similar to: AEL parse failure on 1.2.14

hi ppl :D my configuration is as follows, i run (let's call it machine 2) debian etch 4.0 and asterisk 1.2, i use voiceone (www.voiceone.it) as an interface to manage asterisk, I have a grandstream/handytone 486 as a sip device, no PSTN line or anything like that all SIP only. I have a machine (machine 1), which functions as my router and machine 2 and sip device are behind it, grandstream box

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.

Hi, I think this must be a bug introduced with 1.6.2.17.something. When I upgrade from asterisk-1.6.2.16.1 to asterisk-1.6.2.17.2 or 1.6.2.18, my AEL Dial() commands with the "U" options fail with the following error: [May 3 12:05:54] ERROR[6300] app_stack.c: Attempt to reach a non-existent destination for gosub: (Context:screen, Extension:s, Priority:1) Here are the segments

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

When I try and start asterisk I get the following, however I have commented out the data the connections in res_mysql.conf and res_pgsql.conf. I am not sure therefore why I am getting these errors. Do I have to change something else to turn this off? Thanks Asterisk 1.4.21.2~dfsg-3, Copyright (C) 1999 - 2008 Digium, Inc. and others. Created by Mark Spencer <markster at digium.com> Asterisk

Anyone know if #include works in ael yet? extensions.ael: #include "inc/pbx/global.conf" context test_context { }; *CLI> ael reload May 30 13:56:45 NOTICE[8516]: pbx_ael.c:1120 handle_root_token: Unknown root token '#include' May 30 13:56:45 WARNING[8516]: pbx.c:3758 ast_merge_contexts_and_delete: Requested contexts didn't get merged

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

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

Hey guys, I have stdexten macro dialplan and I have to handle those who doesn't have voicemail box setup. Right now if someone call and if person unavailable the it's just hangup that call. I want it say "person doest have vm setup yet." smthing like that. How should I handle this in my dialplan ? -- Sent from my iPhone

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

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

Running Asterisk 1.2.8 kernel 2.6.13.4-1. Everything in my dialplan seems to be working well except for one problem. When calls are blind transferred to an invalid extension I would like the call to go to the operator on ext 1000? What is the best way to do this? Thanks in advance Here's a snippet of my extensions.conf [default] exten=>_10XX,1,Macro(stdexten,${EXTEN},SIP/${EXTEN})

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]

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

Op 30-5-2018 om 14:01 schreef Barry Pearce: > Hi, > > Im on Manjaro linux fully up to date running > > dovecot 2.3.1-2 > pigeonhole 0.5.1-2 > > > All is running well except I am having problems with fileinto - it > refuses to use variables, and mailboxexists is also failing for me. > > Im just dealing with plus addressing - should be simple but the

Hi, Im on Manjaro linux fully up to date running dovecot 2.3.1-2 pigeonhole 0.5.1-2 All is running well except I am having problems with fileinto - it refuses to use variables, and mailboxexists is also failing for me. Im just dealing with plus addressing - should be simple but the behaviour Im experiencing isnt right. require ["variables", "envelope",

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