Sindicación del Foro Kunena http://www.neomano.org/ Wed, 19 Jun 2013 12:50:36 -0500 Kunena 1.5.11 http://www.neomano.org/components/com_kunena/template/default/images/english/emoticons/rss.gif http://www.neomano.org/ Sindicación del Foro Kunena http://www.neomano.org/index.php/en/component/kunena/79-general/123418-portail-utilisateur-pour-communicaton-unifiee.html?limit=10&start=10#123557 l'autre jour c'était pour une autre finalité Merci en tout cas :) Wed, 19 Jun 2013 12:38:18 -0500 http://www.neomano.org/index.php/en/component/kunena/3-help/123521-where-in-asterisk-are-general-settings.html#123555 I agree Claudio, I am working on another Asterisk but it's not Elastix in fact I have no idea what it is. I need to make a change to the recording sections, but if this in unique to Elastix then is there a way do change this within CLI for Asterisk? Thanks Wed, 19 Jun 2013 12:31:35 -0500 http://www.neomano.org/index.php/en/component/kunena/3-help/123466-busy-here-from-ip-phone.html#123554 No, I fixed it by adding useragent=simton to the sip.conf file , under the line #include sip_general_custom.conf I got the above solution from the phone provider company Wed, 19 Jun 2013 12:05:49 -0500 http://www.neomano.org/index.php/en/component/kunena/11-ayuda/123553-llamadas-entre-inalambricos.html#123553 Buenas!! Mira tengo una situación que me hago a la idea de por donde andan los tiros pero no se como resolverlo. Tengo una elastix con 1 SPA303 y 4 Inalambricos Gigaset C610 IP sobre la misma base. Las comunicaciones sobre mesa hacia inalambricos todo OK, pero entre inalambricos no pueden llamarse, pasarse llamadas, etc.. Creo que esto es debido a que los tres inalambricos pese a ser extensiones diferentes 201, 202 y 203 cuelgan de la misma IP (192.168.1.100) que es la base DECT y asterisk cuando va a hacer la llamada debe encontrar el canal de esa IP ocupado. Vamos que debe ser algo de canales por cada IP y no se como ampliarlo. Log de una transferencia entre inalambricos. -- Started music on hold, class 'default', on SIP/voztelecom-00000288 == Extension Changed 201[ext-local] new state Hold for Notify User 200 RamonPinyas*CLI> RamonPinyas*CLI> RamonPinyas*CLI> RamonPinyas*CLI> == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Executing [202@from-internal:1] Macro("SIP/201-0000029a", "exten-vm,novm,202") in new stack -- Executing [s@macro-exten-vm:1] Macro("SIP/201-0000029a", "user-callerid,") in new stack -- Executing [s@macro-user-callerid:1] Set("SIP/201-0000029a", "AMPUSER=201") in new stack -- Executing [s@macro-user-callerid:2] GotoIf("SIP/201-0000029a", "0?report") in new stack -- Executing [s@macro-user-callerid:3] ExecIf("SIP/201-0000029a", "1?Set(REALCALLERIDNUM=201)") in new stack -- Executing [s@macro-user-callerid:4] Set("SIP/201-0000029a", "AMPUSER=201") in new stack -- Executing [s@macro-user-callerid:5] Set("SIP/201-0000029a", "AMPUSERCIDNAME=Laboral") in new stack -- Executing [s@macro-user-callerid:6] GotoIf("SIP/201-0000029a", "0?report") in new stack -- Executing [s@macro-user-callerid:7] Set("SIP/201-0000029a", "AMPUSERCID=201") in new stack -- Executing [s@macro-user-callerid:8] Set("SIP/201-0000029a", "CALLERID(all)="Laboral" ") in new stack -- Executing [s@macro-user-callerid:9] ExecIf("SIP/201-0000029a", "0?Set(CHANNEL(language)=)") in new stack -- Executing [s@macro-user-callerid:10] GotoIf("SIP/201-0000029a", "0?continue") in new stack -- Executing [s@macro-user-callerid:11] Set("SIP/201-0000029a", "__TTL=64") in new stack -- Executing [s@macro-user-callerid:12] GotoIf("SIP/201-0000029a", "1?continue") in new stack -- Goto (macro-user-callerid,s,19) -- Executing [s@macro-user-callerid:19] Set("SIP/201-0000029a", "CALLERID(number)=201") in new stack -- Executing [s@macro-user-callerid:20] Set("SIP/201-0000029a", "CALLERID(name)=Laboral") in new stack -- Executing [s@macro-user-callerid:21] NoOp("SIP/201-0000029a", "Using CallerID "Laboral" ") in new stack -- Executing [s@macro-exten-vm:2] Set("SIP/201-0000029a", "RingGroupMethod=none") in new stack -- Executing [s@macro-exten-vm:3] Set("SIP/201-0000029a", "VMBOX=novm") in new stack -- Executing [s@macro-exten-vm:4] Set("SIP/201-0000029a", "__EXTTOCALL=202") in new stack -- Executing [s@macro-exten-vm:5] Set("SIP/201-0000029a", "CFUEXT=") in new stack -- Executing [s@macro-exten-vm:6] Set("SIP/201-0000029a", "CFBEXT=") in new stack -- Executing [s@macro-exten-vm:7] Set("SIP/201-0000029a", "RT=""") in new stack -- Executing [s@macro-exten-vm:8] Macro("SIP/201-0000029a", "record-enable,202,IN") in new stack -- Executing [s@macro-record-enable:1] GotoIf("SIP/201-0000029a", "1?check") in new stack -- Goto (macro-record-enable,s,4) -- Executing [s@macro-record-enable:4] ExecIf("SIP/201-0000029a", "0?MacroExit()") in new stack -- Executing [s@macro-record-enable:5] GotoIf("SIP/201-0000029a", "0?Group:OUT") in new stack -- Goto (macro-record-enable,s,15) -- Executing [s@macro-record-enable:15] GotoIf("SIP/201-0000029a", "1?IN") in new stack -- Goto (macro-record-enable,s,20) -- Executing [s@macro-record-enable:20] ExecIf("SIP/201-0000029a", "1?MacroExit()") in new stack -- Executing [s@macro-exten-vm:9] Macro("SIP/201-0000029a", "dial-one,"",tr,202") in new stack -- Executing [s@macro-dial-one:1] Set("SIP/201-0000029a", "DEXTEN=202") in new stack -- Executing [s@macro-dial-one:2] Set("SIP/201-0000029a", "DIALSTATUS_CW=") in new stack -- Executing [s@macro-dial-one:3] GosubIf("SIP/201-0000029a", "0?screen,1") in new stack -- Executing [s@macro-dial-one:4] GosubIf("SIP/201-0000029a", "0?cf,1") in new stack -- Executing [s@macro-dial-one:5] GotoIf("SIP/201-0000029a", "1?skip1") in new stack -- Goto (macro-dial-one,s,8) -- Executing [s@macro-dial-one:8] GotoIf("SIP/201-0000029a", "0?nodial") in new stack -- Executing [s@macro-dial-one:9] GotoIf("SIP/201-0000029a", "0?continue") in new stack -- Executing [s@macro-dial-one:10] Set("SIP/201-0000029a", "EXTHASCW=") in new stack -- Executing [s@macro-dial-one:11] GotoIf("SIP/201-0000029a", "1?next1:cwinusebusy") in new stack -- Goto (macro-dial-one,s,12) -- Executing [s@macro-dial-one:12] GotoIf("SIP/201-0000029a", "0?docfu:skip3") in new stack -- Goto (macro-dial-one,s,16) -- Executing [s@macro-dial-one:16] GotoIf("SIP/201-0000029a", "1?next2:continue") in new stack -- Goto (macro-dial-one,s,17) -- Executing [s@macro-dial-one:17] GotoIf("SIP/201-0000029a", "1?continue") in new stack -- Goto (macro-dial-one,s,25) -- Executing [s@macro-dial-one:25] GotoIf("SIP/201-0000029a", "0?nodial") in new stack -- Executing [s@macro-dial-one:26] GosubIf("SIP/201-0000029a", "1?dstring,1:dlocal,1") in new stack -- Executing [dstring@macro-dial-one:1] Set("SIP/201-0000029a", "DSTRING=") in new stack -- Executing [dstring@macro-dial-one:2] Set("SIP/201-0000029a", "DEVICES=202") in new stack -- Executing [dstring@macro-dial-one:3] ExecIf("SIP/201-0000029a", "0?Return()") in new stack -- Executing [dstring@macro-dial-one:4] ExecIf("SIP/201-0000029a", "0?Set(DEVICES=02)") in new stack -- Executing [dstring@macro-dial-one:5] Set("SIP/201-0000029a", "LOOPCNT=1") in new stack -- Executing [dstring@macro-dial-one:6] Set("SIP/201-0000029a", "ITER=1") in new stack -- Executing [dstring@macro-dial-one:7] Set("SIP/201-0000029a", "THISDIAL=SIP/202") in new stack -- Executing [dstring@macro-dial-one:8] GosubIf("SIP/201-0000029a", "1?zap2dahdi,1") in new stack -- Executing [zap2dahdi@macro-dial-one:1] ExecIf("SIP/201-0000029a", "0?Return()") in new stack -- Executing [zap2dahdi@macro-dial-one:2] Set("SIP/201-0000029a", "NEWDIAL=") in new stack -- Executing [zap2dahdi@macro-dial-one:3] Set("SIP/201-0000029a", "LOOPCNT2=1") in new stack -- Executing [zap2dahdi@macro-dial-one:4] Set("SIP/201-0000029a", "ITER2=1") in new stack -- Executing [zap2dahdi@macro-dial-one:5] Set("SIP/201-0000029a", "THISPART2=SIP/202") in new stack -- Executing [zap2dahdi@macro-dial-one:6] ExecIf("SIP/201-0000029a", "0?Set(THISPART2=DAHDI/202)") in new stack -- Executing [zap2dahdi@macro-dial-one:7] Set("SIP/201-0000029a", "NEWDIAL=SIP/202&") in new stack -- Executing [zap2dahdi@macro-dial-one:8] Set("SIP/201-0000029a", "ITER2=2") in new stack -- Executing [zap2dahdi@macro-dial-one:9] GotoIf("SIP/201-0000029a", "0?begin2") in new stack -- Executing [zap2dahdi@macro-dial-one:10] Set("SIP/201-0000029a", "THISDIAL=SIP/202") in new stack -- Executing [zap2dahdi@macro-dial-one:11] Return("SIP/201-0000029a", "") in new stack -- Executing [dstring@macro-dial-one:9] Set("SIP/201-0000029a", "DSTRING=SIP/202&") in new stack -- Executing [dstring@macro-dial-one:10] Set("SIP/201-0000029a", "ITER=2") in new stack -- Executing [dstring@macro-dial-one:11] GotoIf("SIP/201-0000029a", "0?begin") in new stack -- Executing [dstring@macro-dial-one:12] Set("SIP/201-0000029a", "DSTRING=SIP/202") in new stack -- Executing [dstring@macro-dial-one:13] Return("SIP/201-0000029a", "") in new stack -- Executing [s@macro-dial-one:27] GotoIf("SIP/201-0000029a", "0?nodial") in new stack -- Executing [s@macro-dial-one:28] GotoIf("SIP/201-0000029a", "0?skiptrace") in new stack -- Executing [s@macro-dial-one:29] GosubIf("SIP/201-0000029a", "1?ctset,1:ctclear,1") in new stack -- Executing [ctset@macro-dial-one:1] Set("SIP/201-0000029a", "DB(CALLTRACE/202)=201") in new stack -- Executing [ctset@macro-dial-one:2] Return("SIP/201-0000029a", "") in new stack -- Executing [s@macro-dial-one:30] Set("SIP/201-0000029a", "D_OPTIONS=tr") in new stack -- Executing [s@macro-dial-one:31] ExecIf("SIP/201-0000029a", "0?SIPAddHeader(Alert-Info: )") in new stack -- Executing [s@macro-dial-one:32] ExecIf("SIP/201-0000029a", "0?SIPAddHeader()") in new stack -- Executing [s@macro-dial-one:33] ExecIf("SIP/201-0000029a", "0?Set(CHANNEL(musicclass)=)") in new stack -- Executing [s@macro-dial-one:34] GosubIf("SIP/201-0000029a", "0?qwait,1") in new stack -- Executing [s@macro-dial-one:35] Set("SIP/201-0000029a", "__CWIGNORE=") in new stack -- Executing [s@macro-dial-one:36] Set("SIP/201-0000029a", "__KEEPCID=TRUE") in new stack -- Executing [s@macro-dial-one:37] Dial("SIP/201-0000029a", "SIP/202,"",tr") in new stack == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 == Extension Changed 202[ext-local] new state Ringing for Notify User 200 -- Called SIP/202 -- Got SIP response 486 "Busy Here" back from 192.168.1.100:5060 -- SIP/202-0000029b is busy == Everyone is busy/congested at this time (1:1/0/0) -- Executing [s@macro-dial-one:38] ExecIf("SIP/201-0000029a", "0?Set(DIALSTATUS=)") in new stack -- Executing [s@macro-dial-one:39] GosubIf("SIP/201-0000029a", "0?s-BUSY,1") in new stack -- Executing [s@macro-dial-one:40] MacroExit("SIP/201-0000029a", "") in new stack -- Executing [s@macro-exten-vm:10] GotoIf("SIP/201-0000029a", "0?exit") in new stack -- Executing [s@macro-exten-vm:11] Set("SIP/201-0000029a", "SV_DIALSTATUS=BUSY") in new stack -- Executing [s@macro-exten-vm:12] GosubIf("SIP/201-0000029a", "0?docfu,1") in new stack -- Executing [s@macro-exten-vm:13] GosubIf("SIP/201-0000029a", "0?docfb,1") in new stack -- Executing [s@macro-exten-vm:14] Set("SIP/201-0000029a", "DIALSTATUS=BUSY") in new stack -- Executing [s@macro-exten-vm:15] NoOp("SIP/201-0000029a", "Voicemail is 'novm'") in new stack -- Executing [s@macro-exten-vm:16] GotoIf("SIP/201-0000029a", "1?s-BUSY,1") in new stack -- Goto (macro-exten-vm,s-BUSY,1) -- Executing [s-BUSY@macro-exten-vm:1] NoOp("SIP/201-0000029a", "Extension is reporting BUSY and not passing to Voicemail") in new stack -- Executing [s-BUSY@macro-exten-vm:2] GotoIf("SIP/201-0000029a", "0?exit,1") in new stack -- Executing [s-BUSY@macro-exten-vm:3] PlayTones("SIP/201-0000029a", "busy") in new stack -- Executing [s-BUSY@macro-exten-vm:4] Busy("SIP/201-0000029a", "20") in new stack == Spawn extension (macro-exten-vm, s-BUSY, 