L'appel d'Asterisk devrait être avec un pair, mais il est pris au piège et rest local

Je travaille avec Asterisk et Vicidial, en essayant de passer des appels sortants via un fournisseur de "trunking" SIP. Les appels ne se connectent jamais, mais j'entends soit démo-instruct.gsm ou invalid.gsm en jouant d'Asterisk.

J'ai remplacé le numéro de téléphone par 9876543210.

Dans le journal, je vois:

[Apr 8 23:38:52] VERBOSE[3279] pbx.c: [Apr 8 23:38:52] == Starting Local/8600051@default-00000000;1 at default,919876543210,1 failed so falling back to exten 's' 

Ce qui se passe là-bas? Quand il dit qu'il a échoué, qu'est-ce qui a échoué et pourquoi? Est-ce que c'est la numérotation avec le SIP "trunking" peer? L'intention est de transmettre ce numéro de téléphone au pair afin que le pair puisse le composer sur le RTPC.

bûche:

 [Apr 8 23:37:00] NOTICE[2910] chan_iax2.c: Peer 'ASTloop' is now UNREACHABLE! Time: 0 [Apr 8 23:37:00] NOTICE[2950] chan_sip.c: Peer '201' is now UNREACHABLE! Last qualify: 0 [Apr 8 23:37:00] NOTICE[2950] chan_sip.c: Peer '200' is now UNREACHABLE! Last qualify: 0 [Apr 8 23:37:02] VERBOSE[2991] manager.c: [Apr 8 23:37:02] == Manager 'sendcron' logged on from 127.0.0.1 [Apr 8 23:37:02] VERBOSE[2990] manager.c: [Apr 8 23:37:02] == Manager 'sendcron' logged on from 127.0.0.1 [Apr 8 23:37:02] VERBOSE[2991] manager.c: [Apr 8 23:37:02] == Manager 'sendcron' logged off from 127.0.0.1 [Apr 8 23:37:02] VERBOSE[2990] manager.c: [Apr 8 23:37:02] == Manager 'sendcron' logged off from 127.0.0.1 [Apr 8 23:37:05] VERBOSE[3069] manager.c: [Apr 8 23:37:05] == Manager 'updatecron' logged on from 127.0.0.1 [Apr 8 23:37:05] VERBOSE[3090] manager.c: [Apr 8 23:37:05] == Manager 'listncron' logged on from 127.0.0.1 [Apr 8 23:37:07] VERBOSE[3106] manager.c: [Apr 8 23:37:07] == Manager 'sendcron' logged on from 127.0.0.1 [Apr 8 23:37:07] VERBOSE[3106] manager.c: [Apr 8 23:37:07] == Manager 'sendcron' logged off from 127.0.0.1 [Apr 8 23:37:08] VERBOSE[2931] chan_iax2.c: [Apr 8 23:37:08] -- Registered IAX2 'ASTloop' (AUTHENTICATED) at 127.0.0.1:23289 [Apr 8 23:37:08] VERBOSE[2932] chan_iax2.c: [Apr 8 23:37:08] -- Registered IAX2 to '127.0.0.1', who sees us as 127.0.0.1:23289 with no messages waiting [Apr 8 23:37:08] VERBOSE[2935] chan_iax2.c: [Apr 8 23:37:08] -- Registered IAX2 'ASTblind' (AUTHENTICATED) at 127.0.0.1:33696 [Apr 8 23:37:08] VERBOSE[2937] chan_iax2.c: [Apr 8 23:37:08] -- Registered IAX2 to '127.0.0.1', who sees us as 127.0.0.1:33696 with no messages waiting [Apr 8 23:37:08] NOTICE[2941] chan_iax2.c: Peer 'ASTloop' is now REACHABLE! Time: 4 [Apr 8 23:37:08] NOTICE[2944] chan_iax2.c: Peer 'ASTblind' is now REACHABLE! Time: 3 [Apr 8 23:37:08] VERBOSE[2899] chan_iax2.c: [Apr 8 23:37:08] -- Registered IAX2 'ASTplay' (AUTHENTICATED) at 127.0.0.1:62907 [Apr 8 23:37:08] VERBOSE[2900] chan_iax2.c: [Apr 8 23:37:08] -- Registered IAX2 to '127.0.0.1', who sees us as 127.0.0.1:62907 with no messages waiting [Apr 8 23:37:08] NOTICE[2904] chan_iax2.c: Peer 'ASTplay' is now REACHABLE! Time: 1 [Apr 8 23:37:47] VERBOSE[2950] chan_sip.c: [Apr 8 23:37:47] -- Registered SIP '200' at 192.168.0.24:5060 [Apr 8 23:37:47] VERBOSE[2950] chan_sip.c: [Apr 8 23:37:47] > Saved useragent "YATE/5.4.2" for peer 200 [Apr 8 23:37:47] NOTICE[2950] chan_sip.c: Peer '200' is now Reachable. (60ms / 2000ms) [Apr 8 23:38:02] VERBOSE[3198] manager.c: [Apr 8 23:38:02] == Manager 