search for: nobodyavail

Displaying 20 results from an estimated 31 matches for "nobodyavail".

2008 Dec 26
3
Problem: no such extension 'xx' in context 'default'
...nternal extensions.conf [default] exten =>2,1,Playback(digits/2) ; exten =>2,2,Goto(default,10,1) exten=>3,1,Playback(pbx-invalid) exten=3,2,Goto(default,4,1) exten=4,1,Playback(vm-goodbye) exten=>4,2,Hangup() [internal] exten => 10,1,Dial(SIP/10,10) exten =>10,2,Background(vm-nobodyavail) exten => 11,1,Dial(SIP/11,5) exten =>11,2,Background(vm-nobodyavail) now when I dial 10, I got the following error : no such extension '10' in context 'default' thanks in advance manfred _________________________________________________________________ Hol dir 30 kostenlo...
2007 Aug 08
1
asterisk wait for traling digits
...ons but when i press 1627 then it is wait for 5 second and then rining start alternative press '#' what is the method to break this space of waiting after dialing my extention.conf ;North Delhi NOC Extention exten => _16XX,1,Dial(SIP/mediant/${EXTEN},60) exten => _16XX,2,Playback(vm-nobodyavail) exten => _16XX,102,Playback(all-allbusy) ;Mumbai NOC extention exten => _22XX,1,Dial(SIP/mediant/${EXTEN},60) exten => _22XX,2,Playback(vm-nobodyavail) exten => _22XX,102,Playback(all-allbusy) exten => _17XX,1,Dial(SIP/mediant/${EXTEN},60) exten => _17XX,2,Playback(vm-nobodyava...
2015 Nov 04
4
Find me macro - calling multiple people to get a hold of one
...nesys) ; if you're too late with pressing 1 exten => t,n,Set(MACRO_RESULT=CONTINUE) [findme] exten => s,1,Set(CALLERID(all)="Alarm" <911>) same => n,Playback(please-wait-connect-oncall-eng) same => n,Dial(LOCAL/9999${WIEBE_MOBILE}) same => n,Playback(vm-nobodyavail) exten => t,1,Playback(vm-nobodyavail) ============================= First of all, what is MACRO_RESULT? I can't seem to find anything about that. Googling for it yields basically nothing. But the biggest problem is when the callee answers, then hangs up. The person calling is connected...
2007 Oct 25
2
Grandstream GXV-3000
...e = rfc2833 host = dynamic qualify = yes allow = h263 video=yes videosupport=yes ; 112 is the X-Lite phone [112] type=friend host=dynamic user=112 username=112 secret=secret allow=all nat=no ------------- extensions.conf ------------- exten => 112,1,Dial(SIP/112) exten => 112,2,Playback(vm-nobodyavail) exten => 112,102,Playback(tt-allbusy) exten => 112,103,Voicemail(b112 at default) exten => 113,1,Dail(SIP/113) exten => 113,2,Playback(vm-nobodyavail) exten => 113,102,Playback(tt-allbusy) exten => 113,103,Voicemail(b113 at default) _____________________________________________...
2008 Mar 19
3
How to configure Voice mail for multi users.
...other. Now i want to configure separate voice mail box for separate user. my extensions.conf ..... settings below.. [voicemail] exten => _X.,1,Dial(SIP/${EXTEN}) exten => _X.,n,NoOp(Dial Status: ${DIALSTATUS}) exten => _X.,n,Goto(s-${DIALSTATUS},1) exten => s-NOANSWER,1,Background(vm-nobodyavail) exten => s-NOANSWER,n,VoiceMail(${EXTEN}@usersmail) exten => s-NOANSWER,n,Hangup() exten => s-CONGESTION,1,Background(vm-nobodyavail) exten => s-CONGESTION,n,VoiceMail(${EXTEN}@usersmail) exten => s-CONGESTION,n,Hangup() exten => s-CANCEL,1,Background(vm-nobodyavail) exten =&gt...
2008 Mar 19
0
How configure Voice mail for multi users.
...Now i want to configure separate voice mail box for separate user. my extensions.conf ..... settings below.. [voicemail] exten => _X.,1,Dial(SIP/${EXTEN}) exten => _X.,n,NoOp(Dial Status: ${DIALSTATUS}) exten => _X.,n,Goto(s-${DIALSTATUS},1) exten => s-NOANSWER,1,Background(vm-nobodyavail) exten => s-NOANSWER,n,VoiceMail(${EXTEN}@usersmail) exten => s-NOANSWER,n,Hangup() exten => s-CONGESTION,1,Background(vm-nobodyavail) exten => s-CONGESTION,n,VoiceMail(${EXTEN}@usersmail) exten => s-CONGESTION,n,Hangup() exten => s-CANCEL,1,Background(vm-nobodyavail) ext...
