similar to: How to write custom functions in AEL2 ,

Call to Action! For those who have the courage/ability, go grab an SVN copy of the asterisk release, the HEAD version, and my latest patch, from: http://bugs.digium.com/view.php?id=6021 Right now, the latest version of the patch is 0.10. apply it to the SVN head version, and do a "make". Read the Wiki on AEL2: http://www.voip-info.org/wiki/view/Asterisk+AEL2 Look at the examples

Hey all, I am very interested in using AEL2 (don't want to upgrade to 1.4 to get it though), but am having some problems upgrading/patching my asterisk system. I am following the instructions on the wiki: http://www.voip-info.org/wiki/view/Asterisk+AEL2#AEL2AnnouncementsandNews But get the following error:

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 am struggling to find out how to code hint in AEL2. I did hint(Custom:light1) and it keeps complaining about the : (colon). It works fine for SIP device like hint(SIP/439). Anyone who has tried it before?

I just want to confirm but it seems that if-then-else is not permitted in case structure. It was not really documented but it seems to be the case. Can anyone confirm? switch(${DIALSTATUS}) { case NOANSWER: { // if-then-else not permitted If (${ael-var} = 1) { Playback(beep);

Where's Steve Murphy when you need him? :-) This doesn't seem to work in AEL2... Macro foo(arg1) { ..... Includes { Hangup; } } The error is: File: /etc/asterisk/extensions.ael, Line 59, Cols: 5-12: Error: syntax error, unexpected KW_INCLUDES, expecting '}' The same error does not occur when the includes is in a context. I need to

Hi, I'm trying to pass a single carriage return (0x0d in hexa) to Sendtext in an AEL2 file like this : SendText(${BASE64_DECODE(DQ==)}); Value sent (8 bytes long) is very strange : Content-Type: text/plain;charset=UTF-8 Content-Length: 8 ?ez?== Any workaround ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, When reloading a diaplan (asterisk 1.6.1.X), I can see in console : [Apr 2 09:02:00] WARNING[2217]: ael/pval.c:2522 check_pval_item: Warning: file /etc/asterisk/extensions.ael, line 621-621: application call to Gosub affects flow of control, and needs to be re-written using AEL if, while, goto, etc. keywords instead! What is then the recommended substitution for Gosub() application

Hi, all. I'm making my first foray into AEL. I'm starting with a simple macro, but I've already encountered an odd behaviour. I'm wondering if someone can shed some insight: Asterisk 1.2.9.1 macro newPlaceCallPSTN { s => { TIMEOUT(absolute)=7200; NoOp(Analog Out List: ${ANALOGOUT}); ChanIsAvail(${ANALOGOUT}); NoOp(Available Out List:

Hi! All the AEL examples have a semicolon after the closing curly bracket, e.g: context test { 1 => Hangup(); }; but without ; it works fine too, e.g: context test { 1 => Hangup(); } So - what is the reason for the ; after the closing curly bracket? thanks klaus

Hi all, I've just upgraded to latest 1.6.0 SVN from a few days ago and my Gosubs have stopped working. This is from the verbose logs: -- Executing [03333407271 at incoming-aaisp:4] GotoIf("IAX2/aaisp-3802", "1?5:7") in new stack -- Goto (incoming-aaisp,03333407271,5) -- Executing [03333407271 at incoming-aaisp:5] Gosub("IAX2/aaisp-3802",

I just downloaded and installed asterisk-1.2.13 Reading http://www.voip-info.org/wiki/view/Asterisk+AEL2 says I should be using AEL2, but what's downloaded is pbx_ael not pbx_ael2 as expected. Where and how do I get current release of AEL2 - Is there some 'How To' somewhere? TIA Bart

Being rather new to Asterisk I was wondering what the current status of AEL2 is? I see reference to it back in January but that was many versions ago. Is it in the current code? Doug **************************** * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307 * * * * doug@crompton.com * * http://www.crompton.com * ****************************

I know I compiled AEL2 into 1.2 before, considering I just copied my source from one server to another, yet I can't seem to figure out why I'm getting this error. Anyone have any ideas? make[1]: Entering directory `/usr/local/src/asterisk-svn/asterisk/pbx' flex argdesc.l "argdesc.l", line 19: unrecognized %option: reentrant "argdesc.l", line 20: unrecognized

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

Hi, Is there something available to add AEL2 syntax highlighting support to Kate ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090710/10576943/attachment.htm

Hello, I've started to play with PJSIP and got stuck at the following problem. I need to retrieve SIP Call-ID associated with PJSIP channel. For inbound channel I can use ${PJSIP_HEADER(read,Call-ID)}, but that doesn't work for outbound channel even in pre-dial or hangup handler. Whatever I do PJSIP_HEADER seem to be unable to read headers for outbound channel. Here's what I do:

I am originating a command via the AMI with this... Action: Login Username: xxx Secret: yyy ACTION: Originate Async: yes Timeout: 60000 Exten: callback Channel: Local/6505551212 at LegA Callerid: 849120 Context: default ActionID: 849120 My LegA context: ----------------------- context LegA { _X. => { Dial(SIP/${EXTEN}@Provider); } } And my default context:

>>> Nobody has any information or opinions on any of this? Personally, I don't think MACROS are going anywhere any time soon, so I have not bothered looking into a substitution. As for ael; I've never used it. Doug

Hi, I've new to http://www.voip-info.org/wiki/view/Asterisk+AEL2 I'm using NoOp and Verbose functions inside extensions.ael. Strangely, NoOp is not printing anything in Asterisk console while Verbose is working. Am I missing something obvious ? Cheers -------------- next part -------------- An HTML attachment was scrubbed... URL: