search for: voicemailboxes

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 this ? Jonas -------------- next part --------------

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/attachments/20030430/bd36cdaa/attachment.htm

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

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

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

...he PBX supports a too small number of msns, so I can't give every user a voicemailbox. Mailboxes are assigned after different contexts (in capi.conf the "msn" option). It would be extremely cool, to create new contexts after "incomingmsn", I would only use up 1 msn for all voicemailboxes, and call the context according to the telephone of the user, that was called in his/her absence. Now, that does not work though. The incomingmsn apparently doesn't create a new context. Or does it? Is there a way to do that? Here a diagram of what I want to do: msn 12: the telephone of the...

Hello, I'm having an issue with caller ID in voicemail that I'd appreciate any input on. I have two sip peers defined as extension 100 and 101 each with separate voicemail accounts. Each 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

Hello my friends, I've a doubt, i want to be able to forward the incoming calls from PSTN to my cell phone...i mean, qhen i'm out of the office i need like aq macro that helps me to forward the incoming call that goes for example to my internal extension SIP 207, i 've this macro but i can make it work properly....i can't activate the forward in the phone, is quite confuse:

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:

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 *

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?cil y adem?s gratis

hi ppl :D my configuration is as follows, i run (let's call it machine 2) debian etch 4.0 and asterisk 1.2, i use voiceone (www.voiceone.it) as an interface to manage asterisk, I have a grandstream/handytone 486 as a sip device, no PSTN line or anything like that all SIP only. I have a machine (machine 1), which functions as my router and machine 2 and sip device are behind it, grandstream box

Hi all, Any one tried installing Asterisk on Linksys WRT54G? We have but facing problems with SIP to SIP calls. The phones ring and calls are established but we cannot hear any voice at all. I tried allow=all in the general section but did not work. So I forced ulaw. Can any one please check it out and let me know what is wrong? Here are the conf files: Asterisk Version: Asterisk

I feel I'm on the verge of setting up a pbx for handling internal calls only... The last problem - I think - I've run into is w/ the phone registration running asterisk -vvvc I get a bunch of messages looking like so Apr 6 14:46:05 NOTICE[1116957488]: chan_sip.c:5623 handle_request: Registration from 'sip:2001@192.168.22.254' failed for '192.168.22.1' Apr 6

I am trying to make 1 soft SIP UA behind NAT connect to a public hard CISCO UA via a public asterisk server. The CISCO UA can hear the voice from the SIP UA but not vice versa. I do set nat to yes for the soft phone. Any help would be greatly appreciated. Below is my sip.conf [general] port = 8060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0 ; Address to bind to (all

> Hi all, Hi Marco, > I was thinking about integrating an old PBX with Asterisk and I was wondering > some possible configurations. You didn't mention the number of lines your PBX uses, but think of a third scenario: Install an asterisk with twice the number of BRI/PRI-Ports your current PBX has. Connect half of them to your carrier, the other ones to your old PBX (Some sort of

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

Hi, I have one phone on my network that 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