similar to: Asterisk vs silence detection

Hello there, I tried to found documentation about openloop disconnect on Asterisk/Zaptel. And up to now, I didn't find anything. Is openloop disconnect supported by zaptel/wcfxo drivers? Other question: maxsilence and silencethreshold don't seem to have any effect on voicemail. Could it be possible that in certain situation it just wont work? I tried adding some verbose in

What I'm trying to achieve is that a voicemail message should be at least 3 seconds long for it to be saved, but *after that* a prolonged silence (e.g. 10 seconds) should terminate the call and recording. My current settings (Asterisk 10.7.0 and 11.2.1) are: ; Minimum length of a voicemail message in seconds for the message to be kept ; The default is no minimum. minsecs=3 ;

Is there some way * can trim the trailing silence in a voicemail message? There's the "maxsilence" setting for silence detection which is related to what I'm asking but not the same. Let's say I set the maxsilence to 8 seconds. During the recording of a voicemail, if someone doesn't say anything for 8 seconds, the recording ends. However, the recording still has

Hi all, i use asterisk 11.14.0 and I suspect that the voicemail application counts the time wrong. In my voicemail.conf: [general] minsecs=3 maxsilence=5 format=wav maxsecs=180 silencethreshold=140 [...cut..] In the asterisk-cli: [Jan 26 15:23:49] -- Executing [s at macro-voicemail:77]VoiceMail("SIP/XY-0005175a", "aNumber,su") in new stack [Jan 26 15:24:04] --

It seems voicemail recordings have broken sound. It cuts out randomly throughout the recording. Has anyone had any similar experiences? I've included some snips of my voicemail.conf Cheers, Ben ----------SNIP------- [general] ; Default formats for writing Voicemail ;format=g723sf|wav49|wav format=wav ; Who the e-mail notification should appear to come from

I'm having tons of trouble getting the fromstring to work in voicemail.conf. I've tried both voicemail and voicemail2 but the emails still seem to be coming from asterisk pbx. Has anyone had any luck with this? ================= Here's my voicemail.conf: ; ; Voicemail Configuration ; [general] ; Default formats for writing Voicemail ;format=g723sf|wav49|wav format=wav49|gsm|wav ;

When someone calls into our asterisk server over a PSTN line, dials an extension and then hangs up, the SIP phone related to the given extension will ring about 4 or 5 times before asterisk shows that the channel has been hung up in the console. This isn't such a big deal on its own, but what's happening now is that if a user calls in from a PSTN line, gets voicemail on the extension, and

It should be as simple as editing voicemail.conf : ; Voicemail Configuration ; [general] ; Formats for writing Voicemail. Note that when using IMAP storage for ; voicemail, only the first format specified will be used. format=wav49|wav|gsm ; Who the e-mail notification should appear to come from serveremail=asterisk-voicemail ; Should the email contain the voicemail as an attachment attach=yes ;

I have a call server that runs on a few custom AGI scripts initiating calls and then managing the calls. I'm getting stuck on the detecting silence functions. I wanted to use the silence detecting as a quick method of substituting Answering Machine Detection. However, whenever WaitForSilence is supposed to be detecting silence, it always just ends the interval whether or not there is

I have a call server that runs on a few custom AGI scripts initiating calls and then managing the calls. I'm getting stuck on the detecting silence functions. I wanted to use the silence detecting as a quick method of substituting Answering Machine Detection. However, whenever WaitForSilence is supposed to be detecting silence, it always just ends the interval whether or not there is

Ok its probabally something really eaisy im missing. I've searched the archives and voip-info. Asterisk is trying to send the email notification for voice mail. But the log says Invalid sender. Sender = asterisk@localhost.localdomain and not asterisk@somedomain.com as assigned in conf file. VM Config: [general] format=gsm|wav49|wav serveremail=asterisk@somedomain.com <----Actual

The advanced voice mail options in RC1 is not providing any functionality except "4 - place outgoing call" and "* - return to mail menu". Below is a sample line from voice mail.conf: 210 => 234,test1,test@dotality.com,,|attach=no|cid=yes|review=yes The is a rew install from the tgz file. No patched have been made to the source code. I did try to apply the advanced10

Folks, I'm trying to get Asterisk to load my voicemail configuration from MySQL. I've followed the instructions at: http://www.voip-info.org/wiki-Asterisk+voicemail+database I restarted Asterisk, but no luck: the voicemail.conf does not get updated. I started with a sample voicemail.conf that I found on the Wiki. Or was it from Voicepulse? I can't remember. For initial testing, I

Hi, We have a system that recorded voicemail for about an hour after the caller hungup. I'm going to put a timeout on it but is there anything to look for that can help prevent this? The system is running on a telenet line in Belgium. The answer dialplan I used was: [macro-stddial] exten => s,1,Answer exten => s,2,Playback(transfer) exten => s,3,Dial(${ARG2},60) exten =>

Hi everybody, I'm trying my Asterisk to send emails when a new message arribes to a voicemail user but no email arribes. my voicemail configuration is the following: VOICEMAIL.CONF: [general] format=wav serveremail=asterisk at mydomain.com attach=yes maxmsg=20 maxsecs=180 minsecs=3 maxsilence=10 silencethreshold=128 maxlogins=3 fromstring="My Asterisk" When I look at maillog file,

Here is what happens on startup: Sep 15 13:23:51 DEBUG[28130] res_config_mysql.c: MySQL RealTime: Static SQL: SELECT category, var_name, var_val, cat_metric FROM settings WHERE filename='voicemail.conf' and commented=0 ORDER BY filename, cat_metric desc, var_metric asc, category, var_name, var_val, id Sep 15 13:23:51 DEBUG[28130] res_config_mysql.c: MySQL RealTime: Everything is fine. Sep

Hello, I only know a nice tool for JACK, called silentjack. So if by any chance, you find the strength to use JACK anyway, here is the source code : http://www.aelius.com/njh/silentjack/ The code is very light and simple : if you have some programming skills, you might even adapt it for ALSA. Sorry for this OT answer. Hoggins! Le 09/08/2014 02:39, Dean Sauer a ?crit : > Looking to

On 08/09/2014 12:39 AM, Dean Sauer wrote: > Looking to find something that can do silence detection on a feed from > Icecast 2.3.2 with a silence detection period passed as a variable > > ie: sd my.server.invalid:8000/mymount -t 20m -s /home/myuser/sendalart.sh > > > On detection of silence exceeding this period it calls /home/myuser/ > sendalart.sh to send an email, or

Hi, I have upgraded my PBX to Asterisk 1.2.5 , previously I was using Asterisk 1.0.9, and Every thing was working fine ,But now voice mail is not working. The error I am receiving in log files is like following, WARNING[2413] app_voicemail.c: No entry in voicemail config file for '12' I have searched for solution a lot can Any one of you let me know how can I solve this issue

Asterisk Users, I am running Asterisk-1.4.11 with Zaptel-1.4.4 on Debian Etch System 2.9.18-4-amd64. A TDM03B is installed on the Debian System. Every time, I try to change my voicemail pin via the Sip phone, the voicemail.conf does not get modify and I see this warning message on the Asterisk command line: [Aug 29 00:12:23] WARNING[19142]: app_voicemail.c:799 vm_change_password: