similar to: Handling -1 in dialplans

Helo List, Sorry I missed the rest of my email in my previous post. Please see below. I'm having an issue using the AND (&) operator evaluation in the code of my dialplan. The dial plan is coded to detect inbound DTMF digits from callers. key "1" is equivalent to "yes" and key "2" is equivalent to "no" in two diferent contexts in the dial plan.

Hi, I have problem with Asterisk 1.6.0.1. I need to change language for playing prompts in Lithuanian. But in Asterisk 1.6.0.1 version everytime plays in English, but in Asterisk 1.4.x I haven't any problem. Maybe it is a bug ...? So I paste my test dialpan and prompt's locations. I hope this helps you. Files are: [root at voip asterisk]# find /var/lib/asterisk/sounds/test -name

Hi everybody In advance sorry for my bad english and if my problem was already exposed (I didn't find any tips in the mailing list archive. Bad luck) I have some questions about asterisk 1.6 release : 1) how can I do a n+101 priority jumping if a SIP canal is busy ? I read that the general parameter "priorityjumping" is depreciated in the 1.6 release and I already try the

Hi, I have problems while trying to dial from simple analog phone that attached to my TDM400P card. No matter which number i press i immediately get a congestion tone. when calling from outside (e.g cellphone )to the line on port 4 and pressing extension #123 everything works fine and i manage to make a connection. I've plugged on port(Zap) 4 the analog line and on port 1 the phone.

I need to change the sip port from 5060 to 5061 actually we already used 5060 for proxy to sip any idea to change 5060 to 5061 so all can acces the sip using this port please help........................ On 4/8/11, asterisk-users-request at lists.digium.com <asterisk-users-request at lists.digium.com> wrote: > Send asterisk-users mailing list submissions to > asterisk-users at

Hi, With Asterisk 1.8.3, I can't figure out how to pass an argument to a macro which is used within an originate command. Here is my sample dialplan to illustrate: exten => 123,1,Answer() exten => 123,n,Originate(SIP/20,app,Macro,foo,bar) exten => 123,n,NoOp(This is the NoOp after the originate command) exten => 123,n,Wait(30) exten => 123,n,Hangup() [macro-foo] exten =>

Hi everyone I am having problems with my call queue We currently run a customer care call center which has attendants login during the daytime. Customers who call the 'customer care line (a specific number) always get routed to the cutomer care queue (called 124). After hours, staffs of the Network operating center provide customer care services for customers who call in after the last

The last Originate() option is ignored if using 'app'. It is only there for 'exten'. http://www.voip-info.org/wiki/view/Asterisk+cmd+Originate tells all :) -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Bruce Hopkins Sent: 15 March 2011 21:36 To: asterisk-users at lists.digium.com

Hi, I would have thought that when spawning a channel using the Originate() dialplan command, variables prefixed with two underscores would be preserved. However this does not work in the following case. Dialplan code: [intern] exten => 200,1,Set(__myvar="foo") exten => 200,n,Originate(Local/123 at test_orig,exten,dummy) [test_orig] exten => 123,1,NoOp(${myvar}) exten =>

Hi, I can't get Asterisk to save CDRs for calls transferred via SIP blind transfer. My extensions.conf: [globals] __TRANSFER_CONTEXT = transfer [common] exten => 123,1,Playback(demo-congrats) exten => 123,n,Hangup() exten => _0X.,1,Dial(SIP/${EXTEN}@PSTN-GW,60) exten => _0X.,n,Hangup() exten => i,1,Hangup() exten => h,1,Hangup() exten => t,1,Hangup() [transfer] exten

Hello! I am trying to come up with a way to read a digit *before* the call is answered. My Asterisk version is 1.6.2.0-rc6 SIP early media works fine (I can receive and transmit audio before the call is answered), but as soon as I start the read application, Asterisk answers the call which is not what I want. Here is how to reproduce the problem: send incoming calls from a SIP provider that

I'm Japanese. Sorry,English is not so understood,Please let me question by items. In Asterisk-1.2.1 and 1.2.2,I cannnot understand the operation of Authenticate application's 'j' option. exten => 123,1,Answer() exten => 123,2,Authenticate(789,j) exten => 123,3,Playback(pin-number-accepted) exten => 123,4,SayDigits(111) exten => 123,103,SayDigits(999) In this

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

Hi John, First, I have not much experience dealing with complex dial plans. But since you asked, thought of some feedback. In my opinion .conf files should be kept as simple as possible. It should provide straight forward and simple manipulations to simple & common applications. If more complex manipulations are necessary, then those scenarios could be built using scripts. Therefore, I think

I installed Asterisk 1.0.9 in a Freebsd 5.4 ( with no zaptel card); I have 2 zoom x5v and works great ( in extensions 123 and 321 ) but I was trying to test cmd Playback, MusicOnHold, MP3Player but when I call to extension 100 I don't hear the sound ( mp3 or gsm that I put) , I only hear noise If I leave a message in a mailbox the same, all the record is noise --------- extensionns.conf

I tried putting this as the ata's dailplan: *St4-|#St4-|9|^9t4>$.- this is sip.conf [ata2001] type=friend username=ata2001 secret=SoMeSeCrEt host=dynamic context=fromata canreinvite=no and this in extensions.conf [fromata] ignorepat => 9 exten => _91700NXXXXXX,1,Dial(IAX2/${IAXINFO}@iaxtel.com/${EXTEN:1}@iaxtel) exten =>

This week has been very productive and has shown a huge leap forward in Asterisk development. The creation of the new concepts of an "unstable" branch of the code will, I believe, make for a better development environment in the long run. With that in mind, I'm going to do something I only infrequently do, which is to re-post something in it's entirety and look for comments

Ok I have spent the last week working on getting my small PBX to work. I will in the end only have 4 SIP extensions being either softphones of IP phones. Currently only 1 SIP config for testing. And at the this point it should be all fairly easy with all inbound and outbound to PSTN will be going Via Firefly/Freshtel.net in Australia via IAX. Inbound does work in it's current basic state.

On 2007-05-15 Tony Mountifield wrote (wrt using AGI scripts to dial out): > Can't comment on this one, as I never use AGI to dial. > My AGIs just set the context, extension and priority, > and exit to the dialplan to do any dialling. (http://article.gmane.org/gmane.comp.telephony.pbx.asterisk.user/185537) I would like to do this, but I am having trouble figuring out how. I have

Is it possible to simply send a message to appear as an AMI message/event, from the dialplan? For example exten =>123,1,ami(myEvent, param1, param2) and in the AMI a message appears like: Event: myEvent Privilege: call,all Channel: PJSIP/misspiggy-00000001 Uniqueid: 1368479157.3 ChannelState: 3 ChannelStateDesc: Up CallerIDNum: 657-5309 CallerIDName: Miss Piggy