similar to: calls don't hang up correctly on VM

This is really doing my head in! For some reason, my asterisk box can't playback beep.wav. I have this extension defined in my internal context: '10001' => 1. Answer() [pbx_config] 2. Wait(2) [pbx_config] 3. Record(/tmp/asterisk/10001:gsm) [pbx_config]

I have a dial plan that includes a company phone directory as a main menu option. If they just sit at the main menu, after 20 seconds, they are transferred to the operator. If the user picks an extension from the directory, they are transferred to the proper extension. If the called number is not available, they are transferred into VoiceMailMain. They leave a message, and hang up. The hang

I've RTFM and Googled but can't seem to get sip_autoreg to work (or perhaps I'm just completely missing the point of it). (what I'd like to do is avoid having to put explicit entries for every SIP phone into extensions.conf). Asterisk is creating entries in the (virtual) context sip_autoreg: asterisk*CLI> dialplan show sip_autoreg [ Context 'sip_autoreg'

I have this in my dialplan... [general] static=yes writeprotect=no clearglobalvars=no [start] exten => 5000,1,Answer exten => 5000,n,Wait(1) exten => 5000,n,NoOp(${CALLERID(num)}) exten => 5000,n,Playback(tt-monkeys) which, when I dial 5000, executes this... == Parsing '/etc/asterisk/sip_notify.conf': Found -- Executing [5000 at start:1]

Hi everyone, I'm having an odd issue. I've been doing some testing over the past couple weeks on some Asterisk modules / utilities, but have bumped into a problem which I can't seem to resolve. Asterisk can't seem to play the default sound files (ULAW) in my environment. All necessary debugging information is included below. I'd love to get anyone else's thoughts on this,

Hello. I have been beating my head over this problem for about 6 hours now. I have a SIP peer, who I register to (successfully), who should be directing all incoming calls at my [default] stanza in my extensions.conf: [ Context 'default' created by 'pbx_config' ] 's' => 1. Wait(1) [pbx_config] 2.

I have this in my dialplan... [general] static=yes writeprotect=no clearglobalvars=no [start] exten => 5000,1,Answer exten => 5000,n,Wait(1) exten => 5000,n,NoOp(${CALLERID(num)}) exten => 5000,n,Playback(tt-monkeys) which, when I dial 5000, executes this... == Parsing '/etc/asterisk/sip_notify.conf': Found -- Executing [5000 at start:1]

2013/3/29 Julian Lyndon-Smith <asterisk at dotr.com> > check out the endbeforehexten option in cdr.conf > > this needs to set to "yes" > > Julian > Unfortunately, this doesn't help. Let's drop the hangup handler at the moment, and focus on the "saving to file" part. Then my issue is I can't update CDR value is hangup exten. Here is a

Hi, Here is my confing: [out] Exten => _X.,1,Noop() Exten => _X.,2,Dial(SIP/${EXTEN}@peer,60,gcU(do_dtmf_cc-take-call,s,1)) Exten => _X.,3,Playback(tt-monkeys) Exten => _X.,4,Playback(tt-monkeys) Exten => _X.,5,Playback(tt-monkeys) Exten => h,1,Noop(ABCDEFGHIJKLMNOPQRSTUVWXYZ) [do_dtmf_cc-take-call] Exten => s,1,AGI(agi://127.0.0.1:4579/update_call_status?status=60) Exten

Hello, I'm using Asterisk 1.6.X version and I'm creating IVR. My question : is it possible create CDR record , before client is exiting from contexts ? My test dialplan is: context Sales { _X. => { Ringing(); Wait(4); Answer(); Playback(tt-monkeys); goto Techs|${EXTEN}|1; } } context Techs { _X. => { Playback(tt-monkeys); Playback(tt-monkeys); goto

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 =>

-- jand. more than just a group Asterisk AGI Command GET DATA is usually of this form GET DATA timeout max_digits When I execute this command, I get only a single digit, regardless of what the value of max_digits is, Also the script quits Immediately after the press of the digit regardless of what the value of timeout is, This is really un-desirable as I will like to GET multiple DTMF digits

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

Greetings, Running CVS HEAD about 3 weeks old, I have been beating my head trying to get this to work properly.. Or at least figure out what's going on. Maybe I have done things wrong... I have created a 'catch all' extension at the end of our last context where all phones & voicemail extension exist. This catch all is included in all and works quite nicely except when voicemail

handled. So....what do I do? Thanks, MD =1=================================================== !! Invalid Protocol Profile field 0x11 -- Accepting call from '2004000' to '111' on channel 0/23, span 1 -- Executing NoOp("Zap/23-1", "Incoming call from Meridian1") in new stack -- Executing NoOp("Zap/23-1", " From number: 2004000|

picture. I know the firmware on the Nortel is old, so I'm guessing that libpri is sending something that the Nortel does not know how to handle. Is there a way to dumb down what libpri sends? From everything I've read PRI is an evolving standard - and older devices may struggle with newer extensions/developments. (This might be very handy for users trying to talk to old pbx's.) Is

Greetings all; I have an odd problem - Hangup isn't hanging up, instead Asterisk carries the flow going in the extensions.conf, and the next matching extension gets run. Not good. My extensions.conf (highly simplified) looks like this: [pri] include => dids include => SIPlookup [dids] exten => 13015555555,1,Wait,3 exten => 13015555555,2,Answer exten =>

I'm trying to actually use the example application map in features.conf: testfeature => #9,peer,Playback,tt-monkeys ;Allow both the caller and callee to play ;tt-monkeys to the opposite channel I see the feature get registered at the CLI: == Registered Feature 'monkey' == Mapping Feature 'monkey' to app

On 1.6.0.16-rc1: I'm using app_fax.so to send a fax, and then send a confirm. 'send' => 1. Set(UniqueFile=/var/spool/asterisk/outgoing/call-${UNIQUEID}) [pbx_config] 2. System(env echo -e "Channel:DAHDI/g0/........\\nContext:fax-tx\\nExtension: s\\nPriority: 1\\n" >${UniqueFile}) [pbx_config] [ Context 'fax-tx' created by

So, I'm trying to get the Asterisk vxml (from i6net) working. Having no luck with it. My dial plan has: exten => _X.,1,Answer() exten => _X.,n,Wait(1) exten => _X.,n,Vxml(file:///tmp/menu.vxml) The /tmp/menu.vxml file has: <?xml version="1.0"?> <vxml version="1.0"> <form> <block><audio