'sendcron' logged on from 127.0.0.1 [Apr 8 23:38:02] VERBOSE[3199] manager.c: [Apr 8 23:38:02] == Manager 'sendcron' logged on from 127.0.0.1 [Apr 8 23:38:02] VERBOSE[3199] manager.c: [Apr 8 23:38:02] == Manager 'sendcron' logged off from 127.0.0.1 [Apr 8 23:38:02] VERBOSE[3198] manager.c: [Apr 8 23:38:02] == Manager 'sendcron' logged off from 127.0.0.1 [Apr 8 23:38:07] VERBOSE[3212] manager.c: [Apr 8 23:38:07] == Manager 'sendcron' logged on from 127.0.0.1 [Apr 8 23:38:07] VERBOSE[3212] manager.c: [Apr 8 23:38:07] == Manager 'sendcron' logged off from 127.0.0.1 [Apr 8 23:38:39] VERBOSE[3256] manager.c: [Apr 8 23:38:39] == Manager 'sendcron' logged on from 127.0.0.1 [Apr 8 23:38:39] VERBOSE[3256] netsock2.c: [Apr 8 23:38:39] == Using SIP RTP CoS mark 5 [Apr 8 23:38:41] VERBOSE[3256] pbx.c: [Apr 8 23:38:41] > Channel SIP/200-00000000 was answered. [Apr 8 23:38:41] VERBOSE[3260] pbx.c: [Apr 8 23:38:41] -- Executing [8600051@default:1] MeetMe("SIP/200-00000000", "8600051,F") in new stack [Apr 8 23:38:41] VERBOSE[3260] config.c: [Apr 8 23:38:41] == Parsing '/etc/asterisk/meetme.conf': [Apr 8 23:38:41] VERBOSE[3260] config.c: [Apr 8 23:38:41] == Found [Apr 8 23:38:41] VERBOSE[3260] config.c: [Apr 8 23:38:41] == Parsing '/etc/asterisk/meetme-vicidial.conf': [Apr 8 23:38:41] VERBOSE[3260] config.c: [Apr 8 23:38:41] == Found [Apr 8 23:38:41] VERBOSE[3260] app_meetme.c: [Apr 8 23:38:41] -- Created MeetMe conference 1023 for conference '8600051' [Apr 8 23:38:41] VERBOSE[3260] file.c: [Apr 8 23:38:41] -- <SIP/200-00000000> Playing 'conf-onlyperson.gsm' (language 'en') [Apr 8 23:38:41] WARNING[3260] res_rtp_asterisk.c: RTP Read too short [Apr 8 23:38:42] VERBOSE[3256] manager.c: [Apr 8 23:38:42] == Manager 'sendcron' logged off from 127.0.0.1 [Apr 8 23:38:52] VERBOSE[3277] manager.c: [Apr 8 23:38:52] == Manager 'sendcron' logged on from 127.0.0.1 [Apr 8 23:38:52] VERBOSE[3278] pbx.c: [Apr 8 23:38:52] -- Executing [8600051@default:1] MeetMe("Local/8600051@default-00000000;2", "8600051,F") in new stack [Apr 8 23:38:52] VERBOSE[3277] pbx.c: [Apr 8 23:38:52] > Channel Local/8600051@default-00000000;1 was answered. [Apr 8 23:38:52] VERBOSE[3279] pbx.c: [Apr 8 23:38:52] == Starting Local/8600051@default-00000000;1 at default,919876543210,1 failed so falling back to exten 's' [Apr 8 23:38:52] VERBOSE[3279] pbx_lua.c: [Apr 8 23:38:52] -- Executing [s@default:1] wait("Local/8600051@default-00000000;1", "1") [Apr 8 23:38:53] VERBOSE[3279] pbx_lua.c: [Apr 8 23:38:53] -- Executing [s@default:1] answer("Local/8600051@default-00000000;1", "") [Apr 8 23:38:53] VERBOSE[3279] func_timeout.c: [Apr 8 23:38:53] -- Digit timeout set to 5.000 [Apr 8 23:38:53] VERBOSE[3279] func_timeout.c: [Apr 8 23:38:53] -- Response timeout set to 10.000 [Apr 8 23:38:53] VERBOSE[3279] pbx_lua.c: [Apr 8 23:38:53] -- Executing [s@default:1] background("Local/8600051@default-00000000;1", "demo-congrats") [Apr 8 23:38:53] VERBOSE[3277] manager.c: [Apr 8 23:38:53] == Manager 'sendcron' logged off from 127.0.0.1 [Apr 8 23:38:53] VERBOSE[3279] file.c: [Apr 8 23:38:53] -- <Local/8600051@default-00000000;1> Playing 'demo-congrats.gsm' (language 'en') [Apr 8 23:39:03] VERBOSE[3321] manager.c: [Apr 8 23:39:03] == Manager 'sendcron' logged on from 127.0.0.1 [Apr 8 23:39:03] VERBOSE[3322] manager.c: [Apr 8 23:39:03] == Manager 'sendcron' logged on from 127.0.0.1 [Apr 8 23:39:03] VERBOSE[3322] manager.c: [Apr 8 