search for: voicemailbox

Does Asterisk know when it hits a voicemailbox ? When calling to a cell-phone or GSM, after some rings and no pickup you arrive at a voicemailbox. If Asterisk does not know it's a voicemailbox that has answered the call, the voicemailbox will contain 60minutes of 'silence'. This is very expensive 'silence'. How to avoid t...

Hi: when add a new voicemailbox, asterisk will create a new directory to it. since linux has limitation for the number of subdirectory. i wonder how many voicemailbox can asterisk support? thanks. yan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/...

The demo 1235 extension that Asterisk ships with works fine and the messages are sent to the address I set in voicemail.conf. I guess that means that my configuration is working perfectly so far. But when I set up another extension with a voicemailbox, no mail is sent when a message is left, although I can dial voicemail and listen to the message just fine which I guess rules out voicemailbox misconfiguration. The strange thing is that both extensions and mailboxes are configured exactly the same : in extensions.conf : exten => 1235,1,Voice...

Hi there, see subject. I'd like to be able to use the vmbox prompt of VoiceMailMain2 and use 1234 and 4321 to point to the same mailbox. Will it be sufficient to create a soft link for 4321 --> 1234 in /var/spool/asterisk/default or will I get myself into horrible trouble? Background: I like to be able to map certain functions ("boss", "peasant",

Anyone else has problems with CVS HEAD's from today with voicemail hanging up silently without any debug/error messages when checked? It also keeps insisting that the user's voice mailbox is full and can't store more messages even if I clear/rebuild the /var/spool/asterisk/voicemail stuff. I've tried falling back to voicemail.conf entries from realtime voicemail with the same

I just completed building a feature rich asterisk voicemail system using perl, php, and mysql. My only concern is that the system i built will not be able to handle the call volume needed. Let me start by explaining my setup. Incoming call -> route.agi (perl -> mysql lookup) -> AGI -> voicemailbox (using mysql odbc) or terminate with wrong number message if a message is left in a voicemailbox the following happens: externnotify -> notify.pl (perl -> mysql lookup) -> up to 2 calls originated (using AMI), up to 4 emails sent out (with up to 2 attachemnts of voicemail) this system ma...

Hi, the app HasNewVoiceMail can't find my voicemail. This is the errormessage: Dec 3 14:24:01 NOTICE[12222481]: app_hasnewvoicemail.c:104 hasvoicemail_exec: Voice mailbox 25 at /var/spool/asterisk/voicemail/default/25/(null) does not exist however this is the output of lspbx:~# ls -l /var/spool/asterisk/voicemail/default/25/ total 316 -rwx------ 1 root root 11814 2003-11-22 18:18

Hi List! I use Asterisk as a pure voicemailbox at a customers place. Right now, a telephone uses up two msns, one for the telephone itself, and one for the telephones mailbox. If the user is absent, a telephonecall is redirected to the voicemail msn of that users telephone. The Problem is: The PBX supports a too small number of msns, so I can...

...sip peer has its own DID number, which is established via cid_number = 6021231234. When a call is placed from SIP peer #100 to SIP peer #101, and SIP peer #101 wants to reply to #100's voicemail via the option #3 advanced options menu, * responds with an error that #100 does not have a voicemailbox. Of course * thinks the voicemailbox = the caller ID/DID of #100, which in this case would be 6021231234 rather than 100, which is the defined voicemailbox for 100. My question is, do I need to rewrite callerid for internal extension-to-extension calls or is there something simple I'm m...

...;Get DND key* *exten => s,n,Dial(${ARG1},${ARG2},${MOHCLASS}tw)* *exten => s,n,Goto(s-${DIALSTATUS},1)* *exten => s-ANSWER,1,Hangup* *exten => s-NOANSWER,1,GotoIf(${EXISTS(${DB(CFNR/${EXTENSION})})}?${ARG5},${DB(CFNR/${EXTENSION})},1) ;Get CFNR key* *exten => s-NOANSWER,n,Macro(voicemailbox,${ARG3},${ARG4})* *exten => s-BUSY,1,GotoIf(${EXISTS(${DB(CFB/${EXTENSION})})}?${ARG5},${DB(CFB/${EXTENSION})},1) ;Get CFB key* *exten => s-BUSY,n,Macro(voicemailbox,${ARG3},${ARG4})* *exten => _s-.,1,Macro(voicemailbox,${ARG3},${ARG4})* Thanks for your help -------------- next part --...

I'm wondering if there is a simply way to perform a voicemail callback feature using AGI. For instance, a caller leaves a voicemail, the voicemail will then call the owner of the voicemailbox determined by a database look up. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110405/2ab7c326/attachment.htm>

Hi all Looking for a solution to create a flexible voicemail solution in Asterisk without the need to preregister the voicemail users (via databases etc etc). Scenario: All incoming calls are voicemail calls however the dialled number (called party) does not necessarily have a voicemailbox configured in the Asterisk system. I am looking for * to do the following: * Call comes in * Called number is analysed (e.g. 44123456) * Call is routed to default voicemail messages * Message is recorded (for the 44123456 number) ~~~~~~ * When number (44123456) dials voicemail...

Hi all, When somebody leaves a message in the voicemailbox, is there a way to know the file name of it? I need to return the voicemail file name in the deadagi command. Thanks, Anahi _________________________________________________________________ Convierte las fotos que m?s te gustan en tu nuevo fondo de escritorio para el ordenador. Es f?...

...aa84,1 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing Goto("SIP/9083XXX-0816b208", "s-CONGESTION|1") in new stack -- Goto (macro-incoming-call-to-extension,s-CONGESTION,1) -- Executing Macro("SIP/9083XXX-0816b208", "voicemailbox||voiceone/custom/hello") in new stack -- Executing NoOp("SIP/9083XXX-0816b208", "Answering machine mode: Answer Only or Answer and Record?") in new stack -- Executing GotoIf("SIP/9083XXX-0816b208", "1?answeronly") in new stack -- Goto (macro-...

...00 ; Username on device secret=1234 ; Password for device host=dynamic ; This host is not on the same IP addr every time context=from-sip ; Inbound calls from this host go here mailbox=100 ; Activate the message waiting light if this ; voicemailbox has messages in it [2001] ; Duplicate of 2000, except with different auth data type=friend username=2001 secret=1234 host=dynamic context=from-sip mailbox=101 ==>Extensions.conf [general] static=yes writeprotect=yes [bogon-calls] exten => _.,1,Congestion [from...

...sername on device secret=cisco ; Password for device ;host=192.168.22.1 ; This host is not on the same IP addr every time host=dynamic context=from-sip ; Inbound calls from this host go here mailbox=100 ; Activate the message waiting light if this ; voicemailbox has messages in it [2001] ; Duplicate of 2000, except with different auth data type=friend username=2001 secret=cisco host=dynamic ;host=192.168.22.2 context=from-sip mailbox=101 [2002] ; Duplicate of 2000, except with different auth data type=friend username=2002 se...

...his device takes and makes calls username=2000 ; Username on device host=dynamic ; This host is not on the same IP addr every time context=from-sip ; Inbound calls from this host go here mailbox=100 ; Activate the message waiting light if this ; voicemailbox has messages in it nat=yes [2002] ; CISCO hard phone type=friend username=2002 secret=2002 host=dynamic context=from-sip mailbox=103

...the other ones to your old PBX (Some sort of proxy scenario, isn't it?). Pro: - You don't have to change a single configuration option in your old PBX if you don't want to or are not able to ;-) - Full integration of asterisk users with old PBX users (internal calls, voicemailbox on asterisk even for ole PBX users etc.) - Very flexible configuration. For example let asterisk use 3-digit extensions while switching prior 2-digit extensions to the old PBX Con: - Costs depending on your setup (Quad-Bri Junghanns costs about 600 Euro, Octo-Bri about 1000 don...

hi ppl, i'm using asterisk 1.2 because i'm making use of voiceone, but before i was using asterisk 1.4 and had the same problem, it concerns an italian voip/sip provider called eutelia/skypho, my problem is the following one: when i start my pbx my skypho account is working fine, meaning that e.g. incoming calls are shown in the asterisk CLI and caller and callee can hear each other when

...at just keeps ringing (when I call it) and does not go to voicemail. If the person there is on the phone, and someone calls it they get the busy message, but they never seem to get the 'unavailable' message... instead it will just ring and ring and ring... any ideas? They are setup with a voicemailbox, and it is set to transfer after 15 seconds of ringing.