4) exited non-zero on 'SIP/201-0000029a' in macro 'exten-vm' == Spawn extension (from-internal, 202, 1) exited non-zero on 'SIP/201-0000029a' -- Executing [h@from-internal:1] Macro("SIP/201-0000029a", "hangupcall") in new stack -- Executing [s@macro-hangupcall:1] GotoIf("SIP/201-0000029a", "1?endmixmoncheck") in new stack -- Goto (macro-hangupcall,s,9) -- Executing [s@macro-hangupcall:9] NoOp("SIP/201-0000029a", "End of MIXMON check") in new stack -- Executing [s@macro-hangupcall:10] GotoIf("SIP/201-0000029a", "1?nomeetmemon") in new stack -- Goto (macro-hangupcall,s,28) -- Executing [s@macro-hangupcall:28] NoOp("SIP/201-0000029a", "End of MEETME check") in new stack -- Executing [s@macro-hangupcall:29] GotoIf("SIP/201-0000029a", "1?noautomon") in new stack -- Goto (macro-hangupcall,s,34) -- Executing [s@macro-hangupcall:34] NoOp("SIP/201-0000029a", "TOUCH_MONITOR_OUTPUT=") in new stack -- Executing [s@macro-hangupcall:35] GotoIf("SIP/201-0000029a", "1?noautomon2") in new stack -- Goto (macro-hangupcall,s,41) -- Executing [s@macro-hangupcall:41] NoOp("SIP/201-0000029a", "MONITOR_FILENAME=") in new stack -- Executing [s@macro-hangupcall:42] GotoIf("SIP/201-0000029a", "1?skiprg") in new stack -- Goto (macro-hangupcall,s,45) -- Executing [s@macro-hangupcall:45] GotoIf("SIP/201-0000029a", "1?skipblkvm") in new stack -- Goto (macro-hangupcall,s,48) -- Executing [s@macro-hangupcall:48] GotoIf("SIP/201-0000029a", "1?theend") in new stack -- Goto (macro-hangupcall,s,50) -- Executing [s@macro-hangupcall:50] AGI("SIP/201-0000029a", "hangup.agi") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/hangup.agi == Extension Changed 202[ext-local] new state Idle for Notify User 200 -- AGI Script hangup.agi completed, returning 0 -- Executing [s@macro-hangupcall:51] Hangup("SIP/201-0000029a", "") in new stack == Spawn extension (macro-hangupcall, s, 51) exited non-zero on 'SIP/201-0000029a' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/201-0000029a' -- Stopped music on hold on SIP/voztelecom-00000288 == Extension Changed 201[ext-local] new state InUse for Notify User 200 RamonPinyas*CLI> Disconnected from Asterisk server Executing last minute cleanups Wed, 19 Jun 2013 11:32:50 -0500 http://www.neomano.org/index.php/en/component/kunena/25-newbies-corner-/123552-single-user-on-multiple-call-queue.html#123552 Hi everybody, I'm new to this forum and i want to say hello to everybody. I've an elastix well configure, trunk, queue and grandstream 1450. When call come is correctly managed and routed to the queue, the problem come when the user is busy and another call come on the queue. The caller start hearing the holding music but the operator isn't able to know that there are another call waiting. How can i make a led blinking or how can i inform user of the second call? Many thanks in advance Wed, 19 Jun 2013 10:49:24 -0500 http://www.neomano.org/index.php/en/component/kunena/3-help/123551-vanishing-caller-id.html#123551 Hi All, I am having an issue with a Sangoma ISDN BRI Card, the inbound caller ID appears to be removed somewhere in my call routing. Here are the logs from an inbound call from 01279800101: [code][Jun 19 11:48:10] VERBOSE[26601] sig_pri.c: -- Accepting call from '01279800101' to '658002' on channel 0/1, span 2 [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Executing [658002@custom-inbound:1] Answer("DAHDI/i2/-49", "") in new stack [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Executing [658002@custom-inbound:2] ExecIf("DAHDI/i2/-49", "0?Set(CALLERID(num)=0)") in new stack [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Executing [658002@custom-inbound:3] Goto("DAHDI/i2/-49", "from-pstn,658002,1") in new stack [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Goto (from-pstn,658002,1) [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Executing [658002@from-pstn:1] NoOp("DAHDI/i2/-49", "Catch-All DID Match - Found 658002 - You probably want a DID for this.") in new stack [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Executing [658002@from-pstn:2] Goto("DAHDI/i2/-49", "ext-did,s,1") in new stack [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Goto (ext-did,s,1) [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Executing [s@ext-did:1] Set("DAHDI/i2/-49", "__FROM_DID=s") in new stack [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Executing [s@ext-did:2] Gosub("DAHDI/i2/-49", "app-blacklist-check,s,1") in new stack [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Executing [s@app-blacklist-check:1] GotoIf("DAHDI/i2/-49", "0?blacklisted") in new stack [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Executing [s@app-blacklist-check:2] Set("DAHDI/i2/-49", "CALLED_BLACKLIST=1") in new stack [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Executing [s@app-blacklist-check:3] Return("DAHDI/i2/-49", "") in new stack [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Executing [s@ext-did:3] Gosub("DAHDI/i2/-49", "cidlookup,cidlookup_1,1") in new stack [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Executing [cidlookup_1@cidlookup:1] ExecIf("DAHDI/i2/-49", "0?Set(CALLERID(name)=)") in new stack [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Executing [cidlookup_1@cidlookup:2] Return("DAHDI/i2/-49", "") in new stack [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Executing [s@ext-did:4] ExecIf("DAHDI/i2/-49", "1 ?Set(CALLERID(name)=)") in new stack [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Executing [s@ext-did:5] Set("DAHDI/i2/-49", "__CALLINGPRES_SV=allowed_not_screened") in new stack [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Executing [s@ext-did:6] Set("DAHDI/i2/-49", "CALLERPRES()=allowed_not_screened") in new stack [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Executing [s@ext-did:7] Goto("DAHDI/i2/-49", "app-daynight,0,1") in new stack [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Goto (app-daynight,0,1) [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Executing [0@app-daynight:1] GotoIf("DAHDI/i2/-49", "0?ext-local,vmu103,1:ext-group,600,1") in new stack [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Goto (ext-group,600,1) [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Executing [600@ext-group:1] Macro("DAHDI/i2/-49", "user-callerid,") in new stack [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Executing [s@macro-user-callerid:1] Set("DAHDI/i2/-49", "AMPUSER=") in new stack [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Executing [s@macro-user-callerid:2] GotoIf("DAHDI/i2/-49", "0?report") in new stack [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Executing [s@macro-user-callerid:3] ExecIf("DAHDI/i2/-49", "1?Set(REALCALLERIDNUM=)") in new stack [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Executing [s@macro-user-callerid:4] Set("DAHDI/i2/-49", "AMPUSER=") in new stack [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Executing [s@macro-user-callerid:5] Set("DAHDI/i2/-49", "AMPUSERCIDNAME=") in new stack [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Executing [s@macro-user-callerid:6] GotoIf("DAHDI/i2/-49", "1?report") in new stack [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Goto (macro-user-callerid,s,10) [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Executing [s@macro-user-callerid:10] GotoIf("DAHDI/i2/-49", "0?continue") in new stack [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Executing [s@macro-user-callerid:11] Set("DAHDI/i2/-49", "__TTL=64") in new stack [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Executing [s@macro-user-callerid:12] GotoIf("DAHDI/i2/-49", "1?continue") in new stack [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Goto (macro-user-callerid,s,19) [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Executing [s@macro-user-callerid:19] Set("DAHDI/i2/-49", "CALLERID(number)=") in new stack [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Executing [s@macro-user-callerid:20] Set("DAHDI/i2/-49", "CALLERID(name)=") in new stack [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Executing [s@macro-user-callerid:21] NoOp("DAHDI/i2/-49", "Using CallerID "" ") in new stack [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Executing [600@ext-group:2] GotoIf("DAHDI/i2/-49", "1?skipdb") in new stack [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Goto (ext-group,600,4) [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Executing [600@ext-group:4] Set("DAHDI/i2/-49", "__NODEST=") in new stack [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Executing [600@ext-group:5] Set("DAHDI/i2/-49", "__BLKVM_OVERRIDE=BLKVM/600/DAHDI/i2/-49") in new stack [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Executing [600@ext-group:6] Set("DAHDI/i2/-49", "__BLKVM_BASE=600") in new stack [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Executing [600@ext-group:7] Set("DAHDI/i2/-49", "DB(BLKVM/600/DAHDI/i2/-49)=TRUE") in new stack [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Executing [600@ext-group:8] Set("DAHDI/i2/-49", "RRNODEST=") in new stack [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Executing [600@ext-group:9] Set("DAHDI/i2/-49", "__NODEST=600") in new stack [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Executing [600@ext-group:10] GosubIf("DAHDI/i2/-49", "0?sub-rgsetcid,s,1") in new stack [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Executing [600@ext-group:11] Set("DAHDI/i2/-49", "RecordMethod=Group") in new stack [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Executing [600@ext-group:12] Macro("DAHDI/i2/-49", "record-enable,101-102-103-104-105,Group") in new stack [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Executing [s@macro-record-enable:1] GotoIf("DAHDI/i2/-49", "1?check") in new stack [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Goto (macro-record-enable,s,4) [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Executing [s@macro-record-enable:4] ExecIf("DAHDI/i2/-49", "0?MacroExit()") in new stack [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Executing [s@macro-record-enable:5] GotoIf("DAHDI/i2/-49", "1?Group:OUT") in new stack [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Goto (macro-record-enable,s,6) [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Executing [s@macro-record-enable:6] Set("DAHDI/i2/-49", "LOOPCNT=5") in new stack [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Executing [s@macro-record-enable:7] Set("DAHDI/i2/-49", "ITER=1") in new stack [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Executing [s@macro-record-enable:8] GotoIf("DAHDI/i2/-49", "1?continue") in new stack [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Goto (macro-record-enable,s,13) [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Executing [s@macro-record-enable:13] Set("DAHDI/i2/-49", "ITER=2") in new stack [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Executing [s@macro-record-enable:14] GotoIf("DAHDI/i2/-49", "1?begin") in new stack [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Goto (macro-record-enable,s,8) [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Executing [s@macro-record-enable:8] GotoIf("DAHDI/i2/-49", "1?continue") in new stack [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Goto (macro-record-enable,s,13) [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Executing [s@macro-record-enable:13] Set("DAHDI/i2/-49", "ITER=3") in new stack [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Executing [s@macro-record-enable:14] GotoIf("DAHDI/i2/-49", "1?begin") in new stack [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Goto (macro-record-enable,s,8) [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Executing [s@macro-record-enable:8] GotoIf("DAHDI/i2/-49", "1?continue") in new stack [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Goto (macro-record-enable,s,13) [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Executing [s@macro-record-enable:13] Set("DAHDI/i2/-49", "ITER=4") in new stack [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Executing [s@macro-record-enable:14] GotoIf("DAHDI/i2/-49", "1?begin") in new stack [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Goto (macro-record-enable,s,8) [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Executing [s@macro-record-enable:8] GotoIf("DAHDI/i2/-49", "1?continue") in new stack [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Goto (macro-record-enable,s,13) [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Executing [s@macro-record-enable:13] Set("DAHDI/i2/-49", "ITER=5") in new stack [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Executing [s@macro-record-enable:14] GotoIf("DAHDI/i2/-49", "1?begin") in new stack [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Goto (macro-record-enable,s,8) [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Executing [s@macro-record-enable:8] GotoIf("DAHDI/i2/-49", "1?continue") in new stack [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Goto (macro-record-enable,s,13) [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Executing [s@macro-record-enable:13] Set("DAHDI/i2/-49", "ITER=6") in new stack [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Executing [s@macro-record-enable:14] GotoIf("DAHDI/i2/-49", "0?begin") in new stack [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Executing [s@macro-record-enable:15] GotoIf("DAHDI/i2/-49", "0?IN") in new stack [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Executing [s@macro-record-enable:16] ExecIf("DAHDI/i2/-49", "1?MacroExit()") in new stack [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Executing [600@ext-group:13] Set("DAHDI/i2/-49", "RingGroupMethod=ringall") in new stack [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Executing [600@ext-group:14] Macro("DAHDI/i2/-49", "dial,15,tr,101-102-103-104-105") in new stack [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Executing [s@macro-dial:1] GotoIf("DAHDI/i2/-49", "1?dial") in new stack [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Goto (macro-dial,s,3) [Jun 19 11:48:10] VERBOSE[992] pbx.c: -- Executing [s@macro-dial:3] AGI("DAHDI/i2/-49", "dialparties.agi") in new stack [Jun 19 11:48:10] VERBOSE[992] res_agi.c: -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi [Jun 19 11:48:10] VERBOSE[992] res_agi.c: dialparties.agi: Starting New Dialparties.agi [Jun 19 11:48:10] VERBOSE[992] res_agi.c: dialparties.agi: Caller ID name is 'unknown' number is 'unknown' [Jun 19 11:48:10] VERBOSE[992] res_agi.c: dialparties.agi: Methodology of ring is 'ringall'[/code] Can anyone tell why/where it is being removed? Thanks! Wed, 19 Jun 2013 10:47:14 -0500 http://www.neomano.org/index.php/en/component/kunena/11-ayuda/123403-bloquear-llamadas-entrantes.html#123550 Hola... Gracias por responder. Ya lo probé, de hecho fue una de las primeras opciones que intenté. Leyendo la documentación (Comunicaciones unificadas con elastix vol1, y elastix without tears) encontré que sólo es factible si la ruta de entrada está definida para un canal SIP/IAX, y como lo que yo tengo instalados son canales PSTN, pues nada.... Sigo buscando alternativas. Wed, 19 Jun 2013 10:02:34 -0500 http://www.neomano.org/index.php/en/component/kunena/19-call-center/123401-callcenter-22.html#123549 That´s a feature that was in previous versions? Regards, Claudio Wed, 19 Jun 2013 09:30:51 -0500 http://www.neomano.org/index.php/en/component/kunena/19-call-center/123547-ivr-variable-to-external-url.html#123547 i there i am looking for the way to send an IVR number captured to the incomming campain external URL. the specific case..... the idea is that our customers digit they service contract account number in the IVR and when an agent take the call ELASTIX call our CRM software with the account number and pop-up with custumer info. any idea or work around on this will be apreciate tnks gmo solis Wed, 19 Jun 2013 09:27:00 -0500 http://www.neomano.org/index.php/en/component/kunena/116-security/123397-how-to-change-webadmin-a-mysql-root-password.html#123546 There is no reason for being identical... As for where to change it, 1) You can change admin pass for the web GUI in the Users tab as you stated 2) As for the mysql password, you can do it like this: $ mysqladmin -u root -p'oldpassword' password newpass But what you must have in mind, is that the passwords that you system really uses to work, are the ones stated at /etc/amportal.conf Regards, Claudio Wed, 19 Jun 2013 09:23:33 -0500 http://www.neomano.org/index.php/en/component/kunena/50-novatos/123545-interno-siempre-ocupado.html#123545 Buenas, tengo problemas con un interno en particular el cual siempre da ocupado si los demás internos lo llaman, pero este puede realizar llamada a los demás internos sin problemas. En el flash panel figura como desocupado y probé las siguientes cosas sin tener un resultado aun: -Reinicie el teléfono y lo configure nuevamente. -configure otro teléfono que funciona perfectamente con ese interno y sucede lo mismo, por lo que descartamos sea un error del aparato, sino de algo en elastix. -También probé borrar el interno del elastix, reiniciarlo, volver agregar el interno y sigue la misma situación. Que otro paso podría seguir? Muchas gracias. Wed, 19 Jun 2013 09:22:36 -0500 http://www.neomano.org/index.php/en/component/kunena/27-miscellaneous/123429-nfs-settings.html#123544 Hi, Can you be a little more specific? Don´t really know Vsphere more than it´s VmWare stuff... Regards, Claudio Wed, 19 Jun 2013 09:16:54 -0500 http://www.neomano.org/index.php/en/component/kunena/3-help/123381-updating-elastix-to-latest-version.html#123543 Also, you can check this cool note of Bob Fryer, the idea is always the same despite the different versions... http://www.elastixconnection.com/index.php?option=com_content&view=article&id=111&Itemid=119 Regards, Claudio Wed, 19 Jun 2013 09:14:56 -0500 http://www.neomano.org/index.php/en/component/kunena/3-help/79880-call-recording-management.html#123542 Indeed,that´s a good point. It all come to what you really need... For instance if you want to backup the monitoring folder in real-time and have it in a network drive also; It will be better to mount a shared drive in the PBX(for instance a nfs drive) for example in /mnt/recordings and under "Run after record" in the pbx tab under the General Configuration, put a script that copies the file to /mnt/recordings. That way you will have the recordings under /var/spool/asterisk/monitor, and also in the shared drive, so you have a backup and also you can still access to the recordings by the Web GUI. Regards, Claudio Wed, 19 Jun 2013 09:12:04 -0500 http://www.neomano.org/index.php/en/component/kunena/3-help/123518-automatic-answer.html#123541 Hi, What don´t you put that message in an IVR and later send it as a failover to a Voicemail? Regards, Claudio Wed, 19 Jun 2013 08:42:18 -0500 http://www.neomano.org/index.php/en/component/kunena/3-help/123454-gsm-gateway-voice-blue-not-registered-in-elastix.html#123539 here is my voiceblue configuration [code] [Gateway] ati3=ATI3, (c) alistar systems, 2001-2011 , [ M113 ] B-02.08 VOIP/GSM V-02.07.36rc5 VB, ati4=ATI4, SNumber: M113-0614770097, MacAddr: 00-50-C2-62-20-6A, Enabled: (disabled), Limited: 7871 hours (disabled), Network: (all), [System parametres] X30=#,#,# X31=7*,9# X32=*55,#55 X33=*33,#33 S70=192.168.100.19 S71=255.255.255.0 S72=192.168.100.1 S91=7,0 S98= X24=2222 X20=00.00,00.00 [Ethernet parametres] E01=0,8,4 E02=600,10,600 E03=10000,15000 E09=1 E10=192.168.100.254:5060 E11=192.168.100.254:5060 E14=192.168.100.254:5060 E16=0.0.0.0 E17=0.0.0.0:3478 E20=4,0 E23=1,0 E29=2,0 E80=voice,1234 E81=192.168.100.254 E82=sim_1 Codec=0 E08=0,0 E04=32,32 [GSM parametres] G09=1,0,0,0,4 G02=1,2,2 G04=0,3,3 G08=2,9,9,3 G103=0,2,2,2,2 G101= G06= G07= [Groups assignment] G00=1111 G90=1111 [Outgoing groups] S85= S86= G11=,2,0,0,1,1,1,0 G19=1,0,0,1,0,0, G21=,0,0,0,1,1,1,0 G29=1,0,0,1,0,0, G31=,0,0,0,1,1,1,0 G39=1,0,0,1,0,0, G41=,0,0,0,1,1,1,0 G49=1,0,0,1,0,0, G109= [Incoming groups] G91=0,3,3,10,1,, G95= G191=0, G99=0,0 G92=0,3,3,10,1,, G96= G192=0, G93=0,3,3,10,1,, G97= G193=0, G94=0,3,3,10,1,, G98= G194=0, G199= [Network list] N10=3/ N11=0,1,2,3,4,5,6,7,8,9 N19=,8 N20=/ N21=0,1,2,3,4,5,6,7,8,9 N29=,9 N30=/ N31=0,1,2,3,4,5,6,7,8,9 N39=,9 N40=/ N41=0,1,2,3,4,5,6,7,8,9 N49=,9 N50=/ N51=0,1,2,3,4,5,6,7,8,9 N59=,9 N60=/ N61=0,1,2,3,4,5,6,7,8,9 N69=,9 N70=/ N71=0,1,2,3,4,5,6,7,8,9 N79=,9 N80=/ N81=0,1,2,3,4,5,6,7,8,9 N89=,9 [Autorouting table] A= [Extension table] M= [Lcr table] L=1,0:00/24:00,1,0 [/code] my trunk [code] [sim_1] disallow=all fromdomain=192.168.100.19 host=192.168.100.19 type=peer port=5060 quality=yes allow=alaw allow=ulaw trunk=yes username=sim_1 fromuser=sim_1 secret=1234 canredirect=no dtmfmode=rfc2833 nat=auto insecure=port,invite context=from-internal permit=192.168.100.19/255.255.255.0 [/code] Wed, 19 Jun 2013 07:54:38 -0500 http://www.neomano.org/index.php/en/component/kunena/25-newbies-corner-/123537-dial-patten-and-dial-rules.html#123537 Hey, Can anyone pls help me understand what are the roles of the (prepend) and the prefix? and what do i need to do if some parts of the number that should go to the trunk shouldn't be dialed and some parts that aren't dialed should go to the trunk? for example. 0097241234567 this is a number that will need to go to the trunk the common user will call 041234567 meaning that the first part (00972) should be completed by the machine and the 0 in the dialed number 041234567 should be ignored by the trunk. what do you do in such case where there are some parts that should be completed by the system 00972 and some parts that are dialed by the user that should be ignored? thank you Wed, 19 Jun 2013 06:48:49 -0500 http://www.neomano.org/index.php/en/component/kunena/25-newbies-corner-/122873-number-in-do-not-call-list-show-inactive.html#123534 I changed /etc/php.ini, in lines 312 and 313 increase max_execution_time and max_input_time to 60 or more for example. it helped me but I did not try to add so many numbers as you. Give it a try and see what happens... Wed, 19 Jun 2013 05:48:36 -0500 http://www.neomano.org/index.php/en/component/kunena/3-help/123367-cdr-shows-many-calls-but-we-actually-receive-less.html#123532 Yourname, There are serious moves afoot to update/replace the forum system. As moderators, we have made it clear that this forum needs to be upgraded (images being one of the issues) as well as improved spam protection. Regards Bob Wed, 19 Jun 2013 04:56:38 -0500 http://www.neomano.org/index.php/en/component/kunena/79-general/121037-voicemail-notification-exchange-serveur.html#123531 Salut, je remets au goût du jour de thread car je n'ai pas eu le temps de retravailler dessus depuis. Je n'ai pas essayé les méthodes d'écrites sur le lien car au vu des résultats je peux pas trop me permettre de perdre des données ou autres. Pas contre apres avoir crée un connecteur de réception sur exchange et paramétré le remote smtp de elastix en indiquant mon serveur de mail, j'ai eu des erreurs du type "sasl authentication failed server offered not compatible mechanisms authentication for this type of connection" et du coup je ne trouve pas la combinaison qui marche!! Donc si quelqu'un a cherché a faire la même chose et a réussi, help!!! :) Wed, 19 Jun 2013 04:27:56 -0500 http://www.neomano.org/index.php/en/component/kunena/86-question-installations/122565-debut-dans-elastix.html?limit=10&start=20#123530 