599] app_dial.c: -- SIP/DO_VOXIP-00000b85 answered SIP/6500-00000b84 [2013-12-05 15:32:02] VERBOSE[18836][C-00000599] res_musiconhold.c: -- Stopped music on hold on SIP/6500-00000b84 [2013-12-05 15:32:02] ERROR[1866][C-00000599] netsock2.c: getaddrinfo("PAE1CS2K", "5060", ...): Name or service not known [2013-12-05 15:32:02] WARNING[1866][C-00000599] chan_sip.c: Could not resolve socket address for 'PAE1CS2K:5060' [2013-12-05 15:32:12] VERBOSE[18836][C-00000599] pbx.c: -- Executing [h@macro-dialout-trunk:1] Macro("SIP/6500-00000b84", "hangupcall,") in new stack [2013-12-05 15:32:12] VERBOSE[18836][C-00000599] pbx.c: -- Executing [s@macro-hangupcall:1] GotoIf("SIP/6500-00000b84", "1?theend") in new stack [2013-12-05 15:32:12] VERBOSE[18836][C-00000599] pbx.c: -- Goto (macro-hangupcall,s,3) [2013-12-05 15:32:12] VERBOSE[18836][C-00000599] pbx.c: -- Executing [s@macro-hangupcall:3] ExecIf("SIP/6500-00000b84", " 0?Set(CDR(recordingfile)=)") in new stack [2013-12-05 15:32:12] VERBOSE[18836][C-00000599] pbx.c: -- Executing [s@macro-hangupcall:4] Hangup("SIP/6500-00000b84", "") in new stack [2013-12-05 15:32:12] VERBOSE[18836][C-00000599] app_macro.c: == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/6500-00000b84' in macro 'hangupcall' [2013-12-05 15:32:12] VERBOSE[18836][C-00000599] pbx.c: == Spawn extension (macro-dialout-trunk, h, 1) exited non-zero on 'SIP/6500-00000b84' [2013-12-05 15:32:12] VERBOSE[18836][C-00000599] app_macro.c: == Spawn extension (macro-dialout-trunk, s, 22) exited non-zero on 'SIP/6500-00000b84' in macro 'dialout-trunk' [2013-12-05 15:32:12] VERBOSE[18836][C-00000599] pbx.c: == Spawn extension (from-internal, 033114973, 6) exited non-zero on 'SIP/6500-00000b84' Em 5 de dezembro de 2013 12:33, Jony do Vale <jonydovale...@gmail.com>escreveu: > Teria como enviar a saida da CLI durante o evento? > On Dec 5, 2013 10:53 AM, "Carlos Ferrari" <carlaoferr...@gmail.com> wrote: > >> Prezados, >> >> Estou com um problema no meu Asterisk. >> Existem alguns números específicos para os quais quando eu ligo ao invés >> de completar a ligação cai na música de espera (hold), porém do outro lado >> as pessoas me escutam... >> >> Alguém tem alguma dica pra me ajudar.. >> >> Att >> >> Carlos Alberto Ferrari >> >> _______________________________________________ >> 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 >
_______________________________________________ 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