similar to: ael context ~~s~~ in macros broke Dial() U() option in 1.6.2.17.2 and newer

Hi to all, i'm wondering to realize a dynamic macro that can take the number of extensions to RING,the ring type and all the parameter in a dynamic way. I have done this code to test it: macro pbx-ring-group-ael(pbx_id,num_int,ring_type,timeout,ext_string) { //; pbx_id = Id of PBX in the DB //; num_int = Quantity of extensions to ring //; ring_type = Kind of RING (C=contemporaneous

Am I doing something really stupid in this AEL macro, or is nesting an 'if' inside a 'switch', inside an 'if' not supported in the 1.2 AEL parser? macro stdexten( ext , dev ) { // First determine if the SIP peer is registered here Set(aretheyhere=${SIPPEER(${ext}:status)}); if(${aretheyhere:0:2}) == "OK") {

I have two offices sharing a phone system. They also share a common internal context because all of the employees of the second office also work for the first office. Each office has 4 outside lines and I have defined 2 channel groups in my zapata.conf. The second office needs all of their outgoing calls to go out over their lines so the people they call will have the correct callerID. I

Sorry, I forget to tell I tried, but not works. *Context:* context ivr_temp2 { s => { Proceeding(); str_time_01 = '06:00-12:00|*|*|*'; // Manh? ifTime (${str_time_01}) { Playback(ura/bom_dia); } } } The error is showed on "ael reload". *Console errors:* rs0000sr304*CLI> ael reload Command 'ael reload' failed.

"show dialplan" keeps showing contexts created by AEL. I tried deleting /etc/asterisk/extensions.ael but kept getting these messages in the Asterisk log: Feb 14 21:39:53 WARNING[6074] pbx_ael.c: Unable to open '/etc/asterisk/extensions.ael': No such file or directory Feb 14 21:39:53 WARNING[6074] pbx.c: Requested contexts didn't get merged Is there any way to delete or

R helpers, I have three questions about R and was wondering if I could get some help. First, I am trying to send the output of a a summary command - summary("info") to a bitmap file instead of the default device. Anybody know how to send a summary to a bitmap device? Second - this question is more difficult. What I am trying to do is have Perl (because I already have the data

Greetings list, I've just been upgrading one of our servers from 1.2.17 to 1.2.21.1-r1, and noticed that it's not picking up any of my macros written in AEL. Upon further examination, it looks like pbx_ael is missing. Is this a deliberate change, or is this something I need to address in the pre-compile configuration? Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited For

I have a caller screen queue setup. Basically a caller calls in, goes through a IVR, and before that caller is put into the queue, they get a sub ran on them first asking for them to say there name. That gets saved and they are entered into the queue using Queue(mainqueue,,,,300). In the queues.conf i have a list of members these are local/extension at external-default, there are two

Hello, I'm not understanding how to use GOSUB_RESULT in U() option from Dial app (I'm using 1.6.1) My extensions.ael is : context mylocal { 2 => { Dial(SIP/7530,,U(mynotify)); HangUp(); }; 3 => { Dial(SIP/7531); HangUp(); }; }; macro mynotify () { GOSUB_RESULT=ABORT; }; I

Hi, Using a 1.4 system in which dialplan is written using extensions.conf, I can use a custom .call file. On another system in which dialplan is written using extensions.ael, I can't use any custom .call file : system keeps replying : "apply_outgoing: At least one of app or extension (or keyword message/pdu) must be specified, along with tech and dest in file

I have two questions - 1. I am trying to create R png graphs via cron. I have this part working using Xvfb (X virtual frame buffer). One problem that I have, though, is that all the fonts on my graphs get messed up. Anybody have any nija R commands to make all fonts look great? Anybody have any idea how to fix this? So far, no luck on Solaris or Linux making the Xvfb fonts look good. I was

On starting Asterisk (1.4) I get a whole bunch of WARNING[5858]: pbx_ael.c:4040 ast_compile_ael2: Warning: file /etc/asterisk/extensions.ael, line 932-932: Empty Extension! I find it a bit disturbing that this message has a level of WARNING (instead of NOTICE maybe) because the extensions in question are empty on purpose. The only reason they exist are the hints. hint(SIP/3000) 3000 => {}

Hello-- I've just written and submitted a new module for asterisk, to the asterisk bug database. See http://bugs.digium.com/view.php?id=6021 There is a file there you can download, AEL2v0.3.patch.bz2 and I created a wiki page: http://www.voip-info.org/wiki/view/Asterisk+AEL2 Why did I do it? Because I was very impressed with AEL, but the current AEL compiler isn't real good at

All, I have a weird situation here and haven't been able to turn up any useful information in searches, so I thought I'd post to the list. Essentially, I have a customer who wants us to forward some of their calls to various cell phones. Normally, I'd use FollowMe() for this (that's how most of our customers are set up.) However, this particular customer has some requirements

Hello, trying to load ael module in asterisk ver 1.6.2 got the following: asterisk*CLI> module load pbx_ael.so Unable to load module pbx_ael.so Command 'module load pbx_ael.so' failed. [Feb 18 11:25:47] WARNING[7412]: loader.c:449 load_dynamic_module: Error loading module 'pbx_ael.so': /usr/lib/asterisk/modules/pbx_ael.so: undefined symbol: ast_compile_ael2 [Feb 18 11:25:47]

Hi, I compiled Asterisk 10.7.0 with gcc-4.5.3 and at runtime I'm getting these warnings: loader.c: Error loading module 'chan_dahdi.so': /usr/lib/asterisk/modules/chan_dahdi.so: undefined symbol: ast_smdi_interface_unref loader.c: Error loading module 'app_stack.so': /usr/lib/asterisk/modules/app_stack.so: undefined symbol: ast_agi_unregister loader.c: Error loading module

I have searched and searched for over a week on this but can't seem to find the issue. Calls from CallManager to Asterisk are being disconnected immediately. I have setup CallManager and Asterisk per Shaun Ewing's pdf http://asterisk.edropbox.net/ccmasteriskvm.pdf I have installed Asterisk 1.4.0-beta3 on Fedora Core 5. I got libpri, zaptel, and asterisk compiled and installed.

Hello, i'm using asterisk 1.4.9 in fedora 7, i was compiled its with this package: libpri-1.4.2 asterisk-1.4.9 spandsp-0.0.4 unicall-0.0.5pre1 libmfcr2-0.0.3 libsupertone-0.0.2 libunicall-0.0.3 zaptel-1.4.4 i'm using a E1 pci card with R2 but they not work, when I start the asterisk its generate this log: [May 12 08:53:24] WARNING[30814] channel.c: No channel type

All; I'm trying to install certified asterisk 11.6 cert16 on a Ubuntu 16 server. However, when I try to compile it, I'm getting hundreds and hundreds of errors. Here is a sample of the output. make[1]: Leaving directory '/usr/src/asterisk-certified-11.6-cert16/menuselect' [LD] aelparse.o aelbison.o pbx_ael.o hashtab.o lock.o ael_main.o ast_expr2f.o ast_expr2.o

one under my default context at extention.conf. And what is [pbx_config]? Thanks Eyal -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Tzafrir Cohen Sent: Friday, June 25, 2010 4:05 PM To: asterisk-users at lists.digium.com Subject: Re: [asterisk-users] Is there a default dial plan that is not in