similar to: Commit dialplan & other config. in memory to disk?

with the slip of a finger, i destroyed by extensions.conf (grep -i > extensions.conf) I have a backup that is dozens of hours of code old. is there a way i can use the asterisk cli (or some other asterisky method) to recreate that extensions.conf ?

On 4/13/16 11:57 AM, A J Stiles wrote: > You could try > *CLI> dialplan show Between my older backup and dialplan show, I guess that's my best shot. Thanks :D

I've got an agi that recognises speech (via Google) and another that turns text into speech (tts) (via Microsoft Translate). Both are web APIs, both called via seperate python AGIs. I've googled and I'm probably missing something pretty newbie 101 here, but is there any way, or fiddle, that I can play some audio to let the caller know that their weather forecast is being fetched,

hi all, can anyone give me a guide on any asterisk admin solution / interface for config management, and monitoring? No database use is intended and I prefer open source. Thanks for support. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160516/98f6e448/attachment.html>

Hello, Reading through the Wiki: "Asterisk supports the ability to write dialplan instructions in the Lua programming language. This method can be used as an alternative to or in combination with extensions.conf and/or AEL. PBX lua allows users to use the full power of lua to develop telephony applications using Asterisk" My question is, what is the benefit of using Lua? I recently

Hi, I am wondering, if there is any tool available, which performs a check for suspicious entries in the dialplan. For example a non existing AGI-Script or a double assigned extension ike that: [context] exten => *100*,1,AGI(test_app.pl) ... exten => 190,1,Answer() ... exten => *100*,1,AGI(never_reached.pl) ... A "normal dialplan reload command" would echo no warning or

On Thu, 31 Aug 2017, Joseph Smith wrote: > So I am looking for a better way to allow several thousand callers to > listen to this IVR menu at the same time. I'm thinking multiple hosts. I'm not a fan of 4,000 eggs in one basket. -- Thanks in advance, ------------------------------------------------------------------------- Steve Edwards sedwards at sedwards.com

On Thursday 31 August 2017 at 18:15:54, Joseph Smith wrote: > I was hoping Asterisk would handle more than 4k simultaneous calls. I know from experience that Asterisk can handle more than 4k simultaneous calls, however it's an extreme case to have all of them playing music on hold. I think that if you tested 4k simultaneous calls with standard media streams on the majority of them, you

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 --------------

sorry for top-posting, the two topics started with 2 different reason subject, but then we finished on the same problem. btw,the 2 show channel are reported above: the channel with DTMF working kcenter*CLI> core show channel SIP/pbx2-000004b9 -- General -- Name: SIP/pbx2-000004b9 Type: SIP UniqueID: 1467323106.1275 Caller ID: xxxx Caller ID Name: xxxx

On Wed, 22 Nov 2017, Dmitriy Serov wrote: > ?same => n,System(printf "Action: Originate\nActionID: 1\nChannel: Local/${number}@mycontext\nApplication: Confbridge\nData: ${confnum}\n" > > /var/spool/asterisk/outgoing/${number}-${confnum}) I get: Unknown keyword 'Action' at line 1 of /var/spool/asterisk/outgoing/... Unknown keyword 'ActionID' at line 2 of

Hi Gang I have stumbled of this problem. I need the P-Asserted-Identity header in an AGI scrip. In the Dial-Plan I do: same => n,Set(PAI=${PJSIP_HEADER(read,P-Asserted-Identity)}) In the AGI I do: my $pai = $AGI->get_variable(PAI); This works fine, unless the PAI contains quotes: P-Asserted-Identity: <sip:1000 at 1.2.3.4:5060;user=phone> I get "<sip:1000 at

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 ? >>

I want to capture all SIP messages. I have about 30 hosts in about 6 colos. My first thought was dumpcap, but the output file name format bugs me. What do you use for long term SIP capture? -- Thanks in advance, ------------------------------------------------------------------------- Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST

Hi Thufir, The analysis of a SIP Debug depends on what the problem to be solved. If you experience problems with inbound calls from a SIP trunk or provider, you can type in Asterisk cli 'core set debug 3' and then 'sip set debug ip xxx.xxx.xxx.xxx' where xxx is the IP of your SIP provider or from where it is supposed to come call. Then you make a test call, and look in full log

On Tue, 26 Dec 2017, Eric Wieling wrote: > Don't use an 'h' extension, use a hangup handler.?? Why? -- Thanks in advance, ------------------------------------------------------------------------- Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281

I have an AGI that can sometimes take time complete. I don't want the dialplan to be held up by the agi. Is there any way to call it and have Asterisk continue with the dialplan? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20190410/4c704231/attachment.html>

On Fri, 18 Mar 2016, Trey Hilyard wrote: > I thought this would be as easy as > exten => _XXXXXXXXXX\;rn=+19136630000,1,Goto(from_pstn,${EXTEN:0:10}) Have you tried the '_!.' pattern? -- Thanks in advance, ------------------------------------------------------------------------- Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST

Hi Steve, What language is that please? We're using Perl and so far I haven't found an equivalent there. Thanks for your help. On Fri, 7 Jun 2019 at 12:10, Steve Edwards <asterisk.org at sedwards.com> wrote: > On Fri, 7 Jun 2019, David Cunningham wrote: > > > We have a need to record audio and allow the user to press any DTMF key > > to end the recording.

Hi list, n00b question, but I can't figure it out: [callthrough] exten => _+X.,1,NoOp(nothing here) #include "blockedall.conf" exten => _+X.,n(hangup),Hangup exten => _+X.,n(nohangup),GotoIf($["${CALLERID(num)}" = "anonymous"]?nocli:cli) ... more stuff that is handling the call ... I'm putting CLIs that I don't want to be able to call my