similar to: Queue stats from the dial plan

> Hi Kevin. > > Thank you very much for the hint! It worked very well! > > Your example ' exten => 1234,1,System(echo "This is a test" >> / > var/log/asterisk/test.txt) ' executes when the SIP client (my > softphone Jitsi) sends a SIP INVITE to asterisk. So, the softphone > tries to establish a session with target 1234. > >

> I love this question, simply because it allows me to talk about one > of the neatest features I programmed into my system that barely > anyone knows exists. Plus it lines up pretty much exactly with what > you are trying to do. > > We have our gate control system tied into our Asterisk phone system > so it is possible to dial a code on the phone and open the entrance

When I first to the "add extension" the "show dialplan" has the lines that say "SIP/" but after I do a "save dialplan" and a "stop gracfully" and restart the lines with "SIP/" are gone. ************************ "Show dialplan" before: ************************ asterisk01*CLI> [ Context 'default' created by

I want to have a arbitary named variable within the client's user details in sip.conf [client1] dialplan=NZ .......... In extensions.conf (Logic expressed using PHP style) if ($dialplan == NZ) { $NAT = 0; $INT = 00; }; and in the [outgoing] section ; Australia exten => _${INT}61[278]NXXXXXX.,1,Set(CDR(UserField)=AUSTRALIA) exten =>

Hi, I have a trivial peace of dialplan for exten 100. I try to change it to _1XX and the asterisk act according to a different (Default??) dial plan and not the one I want? Is that possible? Where is the other dialplan sits? In my extention.conf I can't see something that look like what asterisk is dialing. How can I trace\debug my dialplan? Thanks, Eyal -------------- next part

Arch = x86_64 OS = CentOS-6.4 (freepbx) Asterisk = 11.4 FreePBX = 2.11.0.4 I am trying to understand flow control in Asterisk dial plans and not having very much luck. I have read the Asterisk book from O'Rielly, or at least those parts I believe might apply, but that has not helped me much on this particular issue. What I wish is to set three distinct ring tones on our Snom phones for

I think it's called two stage dialing. what I want is the equivalent of dialing 9995551212, waiting a fixed time, say 1 second then dialing 4 more digits, presuming 9995551212 answers. is this possible? If so, how do I code it in the dialplan? Thanks in advance Bruce

Well because I had sooo may problems with chan_agent.c I wrote this. I'm releasing it under LGPL but if you use it or anything please let me know. It'd be interesting if anyone finds this more useful than just a pile of junk. I've included a (working) example extensions file. SIP phones are assumed to have the same identifier as their extension number, but it'd be trivial to

http://www.voip-info.org/wiki/index.php?page=Asterisk+Dialplan+Patterns says: ======== ! wildcard, matches zero or more characters immediately (only Asterisk 1.2 and later, see note) Note: The exclamation mark wildcard, which is available only in Asterisk 1.2 and later, behaves specially - it will match as soon as can without waiting for the dialing to complete, but

I apologize, my English isn't better than the last year So, - why Queue has some options like 'caller can continue in his dialplan' (Dial has g and F options for callee/caller) and not he same option for 'callee' ? - no option in queue to send message (for only calle/caller) when bridged as Dial has - idem to announce periodically 'X minutes before hangup' (in Dial,

> Ok. Thanks for the hint. > > But, what exactly is a "System() dialplan application"? Is it a kind > of command that i can call in dial plan? > > I will look for System() related to dial plans. >From the Asterisk CLI type: core show application System It will print out the syntax for the command. One of the easier dialplan applications. exten =>

Ok. Thanks for the hint. But, what exactly is a "System() dialplan application"? Is it a kind of command that i can call in dial plan? I will look for System() related to dial plans. Thanks. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ________________________________________ De: asterisk-users-bounces at lists.digium.com

Hi Kevin. Thank you again for help me! In my case, in the final application for smartphones or in a softphone for PCs, there will be a button on the GUI and the user will have just to touch it, and the door or gate will open. I mean, during an ongoing call, the callee will see a button in the interface of its SIP application. For example, we can use the lib of Linphone and implement a GUI over

Hi, I have a Linksys PAP2-NA connectd to my asterisk. I would like the device to add 2 characters in front of the dialled number always when it send the call to my asterisk. I dont know how to do that. I will summarise my requirement. My friend dials 1-210-1234345, i want the asterisk to get 55-1-210-1234345. Can someone help me to add this dialplan. Thanks in advance Dan -------------- next

A call comes in and goes into the queue, the queue dials a sip channel using a macro. The macro plays a set of options to the callee and if the callee presses 3 it sets MACRO_RESULT=CONTINUE and the macro ends. For some reason the caller goes back into the queue rather than continueing on in the dial plan. Why is this, i could have sworn in 1.2 if i set MACRO_RESULT=CONTINUE that the

On Thu, 2007-07-26 at 09:37 -0500, Jay Moore wrote: > My boss would like some statistics on how many calls are answered out of > specific queues during a given time period, and I'm not sure how exactly > to obtain those stats. It sounds like you've got quite the queue setup (although I don't quite see why your calls jump out and back into the moh queue). All the of queue

Can I ask for some advice on dial-plan construction please I have setup my dialplan to use 9 to get a zap trunk, leaving everything else for internal extensions. However, this creates a problem in that my callerid is correct, but doesn't work to re-dial the incoming caller. So if I simply click "missed calls" on my Snom phone and hit redial then it tries to dial an internal

Hello, I would like to implement a few decision making process inside the dialplan using information stored in MySQL (like LCR, etc.). I see the MYSQL() application, but as far as I understand I have to connect to the database each time I want to query it; this seems a CPU eater to me. Is this indeed the case, or can I open it once Asterisk starts and leave it open?

The following is my dialplan for outgoing international call. What I want are: - when people dial 9011604xxxxxxx , 9011605xxxxxxx, 90114411xxxxxxx, 90114421xxxxxxx, use voipstunt to dial out - otherwise, use my pstn to dial out. What I've found is when i dial 9011604xxxxxxx , 9011605xxxxxxx, 90114411xxxxxxx, 90114421xxxxxxx, it always use the pstn to dial out. Anything wrong with my dial

> > > > What is a better way to do it then in terms of performance, security, > and > > flexibility? Using exec and a shell script, or agi or something else? > > Setup extconfig to have realtime access to the database/table you want to pull info from, then in the dialplan use the app Realtime. -= Info about application 'RealTime' =- [Synopsis] Realtime