hpacoj

Je crois bien que le répertoire est vide à l'origine! (enfin je ne peut pas vraiment vérifier vu que j'avais déjà mis en français)

Bonjour! Je continue dans mes modifs permettant d'activer les parties du gui qui ne fonctionnent pas dans le package fourni par Synology: après les CDR c'est les logs que j'ai fait fonctionner. en fait il faut modifier le chemin vers le fichier "messages" dans le fichier /volume1/@appstore/Asterisk/var/lib/asterisk/static-http/config/asterisklogs.html On cherche la ligne var tmp_cmd = "/bin/grep /var/log/asterisk/messages -e '" + lm + space + ld + " ' " ; qu'il suffit de remplacer par var tmp_cmd = "/bin/grep /volume1/@appstore/Asterisk/var/log/asterisk/messages -e '" + lm + space + ld + " ' " ; et les logs fonctionnent!

Bon ben j'ai l'air malin moi maintenant!!! En attendant j'ai réussi a faire marcher le cdr-viewer intégré aux options étendues du gui en faisant un lien symbolique vers le fichier Master.csv dans le répertoire /volume1/@appstore/Asterisk/var/lib/asterisk/static-http/config/ (source ici) Donc ma config fonctionne en appels entrants et sortants! cool! Merci!!

donc j'en deduis qu'en rajoutant le 6 dans: [CallingRule_test] exten = _0[123456]XXXXXXXX,1,Macro(trunkdial-failover-0.3,${trunk_1}/${EXTEN:0},,trunk_1,) ca marcherait??? par contre pourquoi cela ne marche pas dans : [CallingRule_portables] exten = _06XXXXXXXX,1,Macro(trunkdial-failover-0.3,${trunk_1}/${EXTEN:0},,trunk_1,) ???

Ben justement... c'est bien mes appels sortants qui ne fonctionnent pas. Le reste n'est pas gênant mais dommage: -configuration de music on hold -les logs dans le gui (toujours vides!) -les CDR vide aussi dans le gui alors que le fichier master.csv dans le rep log/asterisk/cdr-csv/ se remplis bien lui...

Comme je disais, il s'agit là des fichiers créés par le gui. Et peut être dans les options avancées mais sinon ce n'est demandé nulle part! Rectification: Effectivement s'était caché dans les options SIP avancées.

Bonjour. voilà mon SIP.conf: [general] context = default allowoverlap = no udpbindaddr = 0.0.0.0 tcpenable = no tcpbindaddr = 0.0.0.0 transport = udp srvlookup = yes subscribecontext = default language = fr allowexternaldomains = yes allowguest = yes allowsubscribe = yes allowtransfer = yes alwaysauthreject = no autodomain = no callevents = no checkmwi = 10 compactheaders = no defaultexpiry = 1800 dumphistory = no externrefresh = 10 g726nonstandard = no jbenable = no jbforce = no jblog = no maxcallbitrate = 384 maxexpiry = 3600 minexpiry = 60 mohinterpret = default notifyringing = yes pedantic = no progressinband = never promiscredir = no realm = asterisk recordhistory = yes registerattempts = 0 registertimeout = 20 relaxdtmf = no sendrpid = no sipdebug = yes t1min = 100 t38pt_udptl = no tos_audio = none tos_sip = none tos_video = none trustrpid = no useragent = Asterisk PBX usereqphone = no videosupport = yes register = 09XXXXXXXX:MdPfreephonie@freephonie.net disallow = all allow = alaw,ulaw callerid = Maison [basic-options](!); a template dtmfmode = rfc2833 context = from-office type = friend [natted-phone](!,basic-options); another template inheriting basic-options directmedia = no host = dynamic [public-phone](!,basic-options); another template inheriting basic-options directmedia = yes [my-codecs](!); a template for my preferred codecs disallow = all allow = ilbc allow = g729 allow = gsm allow = g723 allow = ulaw [ulaw-phone](!); and another one for ulaw-only disallow = all allow = ulaw mon users.conf [general] fullname = New User userbase = 6000 hasvoicemail = yes vmsecret = 1234 hassip = yes hasiax = yes hasmanager = no callwaiting = yes threewaycalling = yes callwaitingcallerid = yes transfer = yes canpark = yes cancallforward = yes callreturn = yes callgroup = 1 pickupgroup = 1 register = 09XXXXXXXX:MdPfreephonie@freephonie.net vmexten = 0000 [trunk_1] type = peer host = freephonie.net fromuser = 09XXXXXXXX username = 09XXXXXXXX secret = MdPfreephonie trunkname = freetrunk ; GUI metadata ; GUI metadata context = DID_trunk_1 fromdomain = freephonie.net hasexten = no hasiax = no hassip = yes registeriax = no registersip = yes trunkstyle = voip nat = yes insecure = port,invite authuser = 09XXXXXXXX disallow = all allow = alaw,ulaw,gsm,h263 [6000] fullname = paco1 registersip = no host = dynamic callgroup = 1 mailbox = 6000 call-limit = 100 type = peer username = 6000 transfer = yes callcounter = yes context = DLPN_DialPlan1 cid_number = 6000 hasvoicemail = no vmsecret = email = threewaycalling = no hasdirectory = yes callwaiting = no hasmanager = no hasagent = no hassip = yes hasiax = no secret = YYYY nat = yes canreinvite = yes dtmfmode = rfc2833 insecure = no pickupgroup = 1 macaddress = 6000 autoprov = yes label = 6000 linenumber = 1 LINEKEYS = 1 disallow = all allow = ulaw,gsm [6001] fullname = paco2 registersip = no host = dynamic callgroup = 1 mailbox = 6001 call-limit = 100 type = peer username = 6001 transfer = yes callcounter = yes context = DLPN_DialPlan1 cid_number = 6001 hasvoicemail = no vmsecret = email = threewaycalling = no hasdirectory = yes callwaiting = no hasmanager = no hasagent = no hassip = yes hasiax = no secret = YYYY nat = yes canreinvite = yes dtmfmode = rfc2833 insecure = no pickupgroup = 1 disallow = all allow = ulaw,gsm macaddress = 6001 autoprov = yes label = 6001 linenumber = 1 LINEKEYS = 1 [6002] fullname = paco_tab registersip = no host = dynamic callgroup = 1 mailbox = 6002 call-limit = 100 type = peer username = 6002 transfer = yes callcounter = yes context = DLPN_DialPlan1 cid_number = 6002 hasvoicemail = no vmsecret = email = threewaycalling = no hasdirectory = yes callwaiting = no hasmanager = no hasagent = no hassip = yes hasiax = no secret = YYYY nat = yes canreinvite = yes dtmfmode = rfc2833 insecure = no pickupgroup = 1 macaddress = 6002 autoprov = yes label = 6002 linenumber = 1 LINEKEYS = 1 disallow = all allow = ulaw,gsm [6003] fullname = iphone_test registersip = no host = dynamic callgroup = 1 mailbox = 6003 call-limit = 100 type = peer username = 6003 transfer = yes callcounter = yes context = DLPN_DialPlan1 cid_number = 6003 hasvoicemail = no vmsecret = email = threewaycalling = no hasdirectory = yes callwaiting = no hasmanager = no hasagent = no hassip = yes hasiax = no secret = YYYY nat = yes canreinvite = yes dtmfmode = rfc2833 insecure = no pickupgroup = 1 disallow = all allow = alaw,ulaw,gsm macaddress = 6003 autoprov = yes label = 6003 linenumber = 1 LINEKEYS = 1 [6100] fullname = maison registersip = no host = dynamic callgroup = 1 mailbox = 6100 call-limit = 100 type = peer username = 6100 transfer = yes callcounter = yes context = DLPN_DialPlan1 cid_number = 6100 hasvoicemail = yes vmsecret = WWWW email = adresse@mail.fr threewaycalling = no hasdirectory = yes callwaiting = no hasmanager = no hasagent = yes hassip = yes hasiax = no secret = YYYY nat = yes canreinvite = yes dtmfmode = rfc2833 insecure = port,invite pickupgroup = 1 macaddress = 6100 autoprov = yes label = 6100 linenumber = 1 LINEKEYS = 1 disallow = all allow = alaw,ulaw,gsm,h263 [6004] fullname = paco-droid registersip = no host = dynamic callgroup = 1 mailbox = 6004 call-limit = 100 type = peer username = 6004 transfer = yes callcounter = yes context = DLPN_DialPlan1 cid_number = 6004 hasvoicemail = no vmsecret = email = threewaycalling = no hasdirectory = no callwaiting = no hasmanager = no hasagent = no hassip = yes hasiax = no secret = YYYY nat = yes canreinvite = yes dtmfmode = rfc2833 insecure = no pickupgroup = 1 macaddress = 6004 autoprov = yes label = 6004 linenumber = 1 LINEKEYS = 1 disallow = all allow = ulaw,gsm et mon fichier extensions.conf: [general] static = yes writeprotect = no clearglobalvars = no language = fr userscontext = default [globals] TRUNKMSD = 1 ; MSD digits to strip (usually 1 or 0) FEATURES = DIALOPTIONS = RINGTIME = 20 FOLLOWMEOPTIONS = PAGING_HEADER = Intercom trunk_1 = SIP/trunk_1 CID_6000 = toto CID_6001 = toto CID_6002 = toto CID_6003 = toto timeinterval_boulot = 09:00-17:30,mon-fri,*,* CID_6100 = toto FOLLOWME_6000 = 1 CID_6004 = toto CID_trunk_1 = toto [default] exten = _#6XXX,1,Set(MBOX=${EXTEN:1}@default) exten = _#6XXX,n,VoiceMail(${MBOX}) exten = a,1,VoicemailMain(${MBOX}) exten = 0000,1,VoiceMailMain(${CALLERID(num)}@default) [conferences] exten = 6300,1,MeetMe(${EXTEN},M) [ringgroups] exten = 6400,1,Goto(ringroups-custom-1,s,1) [queues] [voicemenus] exten = 7000,1,Goto(voicemenu-custom-1,s,1) exten = 7001,1,Goto(voicemenu-custom-2,s,1) exten = 7001,1,Goto(voicemenu-custom-2,s,1) exten = 7002,1,Goto(voicemenu-custom-3,s,1) exten = 7003,1,Goto(voicemenu-custom-4,s,1) exten = 7000,1,Goto(voicemenu-custom-1,s,1) [voicemailgroups] exten = 6600,1,NoOp(maison) exten = 6600,2,VoiceMail(6100@default) [directory] [pagegroups] [asterisk_guitools] exten = executecommand,1,System(${command}) exten = executecommand,n,Hangup() exten = record_vmenu,1,Answer exten = record_vmenu,n,Playback(vm-intro) exten = record_vmenu,n,Record(${var1},0,500,k) exten = record_vmenu,n,Playback(vm-saved) exten = record_vmenu,n,Playback(vm-goodbye) exten = record_vmenu,n,Hangup exten = play_file,1,Answer exten = play_file,n,Playback(${var1}) exten = play_file,n,Hangup [DID_trunk_1] include = DID_trunk_1_default [DID_trunk_1_default] exten = s/_074831425,1,Goto(voicemenu-custom-2,s,1) exten = s/_00390291596070,1,Goto(voicemenu-custom-2,s,1) exten = s/_0223752201,1,Goto(voicemenu-custom-2,s,1) exten = s/_046271316,1,Goto(voicemenu-custom-2,s,1) exten = s/_06YYYYYYYY,1,Goto(voicemenu-custom-1,s,1) exten = s/_0169XXXXXX,1,Goto(ringroups-custom-1,s,1) exten = s/_06XXXXXXXX,1,Goto(ringroups-custom-1,s,1) exten = s/_0[12345]XXXXXXXX,1,Goto(ringroups-custom-1,s,1) exten = s,1,Goto(ringroups-custom-1,s,1) exten = s,2,Voicemail(6100,u) [page_an_extension] [queue-member-manager] exten = handle_member,1,Verbose(2, Looping through queues to log in or out queue members) exten = handle_member,n,Set(thisActiveMember=${CHANNEL(channeltype)}/${CHANNEL(peername)}) exten = handle_member,n,Set(queue_field=2) exten = handle_member,n,Set(thisQueueXtn=${CUT(QUEUES,,,${queue_field})}) exten = handle_member,n,While($[${EXISTS(${thisQueueXtn})}]) exten = handle_member,n,Macro(member-loginlogout) exten = handle_member,n,Set(queue_field=$[${queue_field} + 1]) exten = handle_member,n,Set(thisQueueXtn=${CUT(QUEUES,,,${queue_field})}) exten = handle_member,n,EndWhile() [macro-member-loginlogout] exten = s,1,Verbose(2, Logging queue member in or out of the request queue) exten = s,n,Set(thisQueue=${thisQueueXtn}) exten = s,n,Set(queueMembers=${QUEUE_MEMBER_LIST(${thisQueue})}) exten = s,n,Set(field=1) exten = s,n,Set(logged_in=0) exten = s,n,Set(thisQueueMember=${CUT(queueMembers,,,${field})}) exten = s,n,While($[${EXISTS(${thisQueueMember})}]) exten = s,n,GotoIf($["${thisQueueMember}" != "${thisActiveMember}"]?check_next) exten = s,n,Set(logged_in=1) exten = s,n,ExitWhile() exten = s,n(check_next),Set(field=$[${field} + 1]) exten = s,n,Set(thisQueueMember=${CUT(queueMembers,,,${field})}) exten = s,n,EndWhile() exten = s,n,MacroIf($[${logged_in} = 0]?q_login:q_logout) [macro-q_login] exten = s,1,Verbose(2, Logging ${thisActiveMember} into the ${thisQueue} queue) exten = s,n,AddQueueMember(${thisQueue},${thisActiveMember}) exten = s,n,Playback(silence/1) exten = s,n,ExecIf($["${AQMSTATUS}" = "ADDED"]?Playback(agent-loginok):Playback(an-error-has-occurred)) [macro-q_logout] exten = s,1,Verbose(2, Logged ${thisActiveMember} out of ${thisQueue} queue) exten = s,n,RemoveQueueMember(${thisQueue},${thisActiveMember}) exten = s,n,Playback(silence/1) exten = s,n,ExecIf($["${RQMSTATUS}" = "REMOVED"]?Playback(agent-loggedoff):Playback(an-error-has-occurred)) [CallingRule_test] exten = _0[12345]XXXXXXXX,1,Macro(trunkdial-failover-0.3,${trunk_1}/${EXTEN:0},,trunk_1,) [CallingRule_portables] exten = _06XXXXXXXX,1,Macro(trunkdial-failover-0.3,${trunk_1}/${EXTEN:0},,trunk_1,) [voicemenu-custom-2] exten = s,1,NoOp(filtrage) exten = s,2,Answer() exten = s,3,Ringing() exten = s,4,Wait(30) exten = s,5,Congestion(6) exten = s,6,Hangup() [voicemenu-custom-3] exten = s,1,NoOp(MCM) exten = s,2,Answer() exten = s,3,Background(beep) exten = s,4,Background(hello-world) exten = s,5,WaitExten(2) exten = 0,1,SayNumber(0) exten = 0,2,Goto(default,6000,1) exten = 1,1,SayNumber(1) exten = 1,2,Goto(default,6001,1) exten = 2,1,SayNumber(2) exten = 2,2,Goto(default,6002,1) exten = 3,1,SayNumber(3) exten = 3,2,Goto(default,6003,1) exten = #,1,Voicemail(6100,u) exten = t,1,Hangup [voicemenu-custom-4] exten = s,1,NoOp(horloge) exten = s,2,Answer() exten = s,3,SayUnixTime(,CET,AdBYkMS) exten = s,4,wait(5) exten = s,5,Hangup() [macro-stdexten] exten = s,1,Set(__DYNAMIC_FEATURES=${FEATURES}) exten = s,2,Set(ORIG_ARG1=${ARG1}) exten = s,3,GotoIf($["${FOLLOWME_${ARG1}}" = "1"]?6:4) exten = s,4,Dial(${ARG2},${RINGTIME},${DIALOPTIONS}) exten = s,5,Goto(s-${DIALSTATUS},1) exten = s,6,Macro(stdexten-followme,${ARG1},${ARG2}) exten = s-NOANSWER,1,Voicemail(${ORIG_ARG1},u) exten = s-NOANSWER,2,Goto(default,s,1) exten = s-BUSY,1,Voicemail(${ORIG_ARG1}, exten = s-BUSY,2,Goto(default,s,1) exten = _s-.,1,Goto(s-NOANSWER,1) exten = a,1,VoicemailMain(${ORIG_ARG1}) [macro-stdexten-followme] exten = s,1,Answer exten = s,2,Set(ORIG_ARG1=${ARG1}) exten = s,3,Dial(${ARG2},${RINGTIME},${DIALOPTIONS}) exten = s,4,Set(__FMCIDNUM=${CALLERID(num)}) exten = s,5,Set(__FMCIDNAME=${CALLERID(name)}) exten = s,6,Followme(${ORIG_ARG1},${FOLLOWMEOPTIONS}) exten = s,7,Voicemail(${ORIG_ARG1},u) exten = s-NOANSWER,1,Voicemail(${ORIG_ARG1},u) exten = s-BUSY,1,Voicemail(${ORIG_ARG1}, exten = s-BUSY,2,Goto(default,s,1) exten = _s-.,1,Goto(s-NOANSWER,1) exten = a,1,VoicemailMain(${ORIG_ARG1}) [macro-pagingintercom] exten = s,1,SIPAddHeader(Alert-Info: ${PAGING_HEADER}) exten = s,2,Page(${ARG1},${ARG2}) exten = s,3,Hangup [macro-local-callingrule-cid-0.1] exten = s,1,Set(CALLERID(all)=${IF($[${LEN(${ARG4})} > 2]?${ARG4}}) exten = s,n,Goto(${ARG1},${ARG2},${ARG3}) [macro-trunkdial-failover-0.3] exten = s,1,GotoIf($[${LEN(${FMCIDNUM})} > 6]?1-fmsetcid,1) exten = s,n,GotoIf($[${LEN(${GLOBAL_OUTBOUNDCIDNAME})} > 1]?1-setgbobname,1) exten = s,n,Set(CALLERID(num)=${IF($[${LEN(${CID_${CALLERID(num)}})} > 2]?${CID_${CALLERID(num)}}}) exten = s,n,Set(CALLERID(all)=${IF($[${LEN(${ARG5})} > 2]?${ARG5}}) exten = s,n,GotoIf($[${LEN(${CALLERID(num)})} > 6]?1-dial,1) exten = s,n,Set(CALLERID(all)=${IF($[${LEN(${CID_${ARG3}})} > 6]?${CID_${ARG3}}:${GLOBAL_OUTBOUNDCID})}) exten = s,n,Set(CALLERID(all)=${IF($[${LEN(${ARG5})} > 2]?${ARG5}}) exten = s,n,Goto(1-dial,1) exten = 1-setgbobname,1,Set(CALLERID(name)=${GLOBAL_OUTBOUNDCIDNAME}) exten = 1-setgbobname,n,Goto(s,3) exten = 1-fmsetcid,1,Set(CALLERID(num)=${FMCIDNUM}) exten = 1-fmsetcid,n,Set(CALLERID(name)=${FMCIDNAME}) exten = 1-fmsetcid,n,Goto(s,4) exten = 1-dial,1,Dial(${ARG1}) exten = 1-dial,n,Gotoif(${LEN(${ARG2})} > 0 ?1-${DIALSTATUS},1:1-out,1) exten = 1-CHANUNAVAIL,1,Dial(${ARG2}) exten = 1-CHANUNAVAIL,n,Hangup() exten = 1-CONGESTION,1,Dial(${ARG2}) exten = 1-CONGESTION,n,Hangup() exten = 1-out,1,Hangup() [CallingRule_free] exten = XXXX,1,Macro(trunkdial-failover-0.3,${trunk_1}/${EXTEN:0},,trunk_1,) [voicemenu-custom-1] exten = s,1,NoOp(Menu1) exten = s,2,Answer() exten = s,3,Wait(1) exten = s,4,SayAlpha(OK) exten = s,5,SayNumber(13) exten = s,6,Busy(10) [ringroups-custom-1] exten = s,1,NoOp(maison) exten = s,n,Dial(SIP/6000&SIP/6001&SIP/6002&SIP/6003&SIP/6100,15,${DIALOPTIONS}i) exten = s,n,Voicemail(6100,u) [DLPN_DialPlan1] include = CallingRule_test include = CallingRule_portables include = CallingRule_free include = default include = parkedcalls include = conferences include = ringgroups include = voicemenus include = queues include = voicemailgroups include = directory include = pagegroups include = page_an_extension

Bonjour et bonne année! Je relance un peu le sujet car comme d'autres ici je suis passé avec la nouvelle version d'asterisk. Tout d'abord je précise que je suis sur un DS210 et que asterisk marchait très bien avec les fichiers de configurations fait "à la main" avant la mise à jour. J'ai donc fait la mise à jour pour avoir une interface "friendly" mais je me retrouve avec des fichiers de confs pré-remplis. Ceci dit je suis chez free et mes appels entrants fonctionnent très bien! par contre j'ai des difficultés avec certains appels sortants. il y a d'autres choses que je n'arrive pas à bien faire fonctionner via le gui: les messageries, les conferences, le Music On Hold... bref 2-3 petites choses!!! Le pire c'est que malgré mon état de débutant je pense avoir compris le fonctionnement de base d'asterisk avec configuration via les fichiers mais pour trouver quelque chose expliquant la configuration par le gui.....

Je viens de recevoir une mise à jour disponible pour le paquet Asterisk. Elle met à jour Asterisk-Gui à sa version 2.01rc! Reste à voir si tout marche convenablement!

Bonjour. Le paquet Asterisk disponible sur nos synos est tout à fait fonctionnel tant que l'on ne passe pas par le Web-GUI. En ssh on édite directement les fichiers *.conf avec vi et dans le centre de package on démarre l'application. juste une modification pour activer le SIP free et c'est fonctionnel. Il n'empeche que j'attend avec hâte la mise à jour de Synology. Cdlt, Hpacoj