similar to: Help with a bug

Hey all, I tried to do some searching but I found snippets and I am having trouble putting it all together. I want to have an option off the IVR that plays back the announcement for the day. At the end of the message, I want the caller to get kicked back to the previous menu. The conditions are that I want the recorder to dial a feature code that prompts him to record the message. He

I want to be able to allow a caller to dial a ddi, system to verify identity etc (this is all done) I then want them to sit listening to music, until an event happens. When this (external) event happens, I want to play a certain file to the caller, using playback (so that they have ff / rw etc), and when finished, go back to the music. 1) I thought of redirecting to an extension that played the

?I have the same exact issue. I cannot push any sounds or even Playtones to the caller, unless the channel is answered, which is not possible for billing reasons. I am also using the Local channel & Dial(PJSIP/...). I think this is a bug in Asterisk 13. The Dial function has not answered yet, so the Local channel should be able to play anything to the caller, without answering, in parallel

Hi to All, I need to implement an automatic telephone messaging system that works like this: -the system generates the call based on mysql records or any database -when the client answer the phone, the Asterisk PBX playback a recorded message -when finish, hang up the channel. Only for voice messages not SMS. Exists some application based on Asterisk that makes this, or any code to

Hi. I wanted to write a "popcorn" app for myself, both to learn how to script in extensions.conf, but also because it was something handy. Along the way, I found myself doing something like: [popcorn] exten => s,1,Set(FUTURETIME=$[${EPOCH} + 10]) ... exten => s,n,While(${EPOCH} < ${FUTURETIME}) exten => s,n,Wait(0.01) exten => s,n,EndWhile() exten => s,n,Play(beep)

Do you see any syntax errors? Positive comments welcomed. The short version of the logic is as follows: create a file based on the NUMBER create a an audio file move to a new extension (label) and play the results exten => 621,1,Answer() exten => 621,n,Read(NUMBER,enteryournumberstartingwithaone,12,,5) ; create a variable from a DTMF entry / 12 characters long exten =>

I have been seeing errors saying the Asterisk cannot establish an RTP connection, so I did this: lsof -i -n -P | grep asterisk | wc -l 10483 but I have only Asterisk 11 has 1 open calls Asterisk 12 has 21 open calls Asterisk 14 has 19 open calls Asterisk 15 has 22 open calls Asterisk 16 has 15 open calls Asterisk 17 has 15 open calls Asterisk 30 has 71 open calls Total 164 active calls The

I need to obtain one variable in the dialplan containing the IP address that Asterisk is using, I mean, the originating IP for any calls coming out of Asterisk via SIP. Is this possible? F.Alves

I just tried this in my extensions.conf exten => **,1,Noop(Testing) exten => **,n,Playback(demo-congrats) Did a reload... and the above does not happen. I created as 12 instead of the ** and that works fine. Is there anyway to get the ** to work? I also am using a polycom phone if that affects things. I'm using asterisk 13.15.0 Thanks Jerry -------------- next part --------------

In a few weeks, no SIP call is going to terminate unless they are signed properly, as mandated by law. We are in the business of Stir-Shaken, signing calls, as an FCC-approved provider. A big differentiator between our service and the rest: we are the only ones who don't need to receive the calls in our servers to sign them. We do this over a MySQL call, easily connectable to Asterisk via

I have 12 T1's going into 3 servers, 4 in each into "Digium, Inc. Wildcard TE410P Quad-Span togglable E1/T1/J1 card 3.3v (rev 02)" cards. Each "group" of T1's have the primary D on 24 and the secondary D on 96. The first server (ts20) and the last server (ts22) can playback "demo-congrats" fine. The "middle" server (ts21) cannot -- just dead air.

Hello list members. I would like to know how to playback an audio file to the caller, and while it's played asterisk to continue executing the next priorities on extensions.conf That's not the case when using "playback" command, because the next priority is executed until the audio file ends playing. I want to evaluate some variables while caller hears the audio file. Any

I'm trying to determine the capacity of my host running Asterisk 11.8.1 on CentOS 6.5. The host is an Intel E3-1240v3 with 8GB RAM, an SSD, and gigabit Ethernet. The primary application will be bridging groups of users using meetme(). I'm using 2 boxes -- 1 to initiate calls using call files (box1), and 1 behaving a bit more like a production box -- bridging calls (box2). The call

I am testing an ivr but I'm having problems. The call keeps looping and it doesn't hangup the call after passing three times through the menu. Here's my conf: exten => s,n,NoOp("Here's Count") exten => s,n,NoOp(${COUNT}) ;123,n,Set(COUNT=$[${COUNT} - 1]) exten => s,n,GotoIf($[${COUNT} = 4]?33,1:44,1 ) exten => 1,1,goto(tech-support,s,1) exten =>

I've got a directory under /var/lib/asterisk/sounds which contains a bunch of sound files. I would like to call the Playback command to play the files, but I need it to select a file to play randomly. Is there any way to do this? ~jay

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,

Hi all, new to the list and this is probably a basic question and couldn't find anything clear googling around but I don't know how to handle calls to sip extensions not defined on sip.conf while using pattern matching. On my example I have sip extensions 10, 11, 12, and 13 on sip.conf. On a basic extension.conf I set up a pattern starting with "1" and a second digit should dial

> > There is a big confusion here about Stir Shaken. It is NOT a provider > issue. Un fact, all providers are whasing their hands and modifying their > swihtches to pass-through the Signature. They cannot sign the call because > then the become the responsible party for the call before the FCC, and > liable for any illegal call. Every owner of a PBX that sends calls to the >

I am sure this is simple, but have been struggling. I want to create a call file that dials out a particular Dahdi channel to enable call forwarding on a POTS line. I have this in extensions.conf: [custom-callfwd] exten => s,1,Answer exten => s,n,Dial(DAHDI/4-1/*717157750) exten => s,n,Verbose(${DIALSTATUS}) exten => s,n,Hangup [custom-callfwdcanc] exten => s,1,Answer exten

I want to be able to pass any number (variable length) to a context and then forward that to another asterisk server for processing by that servers dial plan.? I have the two talking IAX2 so that part is done. I can also dial a number from the sending to the server asterisk. The problem is I don't want to have to create (duplicate) dial plans at originating Asterisk to equal those at the