2006 Nov 12
1
outgoing works, incoming fails on asterisk passthrough to inter-tel
...pdated conferencing on 49, with 0 conference users Nov 12 12:48:54 VERBOSE[32609] logger.c: -- Hungup 'Zap/49-1' Nov 12 12:48:54 DEBUG[32609] app_dial.c: Exiting with DIALSTATUS=NOANSWER. Nov 12 12:48:54 VERBOSE[32609] logger.c: -- Executing Playback("Zap/73-1", "vm-nobodyavail") in new stack Nov 12 12:48:54 DEBUG[32609] chan_zap.c: Took Zap/73-1 off hook Nov 12 12:48:54 DEBUG[32558] channel.c: Avoiding initial deadlock for 'Zap/73-1' Nov 12 12:48:54 DEBUG[32609] channel.c: Scheduling timer at 160 sample intervals Nov 12 12:48:54 VERBOSE[32609] logger.c:...
2004 Jan 25
3
OH323 doesnt hear ringing
...nging. However, when the H323 client rings a SIP client, there is no ringing sound at all, although as soon as the called party picks up the phone, everything works fine. This is the entry from my extensions.conf: exten => _7[5-9]X,1,Dial(SIP/${EXTEN},20,rt) exten => _7[5-9]X,2,Playback(vm-nobodyavail) exten => _7[5-9]X,3,Hangup I assume that because I havr the 'r' in the dial string, the calling party should hear a ringing noice. Any ideas? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20...
2007 Oct 30
1
G.729 transcoder beetween asterisk to avaya
...Oct 30 21:54:26] WARNING[1318]: app_dial.c:1106 dial_exec_full: Unable to create channel of type 'Zap' (cause 58 - Bearer capability not available) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [1648 at from-avaya:2] Playback("SIP/5450-a079d200", "vm-nobodyavail") in new stack [Oct 30 21:54:26] WARNING[1318]: channel.c:2991 set_format: Unable to find a codec translation path from g729 to gsm [Oct 30 21:54:26] WARNING[1318]: file.c:813 ast_streamfile: Unable to open vm-nobodyavail (format 0x100 (g729)): No such file or directory [Oct 30 21:54:26] WARNI...
2006 Oct 31
1
Asterisk does not bridge zap channels on outgoing calls
...isk/monitor/${UNIQUEID},mb) exten => 26,3,Dial(Zap/6,20,tT) exten => 26,4,Hangup() exten => 27,1,SetCDRUserField(${UNIQUEID}) exten => 27,2,Monitor(gsm,/var/spool/asterisk/monitor/${UNIQUEID},mb) exten => 27,3,Dial(Zap/7,20,tT) exten => 27,4,Hangup() ;exten => t,1,Playback(vm-nobodyavail) ;exten => t,2,Hangup() ;-------------------------------- [int_omg] ;-------------------------------- include => parkedcalls include => agentie include => mobline include => int_soft include => int_agentie include => mediasat_sip ;exten => XX,1,SetCDRUserField(${UNIQUEID}...
2007 Oct 14
4
ResponseTimeOut() and t extension
...KuwaitInternal include => EgyptInternal exten => 1000,1,Goto(s,1) exten => s,1,Answer() exten => s,2,ResponseTimeout(5) exten => s,3,Background(WelcomeMessage) exten => 0,1,Dial(SIP/EgyptOperatorSIP,10) exten => 0,2,Background(WelcomeMessage) exten => 0,2,Playback(vm-nobodyavail) exten => 0,3,Hangup() exten => 0,102,Playback(tt-allbusy) exten => 0,103,Hangup() exten => i,1,Playback(pbx-invalid) exten => i,2,Goto(EgyptIncomingPSTN,s,1) exten => t,1,Playback(vm-goodbye) exten => t,2,Hangup() Any help?? Regards Bilal _____________________...
2010 Feb 17
2
asterisk dahdi fax problem
...fax. The CLI says it picks up the line but no dialing. I tried the extension with an analog phone, it works fine, I can dial out (the required fax answers), I can dial extensions. The settings in the extensions.conf are: FAX=DAHDI/21 exten => 51,1,Dial(${FAX}) exten => 51,n,Playback(vm-nobodyavail) exten => 51,n,Hangup() What do I miss? Thanks for you help in advance. Best regards, Peter Gelencser
2010 Nov 06
1
Abandoned queue calls do not produce a CDR?
...(the default) in cdr.conf. The call shows, as expected, in the queue_log as ABANDON The dialplan is: Ringing(); Answer(); // need to answer or no music! goto s,no-ivr; Queue(hotelq,t,,,${QUEUE_TIMEOUT}); Background(vm-nobodyavail); Voicemail(${HOTELVM}); Playback(goodbye); Hangup(); Note that if the queue times out, I am calling Voicemail and if the caller hangs up a CDR is produced. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists....
2013 Sep 22
1
Play subscriber's recorded messages
Hello,For the time being I am using the following line to play the original saved message by Asteriskexten => 7001,n,Playback(vm-nobodyavail)Now I am trying to use the other features for Asterisk's voicemail. I have recorded a message, and I can see it saved on the system, but still Asterisk keeps playing the original message... Is there something I can add to let the subscriber plays his recorded message if he has a one? I couldn...
2011 Dec 03
2
google voice calling dial plan question.
When a caller calls my google voice phone number, I must answer, wait and press one to accept. Sometimes even that does not work. I have tried a few different things to get asterisk to place the call in an answered state and send the DTMF 1 with the Dial macro. I found Malcom Davenports wiki page regarding Google calling which has been very helpful in troubleshooting the issue.
2004 Mar 31
2
Basic Answering Machine Function?
I've had my * setup installed with an X100P card for a couple of weeks and it's great fun! I'm even giving a demo to the local Linux group in a couple of days. But I have a snag. I have the X100P on a shared line, and configured to wait for 20 seconds before answering and doing the auto-attendant/voicemail dance. My problem is I can't find an application command to cancel the
2005 Aug 05
0
Another problem on queues
...ty 26 -- AGI Script Executing Application: (NoOp) Options: () -- AGI Script dialparties.agi completed, returning 0 -- Executing Wait(" Local/8521@from-internal-8268,2", "1") in new stack -- Executing Playback(" Local/8521@from-internal-8268,2", "vm-nobodyavail") in new stack -- Local/8521@from-internal-8268,1 answered SIP/XXX.XXX.XXX.XXX-43921110 -- Stopped music on hold on SIP/XXX.XXX.XXX.XXX-43921110 -- Playing 'vm-nobodyavail' (language 'en') -- Executing Playback(" Local/8521@from-internal-8268,2", &quot...
2006 Feb 23
0
problems while dailing outside
..._ks channel => 4 _*zaptel.conf:*_ fxoks=1 fxsks=4 loadzone=il defaultzone=il _*extensions.conf:*_ [globals] DAVID=Zap/1 OUTBOUNDTRUNK=Zap/4 [incoming] exten => s,1,Answer() exten => s,2,Background(vm-enter-num-to-call) exten => 123,1,Dial(${DAVID},10) exten => 123,2,Playback(vm-nobodyavail) exten => 123,3,Hangup() exten => 123,102,Playback(tt-allbusy) exten => 123,103,Hangup() ;exten => 1,1,Playback(digits/1) # repeat vocaly the digit that pressed exten => i,1,Playback(pbx-invalid) exten => i,2,Goto(incoming,s,1) exten => t,1,Playback(vm-goodbye) exten => t,2,...
2006 Feb 27
0
voipstunt can't get call in asterisk
...vite=yes context=13 --------------------------------------------------------------------------------------- my extensions.conf [general] static=yes writeprotect=no [13] include=default include=outgoinguseruser exten =>13,1,Dial(SIP/13,17,r) exten =>13,2,Answer exten =>13,3,Playback(vm-nobodyavail) exten =>13,4,Voicemail(13) exten =>13,5,Hangup [outgoinguseruser] exten => _XXXX.,1,Dial(sip/${EXTEN}@useruser,60) exten => _XXXX.,2,Congestion exten => _XXXX.,102,Busy [incomingsip.voipstunt.com] exten =>user,1,SetCIDName(${CALLERIDNAME}) exten =>user,2,Dial(Local/13@13)...
2007 May 22
0
Dialplan Problem - Outgoing
...layback(pls-try-again-later) exten => s-CHANUNAVAIL,3,Hangup exten => s-BUSY,1,Playtones(busy) exten => s-BUSY,2,Hangup exten => s-CONGESTION,1,Playtones(congestion) exten => s-CONGESTION,2,Hangup exten => s-NOANSWER,1,Playback(vm-nobodyavail) exten => s-NOANSWER,2,Hangup exten => t,1,Goto(#,1) ; If they take too long, give up exten => i,1,Playback(invalid) ; "That's not valid, try again" exten => i,2,Hangup ................ .........