similar to: OT - How to indent AEL file

Hi, I'm using asterisk 1.6.1 and AEL2. I'm trying to find the best way to write my own custom functions ? At the moment, I'm using this pattern (extensions.ael) : context foo { 123 => { &myfunc(123456); NoOp(${GOSUB_RETVAL}); }; macro myfunc (arg) { Return (${arg}); } 1. First, I keep getting warnings like Warning: file /etc/asterisk/extensions.ael, line

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:

Hello, With an extensions.ael enabled system, I keep getting whatever I change into my "astup.call" file : [Mar 12 00:13:56] WARNING[2538]: pbx_spool.c:267 apply_outgoing: At least one of app or extension (or keyword message/pdu) must be specified, along with tech and dest in file /var/spool/asterisk/outgoing/astup.call [Mar 12 00:13:56] WARNING[2538]: pbx_spool.c:457 scan_service:

Hi, When I was testing an IVR, I realized I miss a function I would call GotoIfTimeWithOffset. Today, this IVR is using function AEL GotoIfTime in several places. The problem is if it's 11pm at the moment I'm testing this IVR, I can't nicely test the 9am or 2pm branch. GotoIfTimeWithOffset would get 2 incoming arguments : - the first is a time range (just like GotoIfTime), - the

Hi, Has anyone met any success, installing localized (ie non-english) menus within SIP firmware enabled Cisco 7941 ? Those phones seem to be trying to download localized menus from Cisco Call Manager but as they are managed by an Asterisk server, I'm looking for a workaround. Any advice ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I succesfully install spandsp chan_misdn and digium card. the rxfax works fine and I get the fax result by email. I would like to do the same using a Patton gw + zaptel but I can't receive fax anymore, the call comes in from ISDN in the Patton gw, patton sends it to asterisk, asterisk run a macro to make a tif file using rxfax, the tif file is correctly created but with a 0 size the call

Hello, Beside Cisco 79x1-GE, I'm not aware of any Gigabit SIP Phone. Did I miss something ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070612/b9b701b3/attachment.htm

Hi, Is anyone aware of a SIP hardphone with Busy Lamp Fields supporting 2 colors (or more) ? This could be very useful to support extended presence, for instance. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090519/0b8f1b62/attachment.htm

Hello, How does extensions.lua compares to extensions.conf or extensions.ael on stability, performance and features ? Would you recommand extensions.lua as an easy/easier way to access memcached, redis or equivalent ? Thoughs ? Comments ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have had a search for this, but didn't come up with any results, so maybe I am using the wrong terms, sorry if this is an FAQ. For those who want to forward their incoming voice calls to a mobile, it could be a cheaper option to call a mobile from another mobile on the same network. This probably wouldn't be useful for users in USA, Canada or Hong Kong as costs to call a mobile is

I'm trying to clarify my understand of gosub, macros and AEL. My understanding is that macros using the Macro() application, which is defined in extensions.conf by: [macro-foo] ... and called in extensions.conf with exten => _9NXXNXXXXXX.,n,Macro(fastbusy) is deprecated in favour of Gosub(). True so far? But then there are "macro"s defined in extensions.ael: macro foo() {

2009/12/3 Olivier <oza-4h07 at myamail.com> > Hello, > > How can you parse a comma separated list using function CUT and AEL ? > > I've tried but it displays error message (though is seems to find the > correct value) : > > STRING=101,102 > VAL=${CUT(STRING,\,,1)}; > NoOp(VAL is ${VAL}); > > Cheers > Sorry for the noise but I mixed up with another

"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

Almost two years ago, a change between how AEL code is built into Asterisk dialplan between minor versions made clear the need to provide a sane entry point into AEL subroutines and that's how AELSub() born. With Asterisk 11 release, they way [stdexten] at extensions.conf is invoked changed from Macro to Gosub using the 'missing context feature' and this caused that any stdexten

Hi, As many remember, almost one year this Skype for Asterisk extension program was announced. Has anyone tried it ? Is there any available pricelist ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090627/37b93684/attachment.htm

Hi, Let's say you have an Asterisk server running. Which parameters would you check to improve service continuity ? I was thinking of : - telco lines status (make sure every is up) - registered hardphones - config files backup (compare live and saved configuration files, if files differ, notifies the administration team) - systems variables (disk and CPU) - log files (trigger an alarm for

Hi, I feel I've read a thread about this previously but I couldn't find it. Is there way for an Asterisk server to check if a sip phone is forwarded without bothering phone's user ? I was thinking of some Alert-Info option that would let the phone reply with a 302 Moved Temporarily or 182 Queued message and not let the phone ring or display anything on its screen. So that, you could

Hello, LLDP is more and more available on various network elements (endpoint, switches, ...). It seems to ease network configuration. Do you have any experience with it ? How would you rate LLDP ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091124/fce6307c/attachment.htm

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.

Hi, I need to replace digital handsets in offices where there cabling is appareantly not Ethernet-compliant. Today's usage is to press a key to toggle between private ou public line before issuing an outgoing call. Are you aware of a DECT handset (to overcome cabling limitations) that mimic this line-key behaviour ? For instance, acceptable behaviours would be to dial number string and press