similar to: IS_REGISTERED from dialplan

Good morning, We recently fell back to the most recent build of asterisk 1.8 down from 11.3 and I believe we've crossed some sort of limit for 1.8. Our dialplan is 515723 entries long with 6263 distinct contexts. Both are loaded realtime via odbc (mysql). Previously at the end of a dialplan reload we would get a summary of how long it took to reload everything. Now it just shows the last line

Hi! I have an T1 line for using with IVR AGI. I receive the calls in my T1 but my dialplan has an error but my extensions doesnt have the error that show me asterisk. I dont know from where asterisk take extension 8 and how is playing ss-noservice because in my dialplan is not exist. Any help or any cluees? Verbosity was 5 and is now 7 -- Starting simple switch on 'Zap/1-1' ==

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all! Now that I have a little more time, I was debugging my dialplan and it was of the following way: - ------------------------------------------------------------------------- ; DGB - 20090615 [macro-dial] exten => s,1,Dial(${ARG1},15) exten => s,n,Goto(s-${DIALSTATUS},1) exten => s-NOANSWER,1,Voicemail(${MACRO_EXTEN}@voicemail,u)

Hello, I'm using asterisk 1.0.7. with a somewhat advanced setup with IAX and CAPI as channels. A call comes in via IAX2 and should be redirected to CAPI. So I wrote the following dialplan: [fromiax] exten => _8XXX,1,Answer exten => _8XXX,2,Dial(CAPI/265:B${EXTEN:1},,r) [fromcapi] exten => 265,1,Answer exten => 265,2,Dial(IAX2/PoC/11@from-lw) exten => 265-BUSY,1,Busy exten

Hello everyone, I'm having some troubles with some dialplan logic I've written which sends missed call notifications via e-mail. It's currently sending these notifications even if the call was answered, marking them all as hung-up. What I've been able to see is that the macro never reaches the "s-ANSWER" bits which mark the call as successful. I've posted my

HI, I tried to configure Asterisk 1.8 on one of my test-hosts. I've installed from centos-asterisk.repo (http://packages.asterisk.org/centos/$releasever/tested/$basearch/): Nov 26 15:34:56 Installed: asterisk-sounds-core-en-gsm-1.4.20-1_centos5.noarch Nov 26 15:34:59 Installed: asterisk18-core-1.8.0-1_centos5.i386 Nov 26 15:35:02 Installed: asterisk18-voicemail-1.8.0-1_centos5.i386 Nov 26

I have a very strange problem. I have two Asterisk servers running 1.4.2. On the first one I have the following options: Connected to Asterisk 1.4.2 currently running on pbxoficina (pid = 7057) Verbosity is at least 3 pbxoficina*CLI> help dialplan dialplan add extension Add new extension into context dialplan add ignorepat Add new ignore pattern dialplan add include Include

Hello, I am running Asterisk 11 on CentOS 6.x. I have configured several queues as follows in extensions.conf: exten => s,1,Queue(myqueue,rtnC,18) same => n,Background(user_unavail) same => n,WaitExten(10) exten => 1,1,Voicemail(1111 at my-vm,s) This rings the phones in the queue for 18 seconds. If no queue members answer, the caller is then prompted to press 1 and leave a

Hello, I have some problems with mISDN. I can't send or receive call from the Billion ISDN card Mi configuration files are thoose: extensions.conf: [general] static=yes writeprotect=yes [SOME] exten => 101,1,Dial(SIP/101,30,Ttm) exten => 101,2,Hangup exten => 102,1,Dial(SIP/102,30,Ttm) exten => 102,2,Hangup include => outgoing [outgoing] exten

hi community, I'm new to this list & asterisk in general, so let me first say thx to everybody involved in providing such great tools & ressources!! I'm currently trying to implement a simple voicebox-system. for demonstration purposes, I've successfully connected my cellphone via bluetooth using the current chan_cellphone-patch on the current SVN-version of asterisk.

Asterisk will try calling both at once. As soon as one is answered it cancels the call to the other. What you can do is for extension 101 to put it in it's own context and then call the agi from the h extension. So something like this: [from-internal] exten = 514316XXXX,1,Answer() same => n,Playback(hello) same => n,Dial(LOCAL/100 at extensions&LOCAL/101 at extensions) [extensions]

Hi, Is there a way to do a blind transfer within an asterisk dialplan (like '##')? The reason I need this (I think) rather than a regular Goto() is that I'm trying to do one-touch parking. I can park a call using one-touch parking and then pick it up again, however if I try to re-park the call, it gets lost. I think that is because asterisk thinks I'm still on the park extension.

I am playing with the dialplan to get it working and I have a challange with this error. I can't find what it means on the wiki :( Any sugestions would be helpful at being able to forward it to the SIP phone if it is online and avaliable but then let that fail and drop into voicemail if it is not online or is busy. cheers David -- Executing Dial("IAX2/firefly@89280250/3",

I was wondering how to get access to the headers of the INVITE on incomming SIP calls in the Dialplan. My scenario is that i use "register" in sip.conf to register a UA on which to accept incomming calls. In sip.conf, calls to that UA is redirected to a specific extension in extensions.conf (btw: can that be a dynamic value, for example ${CALLERID} or

Please don't top post. > Am 29. Mai 2015 09:42:55 MESZ, schrieb Luca Bertoncello > <lucabert at lucabert.de>: >> Zitat von jg <webaccounts173 at jgoettgens.de>: >>> Yes, it is called "core set verbose 42", the other options is "core >>> set debug 42". Enjoy the show! I know you can specify a level to the verbose application,

the caller is out his money anyway when you call any phone and voicemail kicks in, although i think on a payphone they give you a 2 or 3 second window to hang up. Suggest you implement i'm here / i'm away dialplan logic or set the do not disturb button that way when someone calls and the guy is away it hits voicemail right away and the caller can hear this and still have the 2 or 3

Hello list, yesterday I finished work having my whole dialplan available... Today I want to make a call from one local phone to another and I get this : [Aug 28 10:48:57] NOTICE[1895]: chan_sip.c:15144 handle_request_invite: Call from 'test2' to extension '60' rejected because extension not found. Although I have this context : [from-TEST] where all my local extensions are

All, I was a bit too focused on where I thought the problem was - turns out I wasn't crazy and the dialplan does work as expected. The problem was with dtmf detection - setting relaxdtmf=yes did the trick. Sorry for the premature post for help. Begin forwarded message: > From: Ben Witso <benw@bgwcomp.com> > Date: Mon Jun 21, 2004 7:28:42 PM US/Central > To: Asterisk-Users

2015-06-26 17:11 GMT+02:00 Steve Edwards <asterisk.org at sedwards.com>: > On Fri, 26 Jun 2015, Ludovic Gasc wrote: > > 1. What's the "official" notation of each line: "=>" or "=" ? In the wiki >> of Asterisk, I see very often "=>", however, what's the reason for both >> syntaxes authorized ? Historical ? >>

Hello all, I am quite new in asterisk and I am trying to create a dialplan that executes the following steps: 1. A SIP friend dials 102 extension. 2. Asterisk PBX responds with some beeps. 3. The sip friend hangs up the phone. 4. Asterisk PBX calls back to the sip friend after 30 seconds with the application music on hold. I tried to implement this using h extension but I got the following