similar to: Asterisk 1.6 / voicemail / final voice auth-thankyou

Hi there, I am using Asterisk 11.9 (with Sangoma-E1-Card/DAHDI) and it works fine :-) But I am wondering if there is a solution/application which will enable me to implement voice recognition while playing a voice file (barge in). So that the caller hears a voice file and can interrupt it with his voice. Currently (on our platform) the caller has to wait for the end of the voicefie. Then we play

Hi, I am trying to setup a GSM-Card for Asterisk. I currently use a "vgsm I" from voismart (http://www.voismart.it/) but the driver is very bad (compile-problems and no echo cancellation). Is there anybody out there who can recommend me another piece of hardware (pci card)? I need 1 or better 4 gsm-ports. Should be stable and have an echo cancelltaion feature. And of course it

Hi Thank you for your support. The server is actually compromised, I discovered that after making a deep trace using the audit daemon and looking for the kill signal (SIGKILL) that terminates asterisk. I discovered that there is an executable with a random name in the /boot folder that is killing and deleting asterisk !!! This executable is launched by a service in /etc/rc.d/ with the same

Hello, Limiting the call duration with the L-option of the Dial()-command is working fine, however the announcement is not played. Dialplan : exten => _367,n,Set(LIMIT_PLAYAUDIO_CALLER=yes) exten => _367,n,Dial(SIP/test6,,L(11000,5000,5000)) The call lasts for 11 seconds, but 5 minutes before time runs out an announcement should come. I hear no announcement, not on caller-side nor on

Hi, I am using Asterisk 11.5.1. As far as I understood, the following configuration allows a sip client only to receive calls (type=peer) but not to place calls (http://www.voip-info.org/wiki/view/Asterisk+sip+type). Why can I place calls though with this config? sip.conf ... [thorsten] type=peer host=dynamic context=my_context nat=force_rport,comedia secret=... dtmfmode=rfc2833 disallow=all

Hi, I looked for asterisk in /usr/sbin using the commands ls and find and whereis and it was not there. I know that the process is killed because when I start asterisk using the command asterisk -vvvvc it starts and then it exits and the word killed is wrote on the console. Ever time I copy a new executable to /usr/sbin either using cp command or make install it gets deleted too. Now I used

I'm in the process of setting up a queue system where the position message and thankyou message are required to play every 90 seconds. However, if a caller comes in to a queue with active agents logged in, and no one else is in the queue, the messages play immediately, and then the agents are polled. Is there any configuration parameter that will disable the playing of the messages if the

Did you take a look at /var/log/syslog? Am 26.11.2014 21:08, schrieb Antoine Megalla: > Hi, > > I looked for asterisk in /usr/sbin using the commands ls and find and > whereis and it was not there. > > I know that the process is killed because when I start asterisk using > the command asterisk -vvvvc it starts and then it exits and the word > killed is wrote on the

Hi, it's me again with a cdr-issue. I have the following example extensions.conf: # dial in exten => 100,1,Playback(hello) exten => 100,n,Dial(local/200,20,rtg) exten => 100,n,Playback(pleasewait) exten => 100,n,wait(10) exten => 100,n,Playback(goodbye) exten => 100,n,Hangup # for local dial exten => 200,1,Playback(hello) exten => 200,n,wait(10) exten =>

Hi, since version 1.4.12 the libpri package supports ETSI Explicit Call Transfer feature: http://downloads.asterisk.org/pub/telephony/libpri/ChangeLog-1.4.12 Does anyone know, how to use this feature in the dialplan? I can not find any hints in the asterisk doc. Best regards, -Thorsten-

Hi, I use Asterisk 11.5.1 and it works fine. :) Now I want to use TLS and media encryption. I followed this guide: https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial When I place a call via Blink-Client (0.5.0) I get connected and Blink shows 2 locks. The blue lock shows "Signaling is encrypted using TLS" and the orange lock shows "Media is encrypted using

Hi, I am trying now for over 4 hours setting up cdr-logging via odbc into a mysql database. But with no success. Do you have any hint for me? cat /etc/odbc.ini ------------------ [MySQL-asterisk] Description = MySQL ODBC Driver Driver = MySQL Socket = /var/run/mysqld/mysqld.sock Server = localhost User = xxx Password = xxx Database = asterisk Option = 3 Port = and /etc/odbcinst.ini

Hi, I am wondering, if there is any tool available, which performs a check for suspicious entries in the dialplan. For example a non existing AGI-Script or a double assigned extension ike that: [context] exten => *100*,1,AGI(test_app.pl) ... exten => 190,1,Answer() ... exten => *100*,1,AGI(never_reached.pl) ... A "normal dialplan reload command" would echo no warning or

Hi, In voicemail.conf: ; Supported values: ; 'filename' filename of a soundfile (single ticks around the filename ; required) ; ${VAR} variable substitution ; A or a Day of week (Saturday, Sunday, ...) ; B or b or h Month name (January, February, ...) ; d or e numeric day of month (first, second, ..., thirty-first) ; Y Year ; I or l

Hi there, Googling through the archives it looks like I'm the ferst person to want this... My aim is to set up a voicemail application with a custom greeting before *AND AFTER* the punter has left the message. Right now the relevant section of my dialplan is like this: exten => 2,1,Playback(/media/asterisk/answerphone-en) exten => 2,n,VoiceMail(2000,s) exten =>

Hello, We would like to continue a Perl AGI after a Dial() it has done completes following caller hangup. We would like to do this in the same AGI, and not using a new AGI from the 'h' extension. It works fine when the called party hangs up and the 'g' option is used, but not for caller hangup. Is this possible? If not a confirmation that this is the case would be very helpful.

thank you all for your help. Recently, ver 10 came out for scalix and it includes all the 'third party' pieces like JDK and Tomcat. So now I am reading up on kickstart to rebuild this system and start clean. I have copied all of my changed files (I hope!!) onto my notebook, and am doing the one mod i need for kickstart (original was a CD install, this time FTP). So I will know very

Time for a positive message. As a (very) part time administrator of a small network running on an ecclectic bunch of machines I'd like to thank the dovecot team for developing such an easy to use and faultlessly functional product. Having been forced into finally replacing my UW IMAP server, I'm glad to have hit on Dovecot after struggling with mailbox transfer issues with cyrus and

sorry for the misunderstanding, robert! the point is: during the caller is listening to the soundfile played to him the dialplan should continue to dial the sip phone 100 and after sip phone 100 is answered and the announcement file is played the caller should be connected to the sip phone 100. the behaviour is just like MoH, but the problem is, that the caller has to hear a soundfile from the

2008/12/9 Olivier <oza-4h07 at myamail.com> > > > 2008/12/9 Tilghman Lesher <tilghman at mail.jeffandtilghman.com> > > On Tuesday 09 December 2008 09:14:11 Olivier wrote: >> > Hi, >> > >> > In voicemail.conf: >> > ; Supported values: >> > ; 'filename' filename of a soundfile (single ticks around the >>