search for: voiceboxes

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

Hi all I noticed that most caller are quite confused by the standard voicemail announcement text. 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

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.

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

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.

hi list, I'm looking for a way to execute commands in my dialplan specifically when a caller has hung up. my curretn dialplan 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

Salve, I'm somewhat stuck on how to get DTMF working with my setup and googling didn't yield anything similar. My setup consists of one 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.

Tom Harper wrote: > Hi All & Jean Marc, > > Once again I find myself delving into the pre-processing code to fiddle > with the VAD, AGC and denoising code. > > Where i am at is that I have implemented all of Steve Kann's mods, and > they are 90% of the way there in terms of working, except that I am still > having issues denoising open air mics. But that is

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 exclud...

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 /lib/modules/2.4.18/misc/ztdummy.o: insmod

Hi there, I just finished work on my current PBX dialplan configuration. I didn't find anything compareable yet 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) -

hi! I'm working an a asterisk test project at college at the moment. right now we're experiencing two problems. calling our sipphone (optipoint400) from a firefly client leaves us with no audio (no noise...nothing at all...) [the phone is ringing however and the connection seems to be set up] other way round works just fine!! firefly2firefly (stun enabled) also works

Thought I might just as well share these scripts, they may work with other phones too: ######################################################################## *1)* Dialing from the KDE 3.5.5 address book works with a script that gets triggered from the kaddressbook (Settings - Script Hooks - Phone) with my command snom_dial_number %N The script snom_dial_number itself goes like this:

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

I can't check the voicemail for the switchboard. Asterisk hangs up for some unknown reason... ----- s n i p ----- -- Executing [*500 at default:1] Wait("SIP/597-00f0c410", "1") in new stack -- Executing [*500 at default:2] VMAuthenticate("SIP/597-00f0c410", "500 at default|s") in new stack -- <SIP/597-00f0c410> Playing

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

Steve, The main problem I am having with the system is clipping off the start of someone's speech when they first start talking- the ends of the sentences seem to be handled properly. I am wondering whether this is the fault of the audio playback system or whether this is a speex issue- I also get the musical artifacts problem with the denoiser. This seems to be more of a problem on open

Hi, after having read so much 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