similar to: indications.conf

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

Hello I use asterisk ver 0.7.2 Can I play any wave file into the client riciever without billing count ? I call from A IAX client to B IAX client. B client is not available and I would like to play some file with the message user_is_unavailable.gsm But when I look into my CDR table, this call is billed. I don't want to bill these messages. Is it possible ? thank you -- Vit Bohacek

Hi, Sorry for my bad english but I?ll try to explain my problem I got an Asterisk running in my house with ADSL... I?m using X100P and TDM400P cards.... My intention is get calls via PSTN to my house and Redirect to my computer in my work using X-Lite by SIP... Here?s the map with Firewalls Call for anyone to my house => PSTN => X100P => EXTENSIONS => SIP/RTP => ISA MICROSOFT

Here is a question for all you routing guru's out there.. I am using an ADSL line (512/256Kbps) to connect from the internet to my Asterisk server.. At a point I will run out of bandwidth so the cheapest option would be to add a second ADSL line.. The problem is how will the routing work? If I put 2 IP's on one NIC will the return traffice be routed back via the gatway of the IP that

I am trying to reconfigure an asterisk box that was using an HFC-S card with bristuff but is now using 2 analog lines therefore I want to use the TDM02B to connect to two POTS lines. The TDM02B has 2 red modules. I have this in /etc/zaptel.conf loadzone=nl defaultzone=nl fxsks=1-2 I have /etc/asterisk/zapata.conf signalling=fxs_ks echocancel=yes echocancelwhenbridged=yes echotraining=400

Good morning everybody, Can someone explain to me the interconnection between these four things: indications.conf, SetLanguage(), zaptel.conf and ring-back ? If there is any !! :- ) I am having this case where some users cannot hear ring back from a DeadAGI script and it seems to be interconnected to these items. These users are from the iaxfriends table, they _can_ hear ring-back from a

Hi, Not sure where to submit this to so I'll try here. Below is the toneset for Iceland. Hopefully this can be added into the asterisk package. [is] description = Iceland ringcadence = 1000,4000 dial = 425 busy = 425/250,0/250 ring = 425/1000,0/5000 congestion = 425+250/250,0/250 callwaiting = 600/100,0/100,600/100,0/9000 record = 1400/500,0/15000 info = !950/330,!1400/330,!1800/330,0

Dear R-community, In graphical representations of findings concerning bodies of standing water (lakes e.g.) in x-y-plots you often make use of a somewhat different definition of coordinates in a cartesian system: the origin is top-left, the x-axis (depth of the water body) from top to bottom and the f(x)-axis from left to right, so you can project the graphical representation of data in your

Hello list, Hope you all doing fine! I've tried to use the 'alias' directive in the indications.conf file but apparently it doesn't work.... It looks like maybe this feature was removed, because old sample for the indications.conf file have example using the alias parameter, but newer samples don't have it anymore.... also I couldn't find any ticket saying this parameter

I've the following setup : sip phone -> ser (auth and routing) -> asterisk with capi isdn when I call a pstn number everything works fine, but I can't hear anything till the called answer. this is the output from a test call : -- Executing Playtones("SIP/2.7.184.61-08152880", "dial") in new stack -- Executing Dial("SIP/2.7.184.61-08152880",

Aqu? tienes, [es] description = Spain ringcadence = 1500,3000 dial = 425 busy = 425/200,0/200 ring = 425/1500,0/3000 congestion = 425/200,0/200,425/200,0/200,425/200,0/600 callwaiting = 425/175,0/175,425/175,0/3500 dialrecall = !425/200,!0/200,!425/200,!0/200,!425/200,!0/200,425 record = 1400/500,0/15000 info = 950/330,0/1000 dialout = 500 Sergio Serrano Revuelto Avanzada 7 Original Message:

Hello, I have a problem with connecting a Digium X100P card to a Brazilian analog line. Can somebody help me out with this problem? My /etc/zaptel.conf is loadzone=br defaultzone=br fxsks=1 My /etc/asterisk/indications.conf [general] country=br [br] description = Brazil ringcadance = 1000,4000 dial = 425 busy = 425/250,0/250 ring = 425/1000,0/4000 congestion =

Hi there, I have a new asterisk working in New Zeland and everything is working well except when an incoming call to the PSTN hangs up, asterisk wont hang up the zap trunk (X100P). I have found this information: http://bugs.digium.com/bug_view_page.php?bug_id=0001474 Which discusses my problem and i have made sure that i have the latest info in the indications.conf as follows: [general]

Hello I have a problem with my MGCP voice gateway. I use D-Link DG104S Boot PROM Version 3.0B38-D Firmware Version 3.0T86-D I tried asterisk v 0.7.2 and I am using latest CVS version now. When I dial a number very fast, or when I use a redial function, my asterisk receives coupled digits. My co-worker called number 245005111, these are a few lines of my debug. The identifier of first digit

Hi, Does anybody know where I can find any open source ITU G.107 implementation available? I'm looking a way to measure the voice quality in my projects.. Thanks in Advanced, My Best Regards, Andre Lomonaco -------------- next part -------------- An HTML attachment was scrubbed... URL:

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Signed-off-by: Viswanath Puttagunta <viswanath.puttagunta at linaro.org> --- celt/tests/test_unit_dft.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/celt/tests/test_unit_dft.c b/celt/tests/test_unit_dft.c index 5ea10fb..111c249 100644 --- a/celt/tests/test_unit_dft.c +++ b/celt/tests/test_unit_dft.c @@ -177,8 +177,14 @@ int main(int argc,char ** argv)

Signed-off-by: Viswanath Puttagunta <viswanath.puttagunta at linaro.org> --- celt/tests/test_unit_dft.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/celt/tests/test_unit_dft.c b/celt/tests/test_unit_dft.c index e17e26f..f08eb65 100644 --- a/celt/tests/test_unit_dft.c +++ b/celt/tests/test_unit_dft.c @@ -177,8 +177,14 @@ int main(int argc,char ** argv)

Hi list, Has anyone been able to get ISDN-BRI support to work reliably on Asterisk 1.4? If so, I'd love to know how you did it (hardware, distro, kernel, modules, versions, config files). I've tried to get it to work on a Debian etch system with an HFC-PCI card and the zaptel package (v1.4.7, also from xorcom.com), but with no luck: all three channels that are created when the

Can anyone tell me if there is a way to vary the output levels (dB) of the tones generated in indications.conf? I generate a few custom tones and sometimes people tell me they are a little too loud. Thanks Lee -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090522/7ae8c75d/attachment.htm