Wep je vais tester avec virtuel box moi avec le dernier iso on vera ce que ça donne Wed, 19 Jun 2013 04:27:40 -0500 http://www.neomano.org/index.php/en/component/kunena/46-others/117792-cisco-8961-video-not-working.html#123528 Hi, No I never resolved this issue and ended up returning the 8961s to the supplier. Apparently, the 8961 does not support SIP video but some video protocol which is supported through the use of a PC in conjunction with the phone. If you want video functionality on a phone with asterisk, I suggest you look away from Cisco and go with Granstream or Yealink... Wed, 19 Jun 2013 04:21:27 -0500 http://www.neomano.org/index.php/en/component/kunena/1-installation-issues/123490-compatibility-with-cisco-79xx-ip-phones.html#123526 asterisk is compatible with sccp. Check here http://chan-sccp-b.sourceforge.net/ basically, you compile/install the module. Load it into asterisk, and edit the /etc/asterisk/sccp.conf file I use this configuration for 2 cisco 7937g, and some 7921g wireless phones. The config is a bit messy, but it works. http://pastebin.com/03ybVZhA Wed, 19 Jun 2013 04:18:52 -0500 http://www.neomano.org/index.php/en/component/kunena/25-newbies-corner-/123488-badwith-allocation.html?limit=10&start=10#123519 and each call will be 64 kbps for the g711 meaning 5 calls will be 320 kbps right? a fax will also be 64kbps. Wed, 19 Jun 2013 03:41:08 -0500 http://www.neomano.org/index.php/en/component/kunena/3-help/123517-how-to-config-enterparise-service.html#123517 Hi, I want to install elastix server for ISP that could support over 300000 users , please let me know the conditions and requirement. Wed, 19 Jun 2013 03:37:16 -0500 http://www.neomano.org/index.php/en/component/kunena/84-le-coin-du-debutant/123414-touches-supervision-dune-extension.html#123512 Bonjour Jedrosik, Essaye de créer un context "blf" dans ton "extension_custom.conf" Puis ajoute "subscribecontext=blf" dans ton "sip_custom.conf" Puis un sip reload (ou reboot) Disons que ton gigaset a l'extension 222, ca donne : ----extension_custom.conf----- [blf] exten => 222,hint,SIP/222 -----sip_custom.conf------ subscribecontext=blf En console, tu fais un core show hints, le state "hiddle" doit changer Wed, 19 Jun 2013 02:48:20 -0500 http://www.neomano.org/index.php/en/component/kunena/85-aide/123157-besoin-support.html?limit=10&start=10#123511 bonjour, 1)si tu utilise un custom context, il faut avoir en tête que l'ordre des priorités des routes défini dans le menu "outbound route" ne s'applique que si le champs "priority" du context (paragraphe "outbound routes" )ont la même valeure. 2)Tu as réélement besoin de modifier la conf de la patton ? N'y touche pas si elle fonctionnait avant (en esperant que tu ai le backup de ta conf pour un retour arriere). 3)Si j'ai bien suivi le fil, il n'y avait pas de vlan initialement et tu avais du son quand tu emettais/recevait un appel. Peut être un vlan mal configuré qui ne tagg pas correctement le flux serait la cause de cette absence de son dans les 2 sens. (place tes equipements sur le vlan 1 par defaut pour t'en assurer..) Wed, 19 Jun 2013 02:30:57 -0500 http://www.neomano.org/index.php/en/component/kunena/33-a2billing/92972-easy-6-step-install-instruction-for-a2billing.html?limit=10&start=30#123508 Hi Great guide, I was wondering does this work with the latest version of Elastix? 2.4.0 Stable? Wed, 19 Jun 2013 02:19:29 -0500 http://www.neomano.org/index.php/en/component/kunena/84-le-coin-du-debutant/123417-creation-dutilisateur-create-user-elastix-call.html#123503 bonjour iwagg, Tes extensions sont elles enregistrées ? Entre en mode console avec un asterisk -r, puis sip show peers OU sip show peer "ton extension" Merci pour ton retour. Wed, 19 Jun 2013 01:31:37 -0500 http://www.neomano.org/index.php/en/component/kunena/31-asterisk/123439-dial-plan-not-working-properly-187-asterisk.html#123495 You not understanding MAIN, for me not important (hacked sip account) For me important understanding how was dialed number 810810*** via Dahdi if in web interface elastix write: THAT TRUNK NOT USING ANY ROUTE. Wed, 19 Jun 2013 00:29:54 -0500 http://www.neomano.org/index.php/en/component/kunena/3-help/123494-how-playbeak-the-custom-massege-if-extension-busy.html#123494 Hi Dear Friends How Define The Custom massege for my elastix that if call to my extension busy play this massege Please Help me Wed, 19 Jun 2013 00:27:09 -0500 http://www.neomano.org/index.php/en/component/kunena/3-help/123489-bandwith-allocation.html#123489 Hello, I would like to ask what download an upload rates will you need to have in order to make 2-5 simultanious calls? i understand it depends on the codecs and i dont mind using any available codec that might help ( by the way which is the best?) as long as i could hear good. is there any way to have a good quality without using the g729? Thank you Tue, 18 Jun 2013 22:50:19 -0500 http://www.neomano.org/index.php/en/component/kunena/31-asterisk/123377-dialplan-issue-i-think.html#123485 is this a SIP trunk? What is actually in the SIP trunk configuration file? sip.conf? Tue, 18 Jun 2013 20:14:20 -0500 http://www.neomano.org/index.php/en/component/kunena/1-installation-issues/123064-no-packages-marked-to-update-problem.html#123484 That should be fine. This is a weird issue. There hasnt been anything significant changed since you installed so i wouldnt worry too much at this stage Tue, 18 Jun 2013 20:11:19 -0500 http://www.neomano.org/index.php/en/component/kunena/3-help/123413-call-transfer.html#123483 Check in your feature codes and see what the blind transfer feature is set to. Tue, 18 Jun 2013 20:05:09 -0500 http://www.neomano.org/index.php/en/component/kunena/20-elastix-community-/123252-skype-for-asterisk-eol-2013-07-26.html#123480 I believe they offer a SIP service you can connect your asterisk system with onto the skype network. Have you tested this? Tue, 18 Jun 2013 20:00:46 -0500 http://www.neomano.org/index.php/en/component/kunena/25-newbies-corner-/123406-incoming-calls-picotando.html#123479 The file is generated normally with asterisk. If you can SSH into the machine then you can use a program like WinSCP to download the log file. Tue, 18 Jun 2013 19:59:22 -0500 http://www.neomano.org/index.php/en/component/kunena/11-ayuda/90092-cisco-7940-no-registra-y-otros-tlfs-si.html#123478 yo también tengo el mismo problema con esos teléfonos Tue, 18 Jun 2013 19:29:24 -0500 http://www.neomano.org/index.php/en/component/kunena/11-ayuda/123123-tiempo-de-marcacion-largo-en-2-pbx-mediante-iax2.html#123474 ing_mohammed escribió: hola comunidad, quisiera saber si a alguien le ha pasado la siguiente falla, tengo 3 servidores elastix enlazados mediante iax2 y dundi, el problema que estoy presentando es el siguiente: tengo configurado un teléfono snom 870 con dirección de sipserver primario(pbx a) y secundario(pbx b), cuando yo marco de una extensión del (pbx c) hacia el snom suena rápido la marcacion, pero cuando desconectamos el (pbx a) y el teléfono automáticamente se registra al (pbx b), se realiza de nuevo la marcacion desde el (pbx c) pero tarda un lapso de 30seg a 40se en lo que iax2 detecta que el pbx a esta abajo y envía la llamada al pbx b. como puedo hacer para mejorar ese tiempo de espera, habrá algún parámetro que se pueda modificar en el equipo??? gracias por la ayuda Tue, 18 Jun 2013 17:41:31 -0500 http://www.neomano.org/index.php/en/component/kunena/58-freepbx/123473-configurar-toncales-y-salidas-para-wildcard-tdm400.html#123473 Necesito configurar elastix con la tarjeta Wildcard TDM400P REV E/F Board 5, esta tiene 4 tarjetas para líneas de entrada , 1- línea fija,2- enlacae móvil,3-enlace móvil,4-enlace móvil, cuando la detecta la configura como una zap única que engloba las 4 líneas, con lo cual las llamadas entrantes todo ok.. no necesito filtrar, ya lo desvió a las extensiones a gusto... el problema esta cuando quiero que, cuando se llame a un fijo salga por la línea 1, cuando se llame a un móvil que salga por la línea 2, que cuando se llame a un 800 por la línea 3, etc.. alguien me puede guiar.. Pd: he hecho pruebas de todo tipo y siempre funciona cuando le digo que salga por la línea 1 si lo configuro como dahdi trunk , pero cuando le digo que salga por la 2,3.. me dice que destino inaccesible.. Tue, 18 Jun 2013 16:33:54 -0500 http://www.neomano.org/index.php/en/component/kunena/11-ayuda/123469-troncal-sip-alcatel-y-elastix.html#123469 Buenas Gente del foro, les comento el problemita que tengo. Tengo echo un trunk sip entre el alcate y el elastix 2.4.0 El alcatel maneja la E1 que tengo, existen internos alcatel e internos elastix para gente de afuera. Cual es el tema cuando yo llamo a un numero ejemplo 4148100 quiero que la ruta entrante de el elastix me lo direccione al interno 800 del elastix y si llamo al 4148105 que me lo direccione al interno 851 del elastix. La cosa esta que cuando pongo el 4148100 en DID al elastix le pasa el interno 8100 que no existe. Lease quiero que en la ruta entrante se llame al 4148100 y el set destination lo tire al interno seleccionado. Gracias por la ayuda !!! Tue, 18 Jun 2013 15:01:58 -0500 http://www.neomano.org/index.php/en/component/kunena/19-call-center/123161-abandoned-calls-during-outgoing-campaing.html#123465 have you checked the second edition manual, it has most of these outlined when running predictive or have enable over commit on you run the chance of having more calls connected than agents available, dialer should stop when no agents are available however may have dialed 2 calls for the last time an agent went to available status this is an intended way some call centers want to run to maximize productivity if you don't disable predictive the enable over commit, limit the number of line you use in an outbound campaign You can get the results best for you if you adjust the settings. You may want to consider the option of using custom recordings in your MOH. That is how you can play advertising or info messages during the hold time. This option could also help with your abandon rate. Tue, 18 Jun 2013 13:30:04 -0500 http://www.neomano.org/index.php/en/component/kunena/25-newbies-corner-/32680-user-non-admin-audit-calls.html?limit=10&start=20#123464 Hi Jorge I have Elastix 2.3. I added: if(isset($arrGroup['Reports'])) $is=true; I can do a filter in the CDR but no data is displayed. I log in as administrator, make the same filter and I got data. What am I missing? Regards Tue, 18 Jun 2013 13:28:12 -0500 http://www.neomano.org/index.php/en/component/kunena/1-installation-issues/122142-uelastix-web-interface-problem.html#123460 Aiiar wrote: help please I'm no expert..... But poking around... The issue is that the permissions on the distribution are partially messed up. One place is /var/lib/php/sessions. I chowned that to apache:apache. This was blocking session_start etc as specified by /etc/php.ini Specific to this problem..... /var/lib/mysql needs to be chowned (-R) to mysql:mysql I also re-installed the mysql server but I don't think that fixed anything. service msyqld start If you look in the /var/lib/ directory and you see numbers for the user and group, you know that they need to be fixed. Cheers, b Tue, 18 Jun 2013 12:00:15 -0500 http://www.neomano.org/index.php/en/component/kunena/26-tips-and-tricks/123385-una-guia-para-usar-a2billing-correctamente.html#123458 Si buscas por aca en el foro encontraras la respuesta, ahorita no tengo el post a la mano, pero hay uno en donde explican detalladamente como hacer correrlo, Saludos ! Tue, 18 Jun 2013 11:36:12 -0500 http://www.neomano.org/index.php/en/component/kunena/5-new-features/122575-record-the-event-q180-ringingq-in-cdr.html#123452 Nobody? please!! Tue, 18 Jun 2013 09:57:18 -0500 