similar to: agi exec problem (followup)

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

as i said, right now i'm just getting my feet wet. but, i will be needing to build dialplans on the fly. 'add extension' seems like the right call to make. .t > What is the goal of this? It doesn't make much sense to me. Care to > share some insite into what your goal is? > > bkw > > On Sun, 23 Nov 2003, tad wrote: > > > actually, i do have a

hey can anyone help us out with this one? we're working on a system for our community radio station that uses a mysql/php interface a database to manage users and a schedule system. since we have different copies of liveice and ices on different machines for different applicaitons around campus (they are all on different icecast mountpoints on our main server), we need our php script to

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);

On Sat, Aug 12, 2017 at 06:49:01PM -0700, John R Pierce wrote: > On 8/12/2017 6:25 PM, Fred Smith wrote: > >I just stumbled over /var/log/CodeMeter, which contains a number of large > >log files. > > > >I know I'm getting old and forgetful, but I can't remember intentionally > >installing that package. > > > >Yum just says "installed"

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

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

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

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

Anyway, in my P.S. yesterday (the main post was on Codec problems), I described a situation where any IAX softphone was registering successfully, and then having zero sounds heard on either side of the call. Here is an "iax2 debug" output from a DIAX call to a local * server, dialing the extension that goes directly to the "demo" application. AsteriskHouse*CLI> iax2

Hi Rowland, not for all users, some users have a gidNumber not inside the range. I expected to see the users with a gidNumber insite the range. It was a classicupgrad from a very old samba3 so we have gidNumbers below 100 for some users, but we only want to have the users with a gidNumber insite the range. So we don't care about the users outside the range. Stefan > > Have you

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