23:39:03] == Manager 'sendcron' logged off from 127.0.0.1 [Apr 8 23:39:03] VERBOSE[3321] manager.c: [Apr 8 23:39:03] == Manager 'sendcron' logged off from 127.0.0.1 [Apr 8 23:39:05] VERBOSE[3338] manager.c: [Apr 8 23:39:05] == Manager 'sendcron' logged on from 127.0.0.1 [Apr 8 23:39:05] VERBOSE[3278] pbx.c: [Apr 8 23:39:05] == Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-00000000;2' [Apr 8 23:39:05] VERBOSE[3278] pbx.c: [Apr 8 23:39:05] -- Executing [h@default:1] AGI("Local/8600051@default-00000000;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack [Apr 8 23:39:05] VERBOSE[3278] res_agi.c: [Apr 8 23:39:05] -- <Local/8600051@default-00000000;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0 [Apr 8 23:39:05] VERBOSE[3279] pbx.c: [Apr 8 23:39:05] == Spawn extension (default, s, 1) exited non-zero on 'Local/8600051@default-00000000;1' [Apr 8 23:39:05] VERBOSE[3279] pbx.c: [Apr 8 23:39:05] -- Executing [h@default:1] AGI("Local/8600051@default-00000000;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack [Apr 8 23:39:05] VERBOSE[3340] manager.c: [Apr 8 23:39:05] == Manager 'sendcron' logged on from 127.0.0.1 [Apr 8 23:39:05] VERBOSE[3279] res_agi.c: [Apr 8 23:39:05] -- <Local/8600051@default-00000000;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0 [Apr 8 23:39:06] VERBOSE[3338] manager.c: [Apr 8 23:39:06] == Manager 'sendcron' logged off from 127.0.0.1 [Apr 8 23:39:06] VERBOSE[3340] manager.c: [Apr 8 23:39:06] == Manager 'sendcron' logged off from 127.0.0.1 [Apr 8 23:39:08] VERBOSE[3347] manager.c: [Apr 8 23:39:08] == Manager 'sendcron' logged on from 127.0.0.1 [Apr 8 23:39:08] VERBOSE[3347] manager.c: [Apr 8 23:39:08] == Manager 'sendcron' logged off from 127.0.0.1 [Apr 8 23:39:11] VERBOSE[3359] manager.c: [Apr 8 23:39:11] == Manager 'sendcron' logged on from 127.0.0.1 [Apr 8 23:39:11] VERBOSE[3360] pbx.c: [Apr 8 23:39:11] -- Executing [8600051@default:1] MeetMe("Local/8600051@default-00000001;2", "8600051,F") in new stack [Apr 8 23:39:11] VERBOSE[3359] pbx.c: [Apr 8 23:39:11] > Channel Local/8600051@default-00000001;1 was answered. [Apr 8 23:39:11] VERBOSE[3361] pbx.c: [Apr 8 23:39:11] == Starting Local/8600051@default-00000001;1 at default,91919876543210,1 failed so falling back to exten 's' [Apr 8 23:39:11] VERBOSE[3361] pbx_lua.c: [Apr 8 23:39:11] -- Executing [s@default:1] wait("Local/8600051@default-00000001;1", "1") [Apr 8 23:39:12] VERBOSE[3359] manager.c: [Apr 8 23:39:12] == Manager 'sendcron' logged off from 127.0.0.1 [Apr 8 23:39:12] VERBOSE[3361] pbx_lua.c: [Apr 8 23:39:12] -- Executing [s@default:1] answer("Local/8600051@default-00000001;1", "") [Apr 8 23:39:12] VERBOSE[3361] func_timeout.c: [Apr 8 23:39:12] -- Digit timeout set to 5.000 [Apr 8 23:39:12] VERBOSE[3361] func_timeout.c: [Apr 8 23:39:12] -- Response timeout set to 10.000 [Apr 8 23:39:12] VERBOSE[3361] pbx_lua.c: [Apr 8 23:39:12] -- Executing [s@default:1] background("Local/8600051@default-00000001;1", "demo-congrats") [Apr 8 23:39:12] VERBOSE[3361] file.c: [Apr 8 23:39:12] -- <Local/8600051@default-00000001;1> Playing 'demo-congrats.gsm' (language 'en') [Apr 8 23:39:40] VERBOSE[3361] pbx_lua.c: [Apr 8 23:39:40] -- Executing [s@default:1] background("Local/8600051@default-00000001;1", "demo-instruct") [Apr 8 23:39:40] VERBOSE[3361] file.c: [Apr 8 23:39:40] -- <Local/8600051@default-00000001;1> Playing 'demo-instruct.gsm' (language 'en') vici:~ # 

