search for: voicebox

Hello! I have been googling a lot and asked wiki a few times now, but i cant find a howto for setting up a voicebox. Any link/hint would be great! Thanks, Mario

Hi! I have an automated dialout via a call file to a mobile. Can I detect when the call is not answered but forwarded to the mobile operator voicebox? I would like to stop the dialout if this is the case. TIA, Mike

...Especialy as the number read is the 'internal' number. Callers often hang up because they think having called the wrong number when they hear the announcement. Is there a way (like in many other PBXes) that the VoiceMail user could record his own announcement? (like, hello, this is the Voicebox of John Smith, please leave a message after the tone). Mit freundlichen Gr?ssen Benoit Panizzon -- I m p r o W a r e A G - System Services ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +...

hi community, I'm new to this list & asterisk in general, so let me first say thx to everybody involved in providing such great tools & ressources!! I'm currently trying to implement a simple voicebox-system. for demonstration purposes, I've successfully connected my cellphone via bluetooth using the current chan_cellphone-patch on the current SVN-version of asterisk. everything seems to work fine so far (great patch!) what I want to achieve: * incoming call arrives * asterisk/cellphone an...

Hello! I would like to call a number (e.g.35), and when i press a secret code (12345), it should jump to my voicebox menu. On this page http://www.voip-info.org/wiki-Asterisk+cmd+background i found something about "Silently Wait for DTMF Input". In my case it wouldn`t be silence. It woudl just play the away message. Now how can i include such a secret code to my background funktion? I am looking for so...

Making custom "voicemail greetings" seems fairly straight forward, and I've done it. However, I'm looking for a way to make the actual extension answer with "You've reached my Jim Dandy voice mailbox, go take a flying . . .". (OK, so maybe not), instead of "The person at extension xxxx, is unavailable" Possible? Easy? Under my nose? joe a.

...an looks like this: exten => s,1,Answer exten => s,n(restart),BackGround(intro) exten => s,n,Read(Enter,4,4) exten => s,n,Voicemail(${Enter},u) exten => s,n,agi(process.php|${Enter}) exten => #,1,Playback(thanks) exten => #,n,Hangup it lets a user record a message to a chosen voicebox, when the user finishes his message my pressing "#", the script process.php is executed via AGI everything fine. however, when the user finishes recording by simply hanging up, asterisk isn't executing the AGI-command any more. what I'm looking for is some kind of "on hangup&...

...ne CAPI-capable board (AVM Fritz!DSL) connected to a BRI (T-ISDN), one HFC-S board running in NT-mode connected to an internal S0 bus with some ISDN devices (DECT stations, TA) and, of course, some ethernet interfaces. ISDN standard used is Euro-ISDN. Trying local extension "9999" (check Voicebox) and them DTMF- dialing the number of the Voicebox to access does not work at all. here's the output of "pri debug span 1": == Primary D-Channel on span 1 up for TEI 64 < Protocol Discriminator: Q.931 (8) len=30 < Call Ref: len= 1 (reference 1/0x1) (Originator) < Messag...

...s. <p>One thing I've found in using the preprocessor that I need to find a solution for is that sometimes the denoiser is incorrectly detecting speech as noise. This happens when a speaker speaks for a while, and the denoiser tries to remove their intonation (i.e. the sound of their voicebox; the vowels). The result is that towards the ends of sentences, their voice gets "thinned" out. I'd like to find out how to "slow down" the rate of adaptation within the denoiser to make this less likely to happen. I assume that the tradeoff is that it will adapt more...

Today I updated my asterisk cvs, I would love to set up mysql support for voiceboxes. Here is what has happened (by now): cvs does not compile as downloaded, module chan_oss reports an error; if I compile with -i all works fine except for chan_oss. Ok, time to set USE_MYSQL_VM_INTERFACE=1 in apps\Makefile and to do make install -i When I launch asterisk, (with chan_oss excl...

Hi, ztdummy says the following: VoiceBOX:/usr/src/zaptel# modprobe ztdummy /lib/modules/2.4.18/misc/ztdummy.o: unresolved symbol zt_unregister /lib/modules/2.4.18/misc/ztdummy.o: unresolved symbol zt_transmit /lib/modules/2.4.18/misc/ztdummy.o: unresolved symbol zt_receive /lib/modules/2.4.18/misc/ztdummy.o: unresolved symbol zt_register...