http://www.neomano.org/index.php/en/component/kunena/59-a2billing/123316-failed-to-authenticate-on-invite-to-q1136550771q.html#123451 Solucioné este error, cambié de proveedor: [provider-ip] username=7129982280 type=friend secret=9940672206 host=89.14.184.31 fromuser=7129982280 context=provider allow=g729 trustrpid = yes sendrpid = yes directmedia = no Con voicetrading falla: [voicetrading] disallow=all allow=g729,alaw,ulaw authuser=user fromdomain=voicetrading.com fromuser=user host=77.72.169.131 insecure=invite,port nat=yes qualify=yes secret=pass sendrpid=yes type=peer username=user context=voip Tue, 18 Jun 2013 09:41:10 -0500 http://www.neomano.org/index.php/en/component/kunena/99-autre/61265-configurer-un-fax-sur-elastix.html?limit=10&start=10#123446 Salut. Peut-être serait-ce le moment d'en faire un au propre liée à la version 2.4? Si tu as un moment, ce serait cool. Je sais que la démarche est décrite sur la doc Elastix Easy page 84 si ma mémoire est bonne. Tue, 18 Jun 2013 08:42:21 -0500 http://www.neomano.org/index.php/en/component/kunena/1-installation-issues/112815-installation-issue-of-call-center-in-30-alpha.html#123443 I have the same trouble here with latest ALPHA release and I have found no settings or options where to change this. How can I download the add-ons to the box? Tue, 18 Jun 2013 08:03:55 -0500 http://www.neomano.org/index.php/en/component/kunena/10-success-stories/120664-country-wide-deployment-in-south-africa.html#123434 No real problems. Have it hooked up to MS Active Directory as well. Allowing everyone in the organisation to use the same credentials for email/Server logins/IM is a godsend. Only had a problem with one of the openfire addons, fastpath. Needed to load a version that was compatible with openfire 3.7.1 We use fastpath to deal with "Live Chat" queries. Tue, 18 Jun 2013 06:16:51 -0500 http://www.neomano.org/index.php/en/component/kunena/3-help/123391-dialplan-disappeared.html#123433 Did you do any custom editing of your dial plan? Tue, 18 Jun 2013 06:15:17 -0500 http://www.neomano.org/index.php/en/component/kunena/3-help/123394-hangup-chanspy-channel.html#123431 Sometimes soft hangups dont work. You could try updating your version or you can restart asterisk when you have problem channels Tue, 18 Jun 2013 06:12:04 -0500 http://www.neomano.org/index.php/en/component/kunena/35-openfire/123378-dass-die--der-punkt.html#123426 Please remove Tue, 18 Jun 2013 06:03:52 -0500 http://www.neomano.org/index.php/en/component/kunena/116-security/32999-firewall-what-ports-need-to-be-open-what-not-.html#123423 About the RTP range I would also suggest lowering that a bit to something a lot less. That is still 10K ports open. In theory the best way to secure is not to have open ports at all. But only trusted IP addresses. Then if you have external/roaming phones use vpn. Tue, 18 Jun 2013 05:02:16 -0500 http://www.neomano.org/index.php/en/component/kunena/25-newbies-corner-/122831-extensions-are-in-use-aused-channelspya-here-music.html#123419 Just to note if you have been hacked 1 time its always advisable to do a complete system re-install (normally for those who don't really know Linux enough to dig through it like me). Because if someone had access they could have implanted a back door trojan that only activates when you least expect. This is coming from experience and a few thousands of USD lost to hackers. Tue, 18 Jun 2013 03:57:35 -0500 http://www.neomano.org/index.php/en/component/kunena/18-modules/101762-solved-elastix-230--h323.html#123415 Thank you very much Bob. Finally i can see the h323 channel installed. The next step to make the channels work together. If i put the CLI with core shoe channeltypes i get: "Type Description Devicestate Indications Transfer ---------- ----------- ----------- ----------- -------- USTM UNISTIM Channel Driver no yes no OOH323 Objective Systems H323 Channel Driver no yes no Phone Standard Linux Telephony API Driver no yes no EXTRA GSM/CDMA Telephony Driver FOR Asterisk w no yes no ConfBridge Conference Bridge Recording Channel no no no Agent Call Agent Proxy Channel yes yes no MGCP Media Gateway Control Protocol (MGCP) yes yes no Bridge Bridge Interaction Channel no no no Jingle Jingle Channel Driver no yes no DAHDI DAHDI Telephony Driver w/PRI & SS7 & MFC yes yes no SIP Session Initiation Protocol (SIP) yes yes yes IAX2 Inter Asterisk eXchange Driver (Ver 2) yes yes yes MulticastR Multicast RTP Paging Channel Driver no no no Gtalk Gtalk Channel Driver no yes no Local Local Proxy Channel Driver yes yes no ---------- 15 channel drivers registered." So the ooh323 channel is available but maybe isn´t active. Th string in the custom trunk it´s: OOH323/$OUTNUM$@ip_remote_address:remote_port But stilll doesn´t works. Any idea? Thank so much for your help. Tue, 18 Jun 2013 03:29:09 -0500 http://www.neomano.org/index.php/en/component/kunena/3-help/123208-pbxmate-information.html#123412 The PBXMate has a free trial. Why not give it a try ? Also, looking at the publisher site (SoliCall) I see audio samples of advanced noise reduction software (http://www.solicall.com/blog/personalized-noise-reduction-software/). Tue, 18 Jun 2013 02:04:16 -0500 http://www.neomano.org/index.php/en/component/kunena/1-installation-issues/123216-asterisk-1821-jumped-to-114.html#123411 thanks for advance, will do it today .... asterisk died code 1 .... res_fax_digium.so in my case :) thanks mydigia for method :) now it works just fine Tue, 18 Jun 2013 01:56:40 -0500 http://www.neomano.org/index.php/en/component/kunena/46-others/87128-new-cisco-504g-phones-elastix-wont-see-them.html?limit=10&start=40#123409 First post here to post a massive THANKS to Bob for the documentation. Just two points to help things along: 1. The paloSantoFileEndPoint.class.php additions include the SPA514G, but the SQLite instructions don't. I added the following two lines to the list for SQLITE and all worked fine: [code]sqlite3 /var/www/db/endpoint.db "insert into mac (id_vendor,value,description) values (24,'A4:93:4C','Cisco SPA514G');" sqlite3 /var/www/db/endpoint.db "insert into model (name,description,id_vendor) values ('SPA514G','SPA514G',24);" [/code] 2. Got another MAC address for the list for the SPA504G from our old stock: E0:5F:B9 Thanks again for the excellent resource! Tue, 18 Jun 2013 01:03:28 -0500 http://www.neomano.org/index.php/en/component/kunena/11-ayuda/123405-problemas-con-23-troncales.html#123405 Buenas noches, hace unos días instale la versión 2.3, y al principio todo de maravilla solo que ahora me esta dando unos problemas al igual que la 2.4(la cual tuve que cambiar) resulta que yo utilizo troncales análogas y por alguna razón en ocaciones cuando alguna persona llama y cuelga se queda como la llamada "pegada" en la linea osea que la troncal se bloquea como que la linea queda en uso pero a la vez no esta en uso y no se puede usar la troncal, esto me esta pasando muy a menudo en la distribución 2.3 y no es algo bueno para el lugar donde coloque la central, si alguien me podría ayudar se la agradezco. Mon, 17 Jun 2013 23:50:44 -0500 http://www.neomano.org/index.php/en/component/kunena/50-novatos/123404-comunicaciones-unificadas.html#123404 Estimados amigos, Me gustaría saber si existe alguna documentación donde se explique el dimensionamiento de Elastix como servidor de comunicaciones unificadas. Generalmente el dimensionamiento se lo hace referente a telefonia IP. En este caso me gustaria hacer una analisis completo de los recursos de procesamiento que necesito en mi servidor para activar cada servicio adicional (Mensajería instantanea, correo electronico, Video conferencias y FAX). Esperaria me puedan ayudar con este tema Muchas gracias Mon, 17 Jun 2013 22:42:38 -0500 http://www.neomano.org/index.php/en/component/kunena/72-otros/123324-telefonos-fanvil.html#123402 Estimado, los dos fabricantes son Chinos: El modelo C62 Fanvil equivale al T28P de Yealink El modelo C58P Fanvil equivale al T22P de Yealink El modelo C56P Fanvil es superior al T18P pero menor al T20P Las guías de programación se encuentran disponibles en la página de Elastix, para estos modelos Fanvil (C62, C58P y C56P): "Guías de Configuración EHCP" Mon, 17 Jun 2013 17:12:38 -0500 http://www.neomano.org/index.php/en/component/kunena/50-novatos/123398-problema-con-grandstream-gxw4108-y-el-ivr.html#123398 Buenas Tardes, Disculpen soy nuevo en el foro y necesito de su ayuda, Lo que sucede es que compre un Grandstream GXW4108, lo configure como explica este tutorial : http://blog.hardmax.com.pe/2008/10/30/configuracion-grandstream-gxw4104-en-freepbx/ Lo configure y todo y coloque una linea analoga, genere la ruta entrante como any DID/any CID, y coloque el IVR que ya habia configurado. pero cuando marco primero suena un tono como de entrada de llamada luego suena otro pero queda ahi no sale IVR pero si en el 2 tono marco una extension si entra la llamada pero nunca suena el IVR. Gracias, Mon, 17 Jun 2013 16:14:03 -0500 http://www.neomano.org/index.php/en/component/kunena/84-le-coin-du-debutant/123374-elastix-entre-deux-site-distant.html#123396 Hmmm ok, c'est comme un bridge. Peut-être qu'il faut se concentrer sur un problème de QoS. Vérifier ce qui se passe avec un SIP et RTP debug. Mon, 17 Jun 2013 13:48:59 -0500 http://www.neomano.org/index.php/en/component/kunena/26-tips-and-tricks/123384-habilitar-codec-vp8-en-elastix.html#123384 Hola amigos, me pregunto si es posible habilitar el codec de vídeo VP8 en Elastix, parece ser que este codec de vídeo ofrece una calidad muy buena y esta siendo usado en nuevas tecnologías HD. Además lo incluye el softphone Gratuito GPL para Android Linphone y nuevas soluciones para vídeo llamadas WebRTC en Chrome, Firefox incluso Skype. Un saludo. Mon, 17 Jun 2013 08:56:31 -0500 http://www.neomano.org/index.php/en/component/kunena/3-help/28293-nvfax-64bit-files.html?limit=10&start=30#123383 Hello again, These are the nvfax modules for 64bit Elastix 2.4 with asterisk 11.4. Please test them and inform me of any problems. Best Regards, Panagiotis Liakopoulos http://www.elastix.org/images/fbfiles/files v_fax_64bit.rar Mon, 17 Jun 2013 08:35:04 -0500 http://www.neomano.org/index.php/en/component/kunena/79-general/123370-conception-delastix.html#123373 Merci Mon, 17 Jun 2013 06:36:23 -0500 http://www.neomano.org/index.php/en/component/kunena/3-help/123372-restrict-inbound-by-cell-phone-number.html#123372 Hi, My box is running very successfully for the last one year. Now my requirement is as follows.... Am using Digium TEXXX 2 PORT card with 2 PRI's connected to it. How can I configure the server to drop all the other calls, and should allow only certain mobile numbers (India +91xxxxxxxxxx)? Your advice is very much appreciated. -- Bain Mon, 17 Jun 2013 06:33:51 -0500 http://www.neomano.org/index.php/en/component/kunena/18-modules/123366-no-filter-button-in-qcdr-reportq-sub-module.html#123366 I have created a non-admin user but given that user rights on "Reports" module. The user once login, has access to the CDR-Report sub module but there is no filter buttons visible and as such there is no generated CDR-Report. Mon, 17 Jun 2013 02:36:02 -0500 http://www.neomano.org/index.php/en/component/kunena/31-asterisk/123365-calls-not-distributed-to-static-queue-agents.html#123365 I have a random problem whereby a call is not distributed to a static agent even if the agent is logged in the queue. Problem has started after updating to asterisk 1.8.21 Elastix version: 2.4.0 Mon, 17 Jun 2013 02:26:29 -0500 http://www.neomano.org/index.php/en/component/kunena/26-tips-and-tricks/94032-google-text-to-speech-and-speech-to-text.html?limit=10&start=20#123364 yair_pc wrote: i install all rpm when i dial 1235 or 1236 i hear the message "Say something in English, when