Re: [asterisk-users] speech-recog.agi

That's because it uses a deprecated API and endpoint. However, funny you should ask this, because I've just finished updating my Google TTS routine to take advantage of the new streamlined API. If you can wait a couple of days, I've stick it up on the repo - BUT... it's going to require

Re: [asterisk-users] Simple speech recognition for driving IVR - "press or say one".

Please check code of it. It listens for # and it is quite easy to add all > other keys 1-9 and etc > > Then change code accordingly so script returns value of key. > > As far as I remember it wasn't hard. > > With kind regards, > > > Jurijs > > On Wed, Dec 6, 2017 at

Re: [asterisk-users] Simple speech recognition for driving IVR - "press or say one".

ting and recompiling a core application or doing any complex workaround? Thanks On 6 December 2017 at 23:25, Jonathan H <lardconce...@gmail.com> wrote: > Thanks for your responses - it looks like I have the following > options, in order of ease: > > 1: Modify and recompile ap

Re: [asterisk-users] Simple speech recognition for driving IVR - "press or say one".

Thanks Jurijs, Yes, in fact I'm already using that, and it works fine. The problem here is that I cannot find a way of recording speech AND listening for a DTMF digit being pressed as an alternative. That's where the problem lies. J. --

[asterisk-users] Simple speech recognition for driving IVR - "press or say one".

Briefly: I want to be able to have "press or say (number)", with Asterisk listening for a spoken number, but accepting a DTMF digit, too. I'm posting everything I found so far, here, partly to show working, but also in case anyone else finds it useful. So, moving on This looked hopeful for a

Re: [asterisk-users] ASR Suggestions for small dictionnary (<1000 entries) lookup in France/french

Having experimented with something similar myself, I'd say you are about to create a vast amount of complexity by moving away from keypad entry. Also, a lot of the natural language APIs don't support French - for example, Amazon Lex or https://dialogflow.com would be great for this as they

Re: [asterisk-users] speech-recog.agi

<cur...@telecomab.mx> wrote: > On 10/19/17 3:53 PM, Jonathan H wrote: > >> That's because it uses a deprecated API and endpoint. >> >> However, funny you should ask this, because I've just finished >> updating my Google TTS routine to take advantage of the new >&

[asterisk-users] Handling a long-running agi on hangup-handler?

I know that hangup handlers ( https://wiki.asterisk.org/wiki/display/AST/Hangup+Handlers) have to finish quickly. So it's no surprise that my speech to text agi which takes 8 seconds gets killed. However, can anyone think of a way round this? So, once the caller has hung up, I need to take one

Re: [asterisk-users] Can anyone help with a quick app_record.c module improvement and can explain over-riding modules?

ep using valid decoded DTMF data > Catch-all, should never get here. > > /Pseudo-code > > > Don't forget to filter your user sourced data against your white-list, > always assume users are hostile, this is part of the total picture of > defence-in-depth. > > -Tim &

Re: [asterisk-users] Can anyone help with a quick app_record.c module improvement and can explain over-riding modules?

On 20 January 2018 at 23:30, Tim S wrote: > I have seen this take over 2 seconds before on a sluggish machine. Thanks - my host uses SSD and everything seems pretty quick, but I'll give it a 1 second pause. > you'd need to pipe that to a Google Speech API tunnel. >

[asterisk-users] Installation instructions for Opus are incorrect - maybe?

Before I got an log a ticket, can I just check I'm not doing anything wrong? In 15.2, to install Opus: 1) run `make menuselect` 2) Highlight "Codec Translators" and press enter. 3) Scroll down to "codec_opus" in the section labeled "External" 4) Press enter to select the codec if it is not

Re: [asterisk-users] Installation instructions for Opus are incorrect - maybe?

However, while I managed to apt install xmlstarlet, I already had bash, and I can't find res_format_attr_opus. Any ideas? Many thanks, Jonathan > Regards. > > -- > Ludovic Gasc (GMLudo) > > 2018-01-27 2:06 GMT+01:00 Jonathan H <lardconce...@gmail.com>: >> >> Be

[asterisk-users] "Cannot write OGG/Opus streams. Sorry" - any ideas?