...t so I think it might be interesting for someone else too. Short overview: - seperation of the numbers in blocks (just definded) - error/cause handling - advanced access control (even vor ISDN Zap clients) - access levels - denylist with national number blocks (for Germany) - emergency handling - voicebox handling Environment: - located in Germany - Trunking via ISDN-CAPI - extensions via ISDN Zap (HFC-S), PTMP mode It's a little difficult to desribe it in detail but the files are fairly good documentated so it should be possible to understand it ;-) It has been designed to work in Germany bu...

...ine!! firefly2firefly (stun enabled) also works perfectly!(??)!...(testclients and gateway are in different subnets). the other thing is rather a config issue I guess :) when receiving incoming calls from the pstn direct access isn't working (dialing asteriskpstnnoplusextension) -> asterisk voicebox always answers standard config example could be pretty useul I guess ;P thanx in advance seb -- GMX ProMail mit bestem Virenschutz http://www.gmx.net/de/go/mail +++ Empfehlung der Redaktion +++ Internet Professionell 10/04 +++

...but nobody was physically there. What to do? Being logged in on a shell on my remote asterisk machine I used the following script to trigger outgoing calls from an office snom 190 phone to my phone beside me on the desk. A timeout of 3 secs for POTS or 15 secs for my mobile guaranteed that no voicebox would take over but I heard a short ring when calls got through, to add a real life ringtone to remote visual feedback from asterisk -rvvvvv. httpsnom-dialtest ------------------------- #!/bin/bash # Created 070107 by AvH # $1 is the extension to dial if [ "$1" = "" ] the...

Is there a method from the dialplan that I can turn on a voicemail indicator on a polycom phone. Like a blinking light or something. Then I would also need to turn it off. Is there a way to do that? Jerry

Dear, I want to implemanted syslinux in my HDD, is there special case? Pls...help me... -- Best regards, Jack mailto:blue_eye at BonBon.net YM : kanda_sayang ICQ : 111377615

...XXXXX origdate=Thu Nov 1 12:43:38 PM UTC 2007 origtime=1193921018 category= duration=17 ----- s n i p ----- Alexander is the PRI Asterisk that IAX2 forwards all calls to Graham (where the log above is from). And the call to '*500' was made from 597. I have a *xxx extension because the 500 voicebox is a switchboard voicemail and can/should be checked by the whole switchboard (using Queue()'s). File modes: ----- s n i p ----- graham:/usr/share/asterisk/sounds/se# ll /var/spool/asterisk/voicemail/default/500/INBOX/msg0000* -rwxr-xr-x 1 asterisk asterisk 28660 Nov 1 13:43 /var/spool/aste...

Hi, I am currently setting up a voice mail/IVR machine with asterisk (1.4.10 at the moment). During testing and evaluation, all was fine; in the - slightly different - production environment, the IVR contexts do not react sensibly. The environment is: POTS <-- (ISDN) --> PBX <-- (SIP) --> Asterisk with the Asterisk registering with our local PBX. When a user reaches the Asterisk

...thing I've found in using the preprocessor that I need to find a >solution for is that sometimes the denoiser is incorrectly detecting >speech as noise. This happens when a speaker speaks for a while, and the >denoiser tries to remove their intonation (i.e. the sound of their >voicebox; the vowels). The result is that towards the ends of sentences, >their voice gets "thinned" out. > >I'd like to find out how to "slow down" the rate of adaptation within the >denoiser to make this less likely to happen. I assume that the tradeoff >is that...

...about Asterisk, I went on and tried out to create a little sample-setup. I'm using a Fritz Card USB with the AVM Capi Driver and two X-Lite Softphones. Dialing between the softphones makes no problem. Calling the MSN fron an external phone also works. I'm getting to the asterisk demo-voicebox which works flawlessly. Now may next step has been to enable dialing out with the softphones. This does not work as expected. I can dial out and the hard phone on the other end actually rings. When I answer it, I can hear nothing. Noting appears on the Asterisk console, X-Lite still talks abou...