similar to: Re: Asterisk-Users digest, Vol 1 #1994 - 14 msgs

actually, i do have a workaround which bypasses the exec command entirely: system("asterisk -r -x 'add extension s,3,Playback(demo-congrats) into local'"); but it's ugly. seems like it should be possible to do this with exec. .t ---------- Forwarded message ---------- Date: Sun, 23 Nov 2003 21:17:50 -0500 (EST) From: tad <tad@media.mit.edu> To:

hi folks. (apologies in advance if this is a particularly stupid question) just getting my feet wet with asterisk / agi, and am a little stuck using EXEC. it works fine for applicaitons that take simple arguments, but chokes on applications that require multiple words as arguments. for example, this works fine: EXEC Playback(demo-congrats) but this doesn't: EXEC add extension

you can do this with MeetMe, but you don't have to. you can also use Parking, which makes things a little simpler. in either case, the strategy is going to be something like: 1. Record the soundfile 2. Park the inbound caller 3. Use a .call file or the manager interface to initiate an outbound call to the mobile phone 4. play soundfile and prompt the mobile phone user to accept/reject the

Does anyone know how to get CID working in AEL2 ? In extensions.conf you can do: exten => 111/666,1,PlayBack(demo-congrats) exten => 111/666,2,Hangup() exten => 111,1,PlayBack(demo-moreinfo) exten => 111,2,Hangup() and if callerid 666 dialed 111, they would get demo-congrats, everyone else gets demo-moreinfo. In AEL: 111 => { Playback(demo-moreinfo);

I'm having trouble with alignment of a trend line overlayed onto an image plot. The two should be plotted on the same x-axis (time-series). However, the trend line begins about an inch into the image plot x-axis and ends about an inch off of end of the image plot. Once I have the alignment sorted, I need to put a secondary y-axis on the image plot which is scaled for the trend line. An

Hi, I have cvs updated all my modules (zapata, libpri, zaptel and asterisk). I have also read in the archives & seems that no-one has run into this problem. What I'm trying to do is simple. Just make and outbound call using the /var/spool/asterisk/outgoing directory. I copied /usr/src/asterisk/sample.call and only changed the context & extension. I configured my Zap1 to the same

Hi! I just installed Asterisk 0.4.0 with all the default options, and the configuration samples it has. When I try to dial from an h323 client (gnomemeeting) I get this message on the messages file: Jun 19 11:48:45 WARNING[15375]: File file.c, Line 410 (ast_openstream): File demo-congrats does not exist in any format Jun 19 11:48:45 WARNING[15375]: File file.c, Line 553 (ast_streamfile):

I am recieving inbound calls to my asterisk box over IAX. And getting "spawn extension....exited non-zero" errors. Im not entirely sure what this means, and would appreciate any help in fixing my problem. FYI: ********** is the inbound phone number x.x.x.x is a remote asterisk box calling my own asterisk box. When I choose it to dial an internal extension using this dialplan: exten

My first Asterisk install: Debian sarge with the 2.6 kernel, and two X-Lite soft-phones. I followed the online how-to documents and was calling between the two soft-phones and calling the demo system with no problems and had full audio. I then went on to configure the TDM400P's two FXS modules. I got into that a ways and was having some success, but no dial-tone when I was off the

Have wine running in Ubuntu - dapper. Found and insalled winexs which is SO good. However the process of starting a programs exe is a tad time consuming and prone to error. Well for me who is a tad used to double click and it's away sort of windows xp.... Ubuntu Applications > {whatever} seem like a great spot to have menu items of stuff to run under wine but for the life of me I can't

Hi, the first thing I did with Asterisk is listening to `demo-congrats' by Xlite on the same machine. This works perfectly. The config files are those shipped with the package. Now I want to listen to it over ISDN/Capi but I don't succeed. My `capi.conf' is like show in many tutorial on the web. In `extensions.conf' I just added the following lines: [capi-in] exten =>

I have 12 T1's going into 3 servers, 4 in each into "Digium, Inc. Wildcard TE410P Quad-Span togglable E1/T1/J1 card 3.3v (rev 02)" cards. Each "group" of T1's have the primary D on 24 and the secondary D on 96. The first server (ts20) and the last server (ts22) can playback "demo-congrats" fine. The "middle" server (ts21) cannot -- just dead air.

in sip.conf [general] port = 5060 ; The TCP/IP port for SIP communiations bindaddr = 0.0.0.0 ; Address to bind to. 0.0.0.0 all addresses on server. context=other ; Default for incoming calls disallow=all allow=ulaw allow=gsm in extensions.conf [general] static=yes ; These two lines prevent the command-line interface writeprotect=yes ; from overwriting the config file. Leave them here

Folks, Apologies ahead of time if this has already been asked (read the list for the last month looking for something similar). I have been trying to get the Background command to work with no joy yet. Here is what I am trying to do: 1. Answer the call. 2. Play the message in the background, while waiting on DTMF from user. 3. If I get a "1", then interrupt the message and dial the

Brian Takita committed revision 1000 last night. I don''t know what that means - but it seems like an interesting milestone. Congrats to Brian for hitting 1000. Congrats and thank to all of those who have contributed by providing ideas, code, feedback, or simply using rspec and spreading the love. Cheers, David

Hi, I just setup a box with RH 9, and latest asterisk via CVS. The box as a T100P card in it that is currently hooked up to nothing. I did have the sample configs in place via make samples, and the only change I made was to add an entry to sip.conf for my Cisco 7960. When I dial 1000 to get to the main greeting I hear nothing, though the command line output looks fine to me. Any ideas? --

Hi, I just got two digium x100p clones and installed asterisk on fedora core 2 which took some tweaking. After getting asterisk up I installed the zaptel stuff - then modprobed zaptel, wcfxs (for the fxo cards), which worked fine. ztcfg is showing two channels configured, but when I start asterisk and do show channels, i see no active channels. zapata.conf has: signalling = fxs_ks

Hi, I''ve just now started learning Ruby on Rails. Someone told me that I should get RadRails to use as my IDE. RadRails.org seems to be dead now. Has something bad happened? Or do I just have my URL wrong? Is there another good Rails IDE? I''m developing on windows right now. Thanks! TAD -- Posted via http://www.ruby-forum.com/.

[Something seems to have went wrong with my previous posting. It appears on the archive page in another thread. I did not receive anything myself. So I may give it another try:] Hi, the first thing I did with Asterisk is listening to `demo-congrats' by Xlite on the same machine. This works perfectly. The config files are those shipped with the package. Now I want to listen to it over

All, After installing asterisk on Linux, I run "asterisk -vvvc". But I got the following warning message: chan_oss.so] => (OSS Console Channel Driver) May 26 00:37:58 WARNING[-1084845952]: chan_oss.c:980 load_module: XXX I don't work right with non-full duplex sound cards XXX == Registered channel type 'Console' (OSS Console Channel Driver) == Parsing