les pairs, où le context de babytel_out est défini comme «trunkinbound» – certainement incorrect. Pourquoi utilise-t-il ce context?

 vici:~ # vici:~ # asterisk -rx "sip show peers" Name/username Host Dyn Forcerport ACL Port Status 200/200 192.168.0.24 DN 64965 OK (35 ms) 201/201 192.168.0.24 DN 5060 OK (38 ms) 202/202 (Unspecified) DN 0 UNKNOWN babytel_in 198.38.7.34 N 5065 OK (84 ms) babytel_out/19876543210 198.38.7.34 N 5065 OK (85 ms) gs102/gs102 (Unspecified) DN 0 UNKNOWN 6 sip peers [Monitored: 4 online, 2 offline Unmonitored: 0 online, 0 offline] vici:~ # vici:~ # asterisk -rx "sip show peer babytel_out" * Name : babytel_out Secret : <Set> MD5Secret : <Not set> Remote Secret: <Not set> Context : trunkinbound Subscr.Cont. : <Not set> Language : en AMA flags : Unknown Netborder CPD: No Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup : Pickupgroup : MOH Suggest : default Mailbox : VM Extension : asterisk LastMsgsSent : 32767/65535 Call limit : 0 Max forwards : 0 Dynamic : No Callerid : "" <> MaxCallBR : 384 kbps Expire : -1 Insecure : invite Force rport : Yes ACL : No DirectMedACL : No T.38 support : No T.38 EC mode : Unknown T.38 MaxDtgrm: 4294967295 DirectMedia : No PromiscRedir : No User=Phone : No Video Support: No Text Support : No Ign SDP ver : No Trust RPID : No Send RPID : Yes TrustIDOutbnd: Legacy Subscriptions: Yes Overlap dial : No Outb. proxy : nat5.babytel.ca DTMFmode : rfc2833 Timer T1 : 500 Timer B : 32000 ToHost : nat5.babytel.ca Addr->IP : 198.38.7.34:5065 Defaddr->IP : (null) Prim.Transp. : UDP Allowed.Trsp : UDP Def. Username: 19876543210 SIP Options : (none) Codecs : 0x6 (gsm|ulaw) Codec Order : (ulaw:20,gsm:20) Auto-Framing : No Status : OK (85 ms) Useragent : Reg. Contact : Qualify Freq : 60000 ms Sess-Timers : Accept Sess-Refresh : uas Sess-Expires : 1800 secs Min-Sess : 90 secs RTP Engine : asterisk Parkinglot : Use Reason : No Encryption : No vici:~ # 

contexts:

