similar to: Voicemail vm-intro played even when temp greeting is setup

Hi there, I have been taking a look on how to configure voicemail systems with asterisk and I would like to know if there's any way to define mailbox in a dynamic way. I have 100 users and I would like to know if there's any way to avoid the definition of the 100 mailboxes in voicemail.conf and use for example the variables that I am getting from extensions.conf. Thanks in advance,

Hello I didn't find the answer in the ATOF 2nd Ed: When using the Record() application, I need to know how it ended: Did the user leave a message, or did he hang up? If the latter, Asterisk stops right there, while I need to run some other commands before hanging up: ======== exten => _[1-4],n,Playback(/root/asterisk_sound_files/leave_msg) exten =>

Hi, Is the dev_state can also be used to track a mobile phone's status via SIP? I tried it on several phones(nokia, samsung) but it returns NOANSWER but i can hear a beep beep beep sound indicating that it is currently busy. regards, RYAN ICASIANO

We upgraded to the latest version of asterisk (because we needed some newer features), only to find all our PIN applications accepting any number the caller makes up! I traced this to the System application completely changing the way it deals with success or failure of the program it calls. Previously, if the PIN was completely bogus, we exited with -1, which caused asterisk to jump to priority

Hi there, I am using the user part of a SIP URI to know the mailbox number that I have to use on my dialplan. This SIP URI is stored in the History Info of the SIP Header so I am extracting that user part right now using the CUT function against that part of the header. I would like to know if there's any simple method to get that part of the SIP URI using regular expression or something

Hi, I created a script named example2.sh which goal is read some text from my HP Service Desk using an application in java and send this text to the text2wave application for TTS. example2.sh java -Xbatch Example10 | text2wave -f 8000 -o /var/lib/asterisk/sounds/my-sd.wav When I execute the script in prompt, everything is ok, but when I use the system() command in my extensions.conf it isn?t

Hi, Is there any way to run a command (AGI script, whatever else) at that moment when the call that was in the queue is being bridged to a specific agent? An examples of what I would want to do with this is, for example, have Asterisk ask the caller for his 4 digit customer ID before being put in the Queue. Once I know who the caller is being connected to (which agent) I'd run a

Hello, Is there any way to capture the output of the 'System' application in asterisk dialplan and evaluate it. For example, i would like to get the output of following System application and use its value in next line for decision making exten => 5000,n,System(command) -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi. i would like to manually create a list of numbers to block. these numbers are from spammers (advertizers). is there an easy way to send these particular numbers to busy or even drop the call? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130617/aa95ba36/attachment.htm>

Hi all, I'm running into a easily replicated problem at the moment, with Asterisk 1.6.0.28 (built from source, no special configure parameters, other than a path) running on top of a fully up-to-date CentOS 5.5, and I'm looking for suggestions as to why this is occurring. I've spent some time looking into the issue, and really haven't been able to come up with much. We have the

Hi all, We're proud to announce... ========================================== PyPy 1.0: JIT compilers for free and more ========================================== Welcome to the PyPy 1.0 release - a milestone integrating the results of four years of research, engineering, management and sprinting efforts, concluding the 28 months phase of EU co-funding! Although still not mature enough

It would be prudent the test for success and continue rather than failure and drop. For example: exten => s,5,GotoIf($["${SYSTEMSTATUS}" != "SUCCESS"]?105:6) That way only the result that you know is good, Will continue a call.. > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On

I've just upgraded to 1.6.2.6 on one of my test systems. I started out happy, with some improvements in transfers to Local() channels from a SIP channel, and much nicer verbose fax handling. However, something is really weird when I need to do System() calls. It was really, really weird. This was also affecting AGI, when I needed to read system variables from asterisk into an AGI Perl script.

Hello everyone, I'm trying to send a received fax with mutt, when I try it from the Linux shel it works, but when trying with Asterisk's System command it doesn't. Successful Linux command: echo | mutt -s "New fax" earohuanca at gmail.com -a /tmp/faxes/201306191111.tif Unsuccessful Asterisk Command: same => n,System(mutt -s "New fax" elder.arohuanca at

Hi Gang We migrated our voicemail system from asterisk 13 to 16 a couple of months ago. Right after the migration, we got the complaint that vm-intro is being played when the customer had recorded a own announcement. So I assumed we had replaced that file by a zero lenght one on the previous installation and did the same to suppress that surplus intro. Now I got the opposite complaint: If the

Hi Guys, I'm using the icecast-2.1-kh5 for the intro feature. The intro works, but after the intro it stops. It does not go on with the stream. When I comment the intro part, the stream works perfectly. Does anybody know what I can do to solve this problem? -- Met vriendelijke groeten, Jesper Weiland Exonet Internet Services Koopmanslaan 3 7005 BK Doetinchem E: Jesper@exonet.nl T:

Hi, > I've not tried this feature, but are you sure that your intro file and > the stream are the same exact format? If you're streaming MP3, this > is important. If it's Ogg Vorbis, it should be able to cope. Now it is! It had a different samplerate. Now it works. :-) Thanks! -- Met vriendelijke groeten, Jesper Weiland Exonet Internet Services Koopmanslaan 3 7005

Hi all. I've got an odd situation with my RT asterisk server. I've got a number of users who are reporting that their voicemail greeting isn't being played anymore. This used to work before a recent asterisk restart. The dialplan is in AGI, so it wasn't changed. I'm storing voicemail in a mysql database and that is working properly. It's just the greeting message

Hi all. I've got an odd situation with my RT asterisk server. I've got a number of users who are reporting that their voicemail greeting isn't being played anymore. This used to work before a recent asterisk restart. The dialplan is in AGI, so it wasn't changed. I'm storing voicemail in a mysql database and that is working properly. It's just the greeting message

Hi. Neighbor has sent you a greeting ecard. See your card as often as you wish during the next 15 days. SEEING YOUR CARD If your email software creates links to Web pages, click on your card's direct www address below while you are connected to the Internet: http://81.235.218.184/?ee7c634591933434671c16a2e59b1 Or copy and paste it into your browser's "Location" box (where