done press the pound key" and the system do not wait for my answer the asterisk say "failed to get speech data" Hi , I know its too late to ans lol, but i found that it happens when u call from a SIP phone, use a real-phone and it will work.. i hope it will help the new readers :P Mon, 17 Jun 2013 01:21:03 -0500 http://www.neomano.org/index.php/en/component/kunena/72-otros/123363-ht-503-no-corta-cuando-llamo.html#123363 Muy buenas. El escenario es el siguiente: Tengo un HT 503 con elastix 2.0.3 y unos teléfonos Yealink funcionando a la perefección. El único problema es que cuando realizo una llamada saliente y el receptor de la llamada corta el HT 503 no lo detecta y deja la línea abierta. Tengo que cortar yo desde el teléfono desde el cual realicé la llamada. ¿Es raro no? Ya que no sucede lo mismo cuando la llamada en entrante. Mon, 17 Jun 2013 00:21:49 -0500 http://www.neomano.org/index.php/en/component/kunena/5-new-features/123201-sending-sms-messages.html#123362 I am trying to set up an emergency warning systems, so most of the time the volume will be zero. In case of an emergency, I would need to send a batch of 500 messages for each status update. Mon, 17 Jun 2013 00:20:51 -0500 http://www.neomano.org/index.php/en/component/kunena/116-security/123351-is-there-any-way-to-encrypt-calls-in-elastix.html#123361 Hi To have a big security, you may use a SIP TLS and SRTP encryption. But your softphones should be able to support these protocols. Otherwise, you may use a VPN like OpenVPN. Sun, 16 Jun 2013 23:22:42 -0500 http://www.neomano.org/index.php/en/component/kunena/3-help/123321-elatix-and-linux-time.html#123358 Log into the Linux console and set the date with the date command. This will make sure it is correct Sun, 16 Jun 2013 16:20:19 -0500 http://www.neomano.org/index.php/en/component/kunena/3-help/123293-purging-old-data.html#123354 You don't have a problem with that, it works as in any linux system like implementation, you don't have to worry for the logs (Yes, they will increase, but not to a critical state) or any reboot to keep it working. So in your case, you should be seeing the system running smoothly in 2 weeks. Regards, Claudio Sun, 16 Jun 2013 14:03:09 -0500 http://www.neomano.org/index.php/en/component/kunena/3-help/108052-cant-get-addons-from-addon-tab.html#123344 ==================SOAP error===================== yum install php-soap extension=soap.so service httpd restart ================================================= Sun, 16 Jun 2013 04:25:33 -0500 http://www.neomano.org/index.php/en/component/kunena/117-asterisk-a-programming/123298-advanced-developer-guide.html#123342 This is something I am interested in also Sun, 16 Jun 2013 02:08:32 -0500 http://www.neomano.org/index.php/en/component/kunena/3-help/123279-avaya-elastix-migrations.html#123338 Check the configurations and make sure you can use all functions like blf. You don't want a phone that can only register and make calls. Sun, 16 Jun 2013 01:58:31 -0500 http://www.neomano.org/index.php/en/component/kunena/25-newbies-corner-/70411-the-system-can-not-connect-to-the-web-service.html#123336 ==================SOAP error===================== yum install php-soap extension=soap.so service httpd restart ================================================= Sun, 16 Jun 2013 01:23:11 -0500 http://www.neomano.org/index.php/en/component/kunena/50-novatos/123302-cuando-recibo-llamada-descuelgo-y-me-da-tono.html#123335 Hola Encui, Suena extraño, podrías entrar al cli de asterisk (con asterisk -rvvv) y copiar el texto que aparece cuando realizas la llamada? Sat, 15 Jun 2013 22:52:44 -0500 http://www.neomano.org/index.php/en/component/kunena/10-success-stories/98971-multiple-google-voice.html?limit=10&start=130#123328 Sir Sam, I did the requested and sir it still doesn't work. I see that since I last posted that I have ver 2.4.0.1 running and have noticed that now I have to reboot the machine to even get GV to work at all once it goes to sleep. I finially have some spare time this weekend so I'm going to clean the dust off a stored machine and install an earlier version (like 1.6) and go from there as I have read that it appears that GV works well before 2.0. Have a great day Sir, amphibian Sat, 15 Jun 2013 17:53:11 -0500 http://www.neomano.org/index.php/en/component/kunena/53-trucos/110406-extensiones-remotas.html#123325 Hola: Muchachos una pregunta, logre hacer que las extensiones remotas se registren, hay todo bien. configure un telefono IP desntro de mi misma res pero con la IP publica, y funciono perfecto. pero cuando configuro telefonos o equipos desde fuera (otras redes) los equipos registran pero el audio es nulo. timbrar y levantan el auricular pero no se escucha nada. Adicional a esto, parece que mi ruter toma esa llamada entrante como un ataque y lo bloquea. esto es una supocicion. puede suceder? lo digo por que despues de intentar llamar desde afuera hacia mi oficina, se cane las llamadas salientes de mi oficina. un saludo. Carlos A Sat, 15 Jun 2013 15:52:55 -0500 http://www.neomano.org/index.php/en/component/kunena/3-help/122982-caller-id-and-gfwt.html#123320 Solved i have added the following 2 lines to my chan_dahdi.conf cidsignalling=dtmf cidstart=polarity :) Sat, 15 Jun 2013 11:41:34 -0500 http://www.neomano.org/index.php/en/component/kunena/57-asterisk/123319-mejoras-en-los-cdr-y-llamadas-y-usuarios.html#123319 Hola a todos. Estaba viendo el sistema de CDR de elastix, ojala en futuras actualizaciones mejores los CDR del elastix y los hagan mas completos, sobre todo a la hora de facturación para poder crear importes por extensiones o grupos de extensiones, poder hacer mas opciones que mejoren la información obtenida. Tengo varias dudas a ver si alguien me las puede resolver: 1.- Cuando creo un usuario para que accedea al elastix para ver las estadísticas en la opción de grupos le creo permisos para ver los CDR el voice mail y las grabaciones de llamadas. El problema es que si le activo el sumary del CDR o llamadas perdidas, no reconoce la extensión y muestra todo. Saben alguna forma para evitar esto ? Por otro lado seria interesante que un usuario pueda entrar y ver las estadísticas de diferentes extensiones y no solo las de una única extensión ya que un usuario puede tener muchas mas ( por ejemplo una centralita con 10 o xx extensiones ) hay alguna forma de que un usuario pueda ver todas sus extensiones ? Si esto no existe ya seria buenísimo que lo implementaran en actualizaciones, ya que es importante y facilita que un solo usuario vea las extensiones deseadas y no un usuario por extensión. También en los CDR, seria genial el poder indicar a tiempo real que el sistema calcule por ejemplo el total de minutos que ha generado una extensión cuando el numero comienza por 6XXXX o 7XXXX ( en España es móvil ) y cuando por otro numero con varias opciones, asi se podría ver el consumo de cada extensión por el numero de minutos a un rango determinado de números llamadas y ver si se ha pasado o no de su limite. Otra idea es poder poner en cada extensión o grupo de extensiones o incluso usuarios con varias extensiones un limite de minutos y si se supera que el sistema avise ya sea en la propia web en algún apartado de llamadas sobrepasadas o algo asi o por email. Asi se podría avisar al cliente cuando su extensión llega al 80% del total de minutos indicados, o al 90% o se bloquea por ejemplo . Si se pudiera hacer esto estaría muy bien. Bueno ideas sobre los CDR tengo muchísimas el problema es que realmente no se si se pueden hacer e integrar con elastix, que imagino que si se podrán lo que hay que ver es el coste que lleva y si lo harán :D Sat, 15 Jun 2013 10:56:26 -0500 http://www.neomano.org/index.php/en/component/kunena/57-asterisk/117948-mensaje-automatico.html#123318 Hola Churtado, pudiste solucionar el problema ? si pones el código que utilizas para realizar las llamadas automáticas puedo echarle un vistazo a ver donde este el problema. Sat, 15 Jun 2013 10:48:17 -0500 http://www.neomano.org/index.php/en/component/kunena/57-asterisk/115020-cdr-obtener-llamadas-entrantes-y-salientes.html#123317 Cmbv, soborno tiene razón, no obstante para lo que pides lo tienes fácil si le añades a la extensión un nombre de cuenta. al crear o modificar la extensión sip te aparece un campo que es nombredecuenta(Acountname) o algo asi hay pones lo que quieras ( yo particularmente utilizo el numero de la extensión ) asi a la hora de filtrar puedes filtrar por ese campo y te da todas las llamadas salientes y entrantes de esa extensión. También si le das a sumary te dira por extensión las llamadas entrantes y salientes, aunque una cosa si he visto es que el sumary si se le añade a un usuario, no le muestra solo su extensión, salen todas las extensiones, por lo que no se si es un bug o que no se han dado cuenta que al crear un usuario para que pueda acceder con permisos a los cdr y el summary aunque le indiques que su extensión es la 100 en el sumary aparecen todas. La verdad que seria recomendable en cuanto a los CDR ampliar mejoras para poder hacer mas fácil los CDR e incluso la sección de usuarios. Nosotros tenemos creados usuarios donde se pueden ver los CDR y las grabaciones, pero no podemos darles la opcion del sumary a los usuarios que seria interesante ni el de llamadas perdidas porque no muestra un usuario salen todos. También seria buenísimo mejorar el sistema de billing del elastix para poder personalizarlo por usuario Y una cosa que seria muy buena es poder limitar el numero de minutos mensuales consumidos de alguna forma, por ejemplo para limitar que la extensión 100 llame un máximo de 1000 minutos a móvil y 5000 a fijo por ejemplo o que si pueda llamar a un determinado rando de números pero a otros no para evitar llamadas internacionales o caras. Dee cho voy a crear un post preguntando esto porque me suena interesante a ver si alguien sabe como hacerlo :D Sat, 15 Jun 2013 10:42:03 -0500 http://www.neomano.org/index.php/en/component/kunena/31-asterisk/74453-cannot-register-cisco-7942-to-asterisk.html?limit=10&start=50#123313 Hello. I have same problem Please send me the instructions an files to mohammad.sadeghijula@gmail.com. Thank you very much. Sat, 15 Jun 2013 05:58:09 -0500 http://www.neomano.org/index.php/en/component/kunena/52-problemas-de-instalacion/122766-instale-elastix-en-una-notebook-y-no-tengo-acceso.html#123310 Muchas gracias jgutierrez por responder. Hice lo que me sugeriste y los parámetros están bien. El firewall desactivado. No entiendo por qué se da esta falla. Creo que tendré que reinstalar. Tal vez haya quedado en el camino algo por instalarse. Sat, 15 Jun 2013 02:42:36 -0500 http://www.neomano.org/index.php/en/component/kunena/32-freepbx/121142-elastix-24-and-freepbx--210-.html#123308 Hi One thing that FreePbx 2.10 fix is the way callerid is generated in the sip_additional.conf . As an example, for the extension number 3111 with the name John Doe Instead of having callerid= 2.10 will generate callerid=John Doe That solves the problem of displaying the word "device" on the phone when calling an extension. Is there a way to correct that behaviour without 2.10 ? Thanks Onagan Fri, 14 Jun 2013 23:29:33 -0500 http://www.neomano.org/index.php/en/component/kunena/25-newbies-corner-/122762-voip-provider-problem.html#123307 Hi. YOU nailed it !!!!!!!! Yes it works now, I have put "peer" in the details, remove outbound proxy line and put that line in the sip config and voila it works !!!!!!! also you have to put host sip20.videotron.ca not v20.videotron.ca and fromdomain is v20.videotron.ca Now I am able to remove my 3CX system and replace it by better one Great. Other question: how do I secure my phone server from “friendly scanner” that try's to hack sip server