 ; WARNING- THIS FILE IS AUTO-GENERATED BY VICIDIAL, ANY EDITS YOU MAKE WILL BE LOST register => 19876543210@sip.babytel.ca:huihylku6786ghjkghjk:19876543210@nat5.babytel.ca:5065/19876543210 ; VICIDIAL Carrier: BABYTEL - babytel ; Babytel [babytel_in] type=peer qualify=yes host=nat5.babytel.ca port=5065 context=inbound-calls [babytel_out] type=peer username=19876543210 host=nat5.babytel.ca outboundproxy=nat5.babytel.ca:5065 secret=huihylku6786ghjkghjk canreinvite=no insecure=invite qualify=yes [200] username=200 secret=password accountcode=200 callerid="" <200> mailbox=200 context=default type=friend host=dynamic [201] username=201 secret=password accountcode=201 callerid="" <201> mailbox=201 context=default type=friend host=dynamic [202] username=202 secret=password accountcode=202 callerid="" <202> mailbox=202 context=default type=friend host=dynamic [gs102] username=gs102 secret=password accountcode=gs102 callerid="Test Admin Phone" <> mailbox=102 context=default type=friend host=dynamic ; END OF FILE Last Forced System Reload: 2015-04-03 17:14:22 

extensions:

 ; WARNING- THIS FILE IS AUTO-GENERATED BY VICIDIAL, ANY EDITS YOU MAKE WILL BE LOST TRUNKloop = IAX2/ASTloop:password@127.0.0.1:40569 TRUNKblind = IAX2/ASTblind:password@127.0.0.1:41569 TRUNKplay = IAX2/ASTplay:password@127.0.0.1:42569 BABY = SIP/babytel_out ; agent phones ressortingcted to only internal extensions [default---agent] exten => s,1,Answer exten => s,n,AGI(agi-VDAD_inbound_calltime_check.agi,-----NO-----default---agent-------------------------NO) exten => s,n,Set(INVCOUNT=0) exten => s,n,Background(sip-silence) exten => s,n,WaitExten(20) ; hangup exten => t,1,Playback(vm-goodbye) exten => t,n,Hangup() exten => i,1,Goto(s,4) exten => i,n,Hangup() ; hangup exten => h,1,AGI(agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME}) ; custom dialplan ensortinges include => vicidial-auto-internal include => vicidial-auto-phones ; logging of all outbound calls from agent phones [defaultlog] exten => s,1,Answer exten => s,n,AGI(agi-VDAD_inbound_calltime_check.agi,-----NO-----defaultlog-------------------------NO) exten => s,n,Set(INVCOUNT=0) exten => s,n,Background(sip-silence) exten => s,n,WaitExten(20) ; hangup exten => t,1,Playback(vm-goodbye) exten => t,n,Hangup() exten => i,1,Goto(s,4) exten => i,n,Hangup() ; hangup exten => h,1,AGI(agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME}) ; custom dialplan ensortinges exten => _X.,1,AGI(agi-NVA_recording.agi,BOTH------Y---Y---Y) exten => _X.,n,Goto(default,${EXTEN},1) [vicidial-auto-external] exten => h,1,AGI(agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME}) ; Local Server: 192.168.0.19 exten => _192*168*000*019*.,1,Goto(default,${EXTEN:16},1) exten => _192*168*000*019*.,2,Hangup() exten => _**192*168*000*019*.,1,Goto(default,${EXTEN:18},1) exten => _**192*168*000*019*.,2,Hangup() ; Agent session audio playback meetme entry exten => _473782178600XXX,1,Meetme(${EXTEN:8},q) exten => _473782178600XXX,n,Hangup() ; Agent session audio playback loop exten => _473782168600XXX,1,Dial(${TRUNKplay}/47378217${EXTEN:8},5,To) exten => _473782168600XXX,n,Hangup() ; Agent session audio playback extension exten => 473782158521111,1,Answer exten => 473782158521111,n,ControlPlayback(${CALLERID(name)},99999,0,1,2,3,4) exten => 473782158521111,n,Hangup() ; SendDTMF to playback channel to control it exten => _473782148521111.,1,Answer exten => _473782148521111.,n,SendDTMF(${CALLERID(num)},250,250,IAX2/ASTplay-${EXTEN:15}) exten => _473782148521111.,n,Hangup() ; Silent wait channel for DTMFsend exten => 473782138521111,1,Answer exten => 473782138521111,n,Wait(5) exten => 473782138521111,n,Hangup() ; VICIDIAL Carrier: BABYTEL - babytel ; Babytel [general] exten => _NXXXNxxxxxx,1,Dial(${BABY}/${EXTEN}) exten => _Nxxxxxx,1,Dial(${BABY}/${EXTEN}) exten => _1NXXNxxxxxx,1,Dial(${BABY}/${EXTEN}) exten => _011.,1,Dial(Dial({TOLL}/${EXTEN}) exten => _9NXXXNxxxxxx,1,Dial(${BABY}/${EXTEN}) exten => _9Nxxxxxx,1,Dial(${BABY}/${EXTEN}) exten => _91NXXNxxxxxx,1,Dial(${BABY}/${EXTEN}) exten => _9011.,1,Dial(Dial({TOLL}/${EXTEN}) [inbound-calls] exten => 19876543210,1,Dial(SIP/200) [local_200] exten => _9x.,1,Set(CALLERID(all)="Ali Baba" <9876543210>) exten => _9x.,1,Dial(SIP/${EXTEN:1}@babytel_out) exten => 201,1,Dial(SIP/201) [local_201] exten => 200,1,Dial(SIP/200) [vicidial-auto-internal] exten => h,1,AGI(agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME}) ; Voicemail Extensions: exten => _85026666666666.,1,Wait(1) exten => _85026666666666.,n,Voicemail(${EXTEN:14},u) exten => _85026666666666.,n,Hangup() exten => _85026666666667.,1,Wait(1) exten => _85026666666667.,n,Voicemail(${EXTEN:14},su) exten => _85026666666667.,n,Hangup() exten => 8500,1,VoicemailMain exten => 8500,2,Goto(s,6) exten => 8500,3,Hangup() exten => 8501,1,VoicemailMain(s${CALLERID(num)}) exten => 8501,2,Hangup() ; Prompt Extensions: exten => 8167,1,Answer exten => 8167,2,AGI(agi-record_prompts.agi,wav-----720000) exten => 8167,3,Hangup() exten => 8168,1,Answer exten => 8168,2,AGI(agi-record_prompts.agi,gsm-----720000) exten => 8168,3,Hangup() ; this is used for recording conference calls, the client app sends the filename ; value as a callerID recordings go to /var/spool/asterisk/monitor (WAV) ; Recording is limited to 1 hour, to make longer, just change the server ; setting ViciDial Recording Limit ; this is the WAV verison, default exten => 8309,1,Answer exten => 8309,2,Monitor(wav,${CALLERID(name)}) exten => 8309,3,Wait(3600) exten => 8309,4,Hangup() ; this is the GSM verison exten => 8310,1,Answer exten => 8310,2,Monitor(gsm,${CALLERID(name)}) exten => 8310,3,Wait(3600) exten => 8310,4,Hangup() ; agent alert extension exten => 83047777777777,1,Answer exten => 83047777777777,2,Playback(${CALLERID(name)}) exten => 83047777777777,3,Hangup() ; This is a loopback dial-around to allow for immediate answer of outbound calls exten => _8305888888888888.,1,Answer exten => _8305888888888888.,n,Wait(${EXTEN:16:1}) exten => _8305888888888888.,n,Dial(${TRUNKloop}/${EXTEN:17},,To) exten => _8305888888888888.,n,Hangup() ; No-call silence extension exten => _8305888888888888X999,1,Answer exten => _8305888888888888X999,n,Wait(3600) exten => _8305888888888888X999,n,Hangup() [vicidial-auto-phones] exten => h,1,AGI(agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME}) ; Phones direct dial extensions: exten => 200,1,Dial(SIP/200,60,) exten => 200,2,Goto(default,85026666666666200,1) exten => 200,3,Hangup() exten => 201,1,Dial(SIP/201,60,) exten => 201,2,Goto(default,85026666666666201,1) exten => 201,3,Hangup() exten => 202,1,Dial(SIP/202,60,) exten => 202,2,Goto(default,85026666666666202,1) exten => 202,3,Hangup() exten => 102,1,Dial(SIP/gs102,60,) exten => 102,2,Goto(default,85026666666666102,1) exten => 102,3,Hangup() [vicidial-auto] exten => h,1,AGI(agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME}) include => vicidial-auto-internal include => vicidial-auto-phones include => vicidial-auto-external ; END OF FILE Last Forced System Reload: 2015-04-03 17:14:22 

voir également:

http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/DeviceConfig_id216341.html

2 Solutions collect form web for “L'appel d'Asterisk devrait être avec un pair, mais il est pris au piège et rest local”

essayez de renommer

 ; Babytel [general] 

à

 ; Babytel [default] 

Cela dit échoué car il est impossible de find cette extension dans le context donné.

Selon votre context par défaut sip.conf pour vos users SIP est par défaut , mais tous les templates qui peuvent gérer 919876543210 sont dans le context général . Ainsi, Asterisk ne peut pas les find lors de l'appel, car le stream d'appels ne passe pas dans le context général .

Vous voulez probablement définir un context général pour vos users SIP.

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