similar to: transfer function

Hello, All! How to specify the context in call file section Channel? Is it possible? I want to dial external number (12345) and connect it to context "notify", which consist of playback() command: Channel: SIP/12345 Callerid: auto <12345> MaxRetries: 3 RetryTime: 40 WaitTime: 50 Context: notify Extension: 1 Priority: 1 extensions.ael follows: context notify { 1

I saw a few posts of this problem before I could not figure out the reason I am getting it. I am running RHEL 5, Asterisk 1.4.21.2, zaptel 1.4.11 and libpri 1.4.4 Basically, if I dial into a queue and hang up the phone, Asterisk did not detect the hangup request and Asterisk will only hang up when the timer expires. There is no such problem if I do not use Queue(). Any thoughts? Here is my

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

Hi all. Why don't Asterisk support inband DTMF with G729? Is there a way to do that!? Are you using RFC2833? Doesn't it a security hole? Thanks. Denis.

Dear Users ! I have recently installed asterisk 1.4 i got a warning message whenever i use reload or extensions reload. [Jun 29 19:22:11] WARNING[4539]: pbx.c:6236 ast_context_verify_includes: Context 'ael-local' tries includes nonexistent context 'ael-parkedcalls' [Jun 29 19:22:11] WARNING[4539]: pbx.c:6236 ast_context_verify_includes: Context 'ael-dundi-e164-local'

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() {

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

Does anybody know why Asterisk does not support inband DTMF for G.729? Our SIP carrier use inband dtmf for G.729. This causes problem for us to use it for our Asterisk IVR system. Any suggestion to solve this problem? Gary -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070622/43308a1f/attachment.htm

Hi, I can't figure out what syntax to use with the Dial() "M" parameter for the AEL parser to interpret properly. Creating an AEL macro named "macro-screen()" partly works as a hack, but it must not turn into a gosub properly, so I get warnings about the "return;". Dial(...,tgM(&screen)) with the ael macro named "screen" does not work

Hi list! ConfBridge dtmf_passthrough=no doesn't seem to have any effect. DTMF gets transmitted throughout the conference. I've tried Asterisk 10.7.1 from the official RPMs and 10.8.0 compiled from source. I've confirmed that it's disabled via the CLI "confbridge show profile user <profilename>". It's an all-SIP scenario with RFC2833 as the DTMF protocol.

Hi I'm using a macro to dial in a AEL dialplan. The problem is the macro do not set the field CDR(dst), showing only ~~s~~. I tried various configurations, but without solutions. This is the macro: macro dial-out(destno,dialstring,route_descr,interno) { __TRANSFER_CONTEXT=ipbx; if(${interno} = 1) { Set(__PICKUPMARK=${destno}); if(${ODBC_verify_user(${CALLERID(num)})} > 0) { t = tT; }

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

The solution to the problems with the Grandstream 1.0.4.39 firmware is to use inband (in-audio) DTMF. Neither the RFC2833 nor INFO seem to work. However, this presents another problem. When I'm using g729 to place a call, I get the warning "Unable to process inband DTMF" because inband is not supposed to work with g729 (although it does seem to work when I've tried it so far).

"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

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 just wondering why DTMF are not recognized by aterisk on incoming calls from my SIP provider ... ANy suggesteions ?` /Mike

Hi can anyone help me on this error msg?? dsp.c:1467 ast_dsp_process: Unable to process inband DTMF on 2 frames thnx St

Hi, After seeing in "pbx/ael/ael-test/ael-test5/extensions.ael" some interesting use case of RealTime to store extension data (forwardto, dnd, ...) i started to play with it. To my surprise the two applications RealTime() / RealtimeUpdate() have been deprecated in favor of the REALTIME() function. While it's not a big deal for the write part since it's only a matter a shifting

Incoming FWD calls from other FWD users, iaxtel, or via ipkall, need to have dtmfmode=rfc2833. However, incoming FWD calls from the dialup access numbers (such as libretel) need to have dtmfmode=inband. To solve this problem, I created a second FWD account and configured sip.conf as follows, in order to match the incoming number to the proper dtmfmode: [fwd-rfc] type=friend secret=*****

Hi all ! We currently have an asterisk box that is rather old (runs Asterisk 1.4.21.2), and it's connected to the PSTN with a sangoma A104d card. Now we have a new PRI at another location, and I use that occasion to build 2 new servers, one to replace our aging one and a new one for this new pri. So I downloaded the lastest libpri / asterisk / wanpipe driver, but the previous version of