before I activate this new server. In 3CX that have a automatic system that block the IP of those scanner. Quickly. You may use Deny/Premit parameters for each trunk and extensions. These parameters define from where it's permit to connect. Next, you must enable the firewall and put some good rules. For example, you could accept your trusted IP address and dropping the others. There's several threads about that on the forum. Please, use the search engine on this forum to read them. Read the Bob's document about the security too. Fri, 14 Jun 2013 23:00:04 -0500 http://www.neomano.org/index.php/en/component/kunena/3-help/123306-elastix-empieza-a-cobrar-antes-de-que-contesten.html#123306 Hola amigos del foro, hace tiempo he instalado un servidor elastix y me ha funcionado de maravilla, lo unico que no he podido cuadrar es que cuando realizo una llamada de mi linea fija el servidor comienza a facturar desde que empieza a timbrar y no desde que contesta, cosa que no ocurre cuando saco una llamada por un proveedor sip, alguien me podria ayudar, gracias Fri, 14 Jun 2013 21:29:25 -0500 http://www.neomano.org/index.php/en/component/kunena/19-call-center/123303-how-to-relaunch-the-function-dologin.html#123303 Hi, is there a way to re-launch the javascript function do_login() after receiving the message: Lost connection to server (SSE), retrying... its possible to add a button in the agent_console to launch it or can I just call the function below the code? [code]function do_checkstatus() { params = { menu: module_name, rawmode: 'yes', action: 'checkStatus', clientstate: estadoCliente }; if (window.EventSource) { params['serverevents'] = true; evtSource = new EventSource('index.php?' + $.param(params)); evtSource.onmessage = function(event) { manejarRespuestaStatus($.parseJSON(event.data)); } evtSource.onerror = function(event) { mostrar_mensaje_error('Lost connection to server (SSE), retrying...'); } do_login(); ....... [/code] Best Regards. Fri, 14 Jun 2013 18:25:07 -0500 http://www.neomano.org/index.php/en/component/kunena/25-newbies-corner-/123179-ssh-on-uelastix-cant-connect.html#123301 ok i find it login is root and no admin it s my fault, thank you :P Fri, 14 Jun 2013 17:57:46 -0500 http://www.neomano.org/index.php/en/component/kunena/11-ayuda/123295-falla-con-transferencia.html#123295 Estimados hay una falla con la interfas web hago una llamada desde el addressb book es decir del directorio telefonico del elastix y no hace la tranferrencias. que pordria der ? Fri, 14 Jun 2013 14:26:06 -0500 http://www.neomano.org/index.php/en/component/kunena/84-le-coin-du-debutant/123099-configuration-de-fax-sous-elastix.html#123291 Salut SLIM et bienvenue sur notre forum Elastix. :) Tout est dans les différentes doc ici: http://www.elastix.org/index.php/en/product-information/manuals-books.html Prendre par exemple Elastix Easy page 84. Bien sure le forum est là en aide, mais il faut savoir chercher. Elastix est une des distributions la plus documentée. N'hésitez pas à télécharger les documents et éplucher le fonctionnement du serveur. Il y a aussi le moteur de recherche du forum. Car pensez bien que vous n'êtes pas les seuls à s'être demandé comment configurer un serveur fax! Donc c'est forcément dedans. Sans parler qu'il y a l'aide contextuelle dans la partie FAX du menu Elastix GUI! Fri, 14 Jun 2013 11:43:11 -0500 http://www.neomano.org/index.php/en/component/kunena/11-ayuda/51955-web-no-funciona.html?limit=10&start=10#123290 Una solución es que el password de MySql se alla cambiado y por tal motivo no carga la interfase web el archivo que cambie es /etc/elastisx.conf y cambie el mysqlrootpwd= por el que le había puesto anteriormente. saludos Fri, 14 Jun 2013 11:29:52 -0500 http://www.neomano.org/index.php/en/component/kunena/1-installation-issues/119203-missing-dependency-perlasteriskagi.html#123289 OK, thanks for your help I will try to do, just frustrate me to found this issues and there is no documentation about how to work around the elastix rpm installation. Fri, 14 Jun 2013 11:26:18 -0500 http://www.neomano.org/index.php/en/component/kunena/11-ayuda/123286-account-code.html#123286 Hola que tal Buen día a todos, tengo una pregunta, tengo un elastix 2.2 el cual me esta funcionando bien, cuando quiero sacar un reporte de llamadas realizadas por medio de un pin set me lo muestra sin ningún problema siempre y cuando yo sea el usuario admin, el problema es cuando doy de alta otro usuario este nuevo usuario me pide que tenga una extensión ligada, y aun que le ponga privilegios de administrador no me da el reporte correctamente; ¿Alguien sabe que se puede hacer para que otro usuario saque estos reportes por medio de Pin set? en la interfase web. Fri, 14 Jun 2013 10:43:23 -0500 http://www.neomano.org/index.php/en/component/kunena/50-novatos/116475-problemas-nat-al-recibir-llamadas-por-trunk-sip.html#123284 Que tal; soy nuevo en el foro y nuevo en el manejo de elastix, tengo exactamente el mismo problema, en mi caso las llamadas que recibo son de un router cisco que ya esta configurado, utilizando el wireshark veo que las llamadas si llegan a la IP del elaxtix pero son rechazadas y muestran el error 404, el router cisco no tiene claves como un proveedor de VoIP por lo que no tengo clave y solo he asignado como host la IP del router. La pregunta seria como puedo armar mi cadena de registro si no tengo claves??, solo tengo host y el DID. Gracias por su tiempo y por la ayuda Fri, 14 Jun 2013 10:18:24 -0500 http://www.neomano.org/index.php/en/component/kunena/84-le-coin-du-debutant/123271-pas-de-detection-du-decrochage.html#123277 Salut. Rapidement avant de manger. Il faut absolument que tu valides le firewall sur Elastix! Tu tentes le diable. Fail2ban seul n'est pas suffisant. Il faut que tu créés les bonnes règles qui vont bien. Valides bien les DNS sur ton serveur. Pour le nat.... hmm je dirais que ça n'a pas grand chose à voir. Essayes toute fois de le valider par FreePBX en auto-détection. S'il trouve des trucs, mettre nat=yes. Mais bon je n'y crois pas. Mais bon en mettant Elastix sur OVH ... Ils ont déjà virer leurs solutions Elastix hébergées il y a quelques années de çà.... Vérifies si tes postes sont bien enregistré. Si tes postes SIP se connectent depuis une ip fixe, renseignes à font tout ce qui est deny/permit. Fri, 14 Jun 2013 05:12:57 -0500 http://www.neomano.org/index.php/en/component/kunena/19-call-center/123274-what-is-agent-inactivity-timeout.html#123274 In the updated call center module there is a new option in the configuration menu "Agent inactivity timeout" Does anybody konow what it is?? Thanks Fri, 14 Jun 2013 04:30:58 -0500 http://www.neomano.org/index.php/en/component/kunena/3-help/123256-ayuda.html#123256 Me acaban de instalar una central teléfonica IP en la nube, la verdad no se mucho de ésto pero con los manuales lo he ido resolviendo. Mi problema es que el aparato no me da tono, ademas si marco, puedo llamar y escucho a la otra persona pero la otra persona no me escucha a mi, en lo que me puedan ayudar, gracias Thu, 13 Jun 2013 23:17:07 -0500 http://www.neomano.org/index.php/en/component/kunena/26-tips-and-tricks/123152-elastix-240-lmsensors-for-amd--via-board.html#123254 Run sensors-detect and you will see a few more chips added Thu, 13 Jun 2013 21:10:35 -0500 http://www.neomano.org/index.php/en/component/kunena/3-help/123190-intercom-and-door-release.html#123249 Thanks mbit. I am in Australia. If these systems are insecure what is a better system? Thu, 13 Jun 2013 19:46:21 -0500 http://www.neomano.org/index.php/en/component/kunena/25-newbies-corner-/123169-analog-paging.html#123247 Yes SIP paging devices can be extremely loud. I installed one recently to be purely a loud ringer and i was asked to have it turned down and it wasnt on the loudest setting. Thu, 13 Jun 2013 19:25:10 -0500 http://www.neomano.org/index.php/en/component/kunena/3-help/123205-strange-problem-spaxxx-and-registration.html#123246 Try rebooting your switch. The ARP table maybe have an issue. This is pretty common to a few SMB Cisco switches. Thu, 13 Jun 2013 19:23:36 -0500 http://www.neomano.org/index.php/en/component/kunena/57-asterisk/115092-extension-sip-remoto-con-zoiper.html#123242 Si tienes una IP Fija en tu Router tiens que ponerla en el sip_nat.conf, tambien tienes que ver en la consola de tu central si que es lo que pasa, pueden ser varios factores, prueba la extencion remota como extension local a ver si pasa loa mismo, si lo tas usando en un Smartpohne te recomiendo 3cx es mas lijero y he tenido mejores resultados. Thu, 13 Jun 2013 16:58:07 -0500 http://www.neomano.org/index.php/en/component/kunena/50-novatos/45317-consulta-sobre-dial-plan-y-outbound-route.html?limit=10&start=20#123240 Estimado Dario si te es posible podrias pasarme la lista de las caracteristicas, estoy armando un elastix y soy nuevo en esto, gracias es mi usuario @gmail.com mi correo, gracias Thu, 13 Jun 2013 15:54:54 -0500 http://www.neomano.org/index.php/en/component/kunena/19-call-center/122829-agent-log-in-terminated.html#123237 It may be a nat or dtmf issue. Try to dial to *97 and make sure if you can enter your voicemail password and being able to login. Thu, 13 Jun 2013 14:57:34 -0500 http://www.neomano.org/index.php/en/component/kunena/53-trucos/123235-personalizacion-de-ivrs-creados-desde-elastix.html#123235 Hola a todos. hace algunos dias tuve la necesidad de personalizar un ivr creado desde la interfaz web de Elastix, mas especificamente hacer una consulta a una base de datos, con unos datos ingresados por el usuario, pero este tipo de opciones no se pueden encontrar desde la interfaz web, asi que me di a la tarea de buscar como hacerlo, lastimosamente busque y busque, pero las soluciones no eran del todo optimas, ya que la mayoria recomendaban hacerlo a mano en el archivo extensions_custom.conf, el cual no es sobreescrito por elastix al hacer una actualizacion o reinicio del servidor. Buscando y buscando por los archivos de configuracion, me di a la tarea de revisar el archivo extensions_additional.conf, el cual es en donde elastix coloca los ivr's creados desde la interfaz web, y efectivamente encontre el ivr que habia creado y necesitaba personalizar. Pero lastimosamente no podia hacer las modificaciones alli, por que cada vez que hacia un cambio en algun parametro en Elastix, mis modifi! caciones eran borradas. Asi que, que debia hacer?. La respuesta estaba ante mis ojos, justo abajo del nombre del contexto de mi ivr (Elastix coloca los ivr's creados en el archivo extensions_additional.conf con el nombre ivr-(algun numero)) habia una instruccion "include" la cual aparecia "include => ivr-(algun numero)-custom", y pense, sera que si coloco un contexto con ese mismo nombre en el archivo extensions_custom.conf tomara mis personalizaciones?. La respuesta es si. Creé en el archivo extensions_custom.conf mi personalizacion, en mi caso la consulta a la base de datos y funciono perfectamente. Solo basta con colocar en ese contexto la opcion que debe escojer, en mi caso era la 7, la cual no se programo desde la interfaz web y listo. Espero les sea de ayuda. mi version de Elastix es la 2.4.0 Thu, 13 Jun 2013 14:33:57 -0500 http://www.neomano.org/index.php/en/component/kunena/84-le-coin-du-debutant/123124-une-annonce-qui-sonnerait-.html#123234 EXCELLENT !!!!!!!!! MERCI beaucoup ! Thu, 13 Jun 2013 14:14:46 -0500 http://www.neomano.org/index.php/en/component/kunena/79-general/123210-probleme-de-son-seulement-dans-un-sens.html#123228 Salut et bienvenue sur notre forum Elastix. Si tu as un 200, c'est que tu as bien décroché. As-tu le ACK après le ACK? Si tu as un problème de son, c'est en général: - Soit un problème de port RTP. - Soit un problème de codec sur sur l'appelé. Voir