So as y'all know, with your help I managed to get Opus installed at last. Yay! With excitement, I wrote my dialplan, dialled in, and [Jan 28 21:30:11] ERROR[29977][C-001d]: format_ogg_opus.c:95 ogg_opus_rewrite: Cannot write OGG/Opus streams. Sorry :( [Jan 28 21:30:11]

Re: [asterisk-users] "Cannot write OGG/Opus streams. Sorry" - any ideas?

. Thanks. On 28 Jan 2018 9:47 pm, "Joshua Colp" <jc...@digium.com> wrote: > On Sun, Jan 28, 2018, at 5:34 PM, Jonathan H wrote: > > So as y'all know, with your help I managed to get Opus installed at > last. Yay! > > > > With excitement, I wrote my dialplan

Re: [asterisk-users] Installation instructions for Opus are incorrect - maybe?

On 27 January 2018 at 16:14, Ludovic Gasc wrote: > (M) means it's a module, not it's an error. > Do you relaunch configure after xmlstarlet install ? > For bash error, no idea. Many thanks! Stupid mistake of mine, I hadn't tried when I saw bash was still on (E) (Error, I

Re: [asterisk-users] What does pct mean?

Um, I may be missing something here, but if it was "percent", wouldn't it simply be the internationally recognised symbol for percent, the, um, percent symbol? % That's why I don't think it can be percent. On 13 February 2018 at 18:32, Eric Wieling wrote: > Could this gap

[asterisk-users] ast_iostream_close: SSL_shutdown() failed: Underlying BIO error: Bad file descriptor

Before I file a bug any ideas? Got valid certs, working fine for everything I need them from, but this seems to popup randomly in my logs. [Feb 9 12:43:07] ERROR[14968]: iostream.c:507 ast_iostream_close: SSL_shutdown() failed: error:0005:lib(0):func(0):DH lib, Underlying BIO error: Bad

[asterisk-users] TALK_DETECT - having trouble figuring it out.

Is there a bit more of a detailed explanation of TALK_DETECT anywhere? I googled and found nothing really beyond the wiki at https://wiki.asterisk.org/wiki/display/AST/Asterisk+15+Function_TALK_DETECT I really only want it to listen for one side (the caller) but it seems to listen to both. OK, I

Re: [asterisk-users] SRV with pjsip on Asterisk 15.5: yes or no?

OK, thanks. Shall I file a ticket to get that example file updated? On Sat, 28 Jul 2018 at 21:50, Joshua Colp wrote: > > On Sat, Jul 28, 2018, at 5:42 PM, Jonathan H wrote: > > I'm trying to configure sip2sip, which says: > > http://wiki.sip2sip.info/projects/sip2sip/wiki/

Re: [asterisk-users] Any way of "flattening out" 2 channels back into one?

ds Asterisk Hankup the call > > Regards > > --- > I'm SoCIaL, MayBe > > On 7/28/18 16:08, Jonathan H wrote: > > Last question for today, I promise! > > > > The problem: In order to disconnect calls after x minutes, I need to do > > this: > > > > [se

Re: [asterisk-users] Can someone please help with this sip2sip pjsip_wizard "no matching endpoint" issue?

again. Shall I file a bug? On Sat, 28 Jul 2018 at 21:55, Joshua Colp wrote: > > > > On Sat, Jul 28, 2018, at 3:06 PM, Jonathan H wrote: > > Using pjsip 2.7.2 on Asterisk 15.5 > > Really struggling to make sense of translating these old 1.8 SIP > > instructions into a n

[asterisk-users] Any way of "flattening out" 2 channels back into one?

Last question for today, I promise! The problem: In order to disconnect calls after x minutes, I need to do this: [setup] exten => setup,1,Answer() same => n,Set(LIMIT_PLAYAUDIO_CALLER=yes) same => n,Set(LIMIT_WARNING_FILE=/var/lib/asterisk/sounds/en_GB_TNS/time_limit_reached) same

[asterisk-users] SRV with pjsip on Asterisk 15.5: yes or no?

I'm trying to configure sip2sip, which says: http://wiki.sip2sip.info/projects/sip2sip/wiki/SipDevicesAsterisk "Asterisk, is currently unable to handle more that one result for a DNS SRV lookup, and the Asterisk configuration needed for getting it work with the