si tu as le même problème le A vers B et de B vers A. Utiliser les commandes de maintenances: CLI> set rtp debug ip CLI> set sip debug peer ou ip . et regarder le SDP (codec port...etc). CLI> sip show channels durant la comm pour voir quel codec est présenté. Voir si dans les trunks tu as un progessinband=yes, no, ou never. Suivant les cas: (origine ici http://www.voip-info.org/wiki/view/Asterisk+sip+progressinband) progressinband=yes When "RING" event is requested, always send 180 Ringing (if it hasn't been sent yet) followed by 183 Session Progress and in-band audio progressinband=no Send 180 Ringing if 183 has not yet been sent establishing audio path. If audio path is established already (with 183) then send in-band ringing (this is the way asterisk historically behaved because of buggy phones like polycom) progressinband=never Whenever ringing occurs, send "180 ringing" as long as "200 OK" has not yet been sent. This is the default behaviour of Asterisk. Ça peut être ton problème du 180. Thu, 13 Jun 2013 13:35:13 -0500 http://www.neomano.org/index.php/en/component/kunena/55-call-center/119570-marcador-predictivo-mas-de-la-que-el-calcula.html#123226 Eso sucede porque lo has de haber hecho de forma incorrecta, revisa el link y mira si es que el código ha cambiado y haz los ajustes necesarios. Thu, 13 Jun 2013 12:36:18 -0500 http://www.neomano.org/index.php/en/component/kunena/55-call-center/122812-reproduccion-de-audio.html#123225 Muy probablemente estás utilizando líneas analógicas, y el tema es la detección de contestado a través de asterisk. Para solucionar el problema, deberás contratar el servicio de polaridad reversa con tu proveedor analógico y configurar chan_dahdi.conf o dahdi-channels.conf para que monitoree el cambio de polaridades. Otra alternativa es que utilices líneas digitales, es decir: troncal SIP, E1/T1 Thu, 13 Jun 2013 12:35:17 -0500 http://www.neomano.org/index.php/en/component/kunena/1-installation-issues/122701-migrate-a-configuration.html#123223 Hello, I did the migration, I manually copied the configurations, I had some problems: - New version of Elastix wants a SIP secret more complex (numbers, minus and capital letters, min. 6 digits...) so I did a small trick on the web interface (founded by googling) to let it accept the simple secret already in use. So I avoided to reconfigure all the telephones :-) - A number of personalizations in the asterisk *custom*.conf files to apply - A problem in a Macro, with the character "|" instead of "," (due to the new version of asterisk?) - A custom web application for realtime statistics to adapt to work on the new system - The openvox card not being recognized at first (had to change the slot) - some other things that I do not remember now By the way, now the new system is running good so far. Many thanks for the support! Thu, 13 Jun 2013 12:20:15 -0500 http://www.neomano.org/index.php/en/component/kunena/50-novatos/122747-registro-de-llamadas.html#123222 central digium asterisk? Este es el foro de Elastix, si es que te refieres a una central Elastix, entonces uedes revisar los detalles del cdr o de logs desde la interfaz web. Thu, 13 Jun 2013 12:18:58 -0500 http://www.neomano.org/index.php/en/component/kunena/11-ayuda/123221-problema-fax-linea-analogica-gxg4108.html#123221 Hola necesito ayuda. Tengo creado el fax virtual con la extensión iax2, internamente funciona muy bien, tengo un fax analógico conectado a un gateway y desde ahí puedo enviar sin problemas y los recibo en el correo electrónico, pero cuando un fax es enviado desde afuera por linea analogica a través de un gateway granstream GXW4108, da señal pero luego da error de linea en el fax. Alguien me puede ayudar con esto. Gracias Thu, 13 Jun 2013 12:08:29 -0500 http://www.neomano.org/index.php/en/component/kunena/31-asterisk/123183-system--dashboard--communication-activity-applet.html#123219 No, unfortunately Im not that familiar with the console yet. Thu, 13 Jun 2013 11:31:07 -0500 http://www.neomano.org/index.php/en/component/kunena/26-tips-and-tricks/123171-distinctive-ring-for-inbound-calls.html#123212 ms2oo8, Set all the phones to the ring type that you want for internal calls (thats because there is no way of setting on Elastix for internal calls (not that I have seen anyhow)....so we are relying on the default ring set on the phone for internal calls. In inbound routes Just drop Classic-4 into the Alert Info. So incoming calls will use the Classic-4 as the ring tone. Regards Bob Thu, 13 Jun 2013 08:03:48 -0500 http://www.neomano.org/index.php/en/component/kunena/1-installation-issues/110172-uelastix.html?limit=10&start=30#123209 change the file /etc/sysconfig etwork-scripts/ifcfg-eth0 and add the parameters for your own network Also its possible if not mistaken to do the changes on /etc/sysconfig etworking/devices/ifcfg-eth0 change the settings in /etc/resolv.conf and put your nameserver address (I use Google 8.8.8.8) Thu, 13 Jun 2013 04:18:16 -0500 http://www.neomano.org/index.php/en/component/kunena/19-call-center/123207-break-problem.html#123207 Hello, My Call Center agents are facing problem during End Break after come after the break. They told me, the break never ends until logout from Agent Console. Why this problem occurs? Is anybody tell me? Best Regards Kamrul Shaker Engineer, Banglalion Thu, 13 Jun 2013 03:22:05 -0500 http://www.neomano.org/index.php/en/component/kunena/19-call-center/123206-call-center-module-update.html#123206 Hello, I am an Elastix Server Administrator of Banglalion Communications Ltd. We are using Elastix Server from last 4 months. Its really easy to use. But I have found some problems is this Elastix Box. I will talk about those problems in other thread. Here, i want to know about the Elastix Call Center module new update (Call Center 2.2.0-1). Is this update will change much in Elastix? How much helpful will be the update? Will this solve some existing problem like Break Problem and Queue Distribution problem? Can anyone tell me the answer of above question? B) Best Regards :) Kamrul Shaker Engineer, Banglalion Thu, 13 Jun 2013 03:19:29 -0500 http://www.neomano.org/index.php/en/component/kunena/25-newbies-corner-/123174-restrict-sip-registrations-by-useragent.html#123199 mbit wrote: Have you tried doing this in the elastix interface under pbx, then extension? Which field exactly under extension ? I do not find any suitable field under extensions. "Allow" "Disallow" fields are for specifying codecs and "deny" "permit" are for network ACL... Apart from these fields I don't see anything relevant. Let me know if I am missing something here :) Thu, 13 Jun 2013 02:25:27 -0500 http://www.neomano.org/index.php/en/component/kunena/116-security/122238-fail2ban-elastix240.html#123195 Also elastix comes with a web configuration tool for iptables. Try and lock down people accessing your machine to certain IPs. This will also stop hackers. Thu, 13 Jun 2013 02:06:35 -0500 http://www.neomano.org/index.php/en/component/kunena/18-modules/123189-greek-characters-in-campaign-not-recognised.html#123189 Hi. I am using elastix call center module in order to make outbound campaigns. When I upload csv files with the numbers and contact details in latin these are shown with no problem when the call is connected to the agent screen. When I upload a csv with the name of the customer in greek nothing is shown in the agent's screen. Is there a solution for these problem??? Thanks in advance. Thu, 13 Jun 2013 00:30:35 -0500 http://www.neomano.org/index.php/en/component/kunena/3-help/123173-add-new-sip-header.html#123187 Hi. You mean this: http://www.voip-info.org/wiki/view/Asterisk+cmd+SipAddHeader Wed, 12 Jun 2013 22:47:28 -0500 http://www.neomano.org/index.php/en/component/kunena/11-ayuda/84321-elastix-en-joomla-es-posible.html#123186 bueno aparte de la implementacion de joomla deberas aplicar click2dial. yo estoy montando ahora archivos en php al apache de elastix..funciona bien..ya con joomla es muy inseguro si esta en internet..pero si es una intranet..no le veo ningun problema Wed, 12 Jun 2013 22:46:24 -0500 http://www.neomano.org/index.php/en/component/kunena/31-asterisk/122145-not-able-to-hear-the-voice-one-way-voice.html#123185 Site 1: Network: 192.168.1.0 Default Gateway 192.168.1.1 Extn: 482 - Raj -192.168.1.101 ------------------------ Elastix Easy : 192.168.1.200 - IP PBX Server --------------------------- Site 2: Network 192.168.2.0 Default Gateway 192.168.2.1 Extn : 216 - Jasmine - 192.168.2.216 Please check and tell me how i can fix this issue. Thank you Wed, 12 Jun 2013 22:26:41 -0500 http://www.neomano.org/index.php/en/component/kunena/3-help/123181-play-radio-music-when-sip-phone-is-idle.html#123184 Add the following somewhere within the [from-internal-custom] context: exten => 466,1,Answer exten => 466,2,Playback(pls-hold-while-try) exten => 466,3,SetMusicOnHold,stream name exten => 466,4,WaitMusicOnHold,300 exten => 466,5,Hangup You can change the ext number to what ever you like, and stream name is the name of your stream Wed, 12 Jun 2013 17:05:19 -0500 http://www.neomano.org/index.php/en/component/kunena/11-ayuda/103997-configurar-sip-remoto.html?limit=10&start=10#123182 Buenas hace ya mucho que nadie responde a este post, tengo las miasmas dudas que el compañero, podrían ayudarnos y seguir con esto, gracias de antemano. Si ya lo solucionaste amigo pon aquí como le hiciste. Wed, 12 Jun 2013 16:14:45 -0500 http://www.neomano.org/index.php/en/component/kunena/84-le-coin-du-debutant/123133-sccp-avec-elastix.html#123180 Bonsoir franck, oui pas mal ton module, j'aime bien, sauf que le jeeves intègre déja ce type d'apply, check in, check out etc...... je teste demain l'interconnexion. :) Patrick Wed, 12 Jun 2013 14:43:37 -0500 http://www.neomano.org/index.php/en/component/kunena/19-call-center/123177-outgoing-campaign-not-work-after-some-time-.html#123177 Hi! I have installed elastix call center module for 5 agents, for testing i am using 1 agent. i have uploaded leads in csv format. and everything once run successfully. issue : 1)sometimes is works and sometimes it don't, so i need to restart and rebuild campaign to make it work 2)Numbers are not dialed in sequence and when i check report some are not even dailed can you pin point the issue? Regards, VoipBuddy Wed, 12 Jun 2013 13:52:34 -0500 http://www.neomano.org/index.php/en/component/kunena/39-others/123176-rutas-entrantes-con-otro-server.html#123176 Buenas tardes Este foro me ha servido mucho en anteriores inconvenientes y/o dudas... Ahora me encuentro con un escenario nuevo al menos para mí, paso a explicar, a ver si alguno me pueda dar una mano. Tengo dos elastix 2.2 conectados mediante troncal IAX. Uno de los servers (server51) está solamente para la conexión con un astribank, donde van las lineas analógicas, no tiene extensiones configuradas. El otro server (server54), es donde están todas las funcionalidades, extensiones, grabación, monitoreo, reportes, etc, etc, etc. También tengo unas troncales IAX y SIP para conexiones con otros equipos (cellular gateway, elastix remotos) El problema es que al configurar rutas entrantes en el server51, debo direccionar todas las llamadas entrantes a un grupo de extensiones, ya configurado en server54.. y no puedo dar con una solución efectiva... Una idea es en el server51 crear la ruta entrante, y direccionar a la troncal con server54. Y aquí me confundo : si es que debo crear una ruta entrante que tome solo las llamadas de esa troncal con server51, cómo configuro esto? ya que server54 tiene otra ruta entrante... Espero haber expuesto mi problema de manera clara... y si alguien puede darme una mano, le agradecería un montón. Gracias a todos desde ya. Wed, 12 Jun 2013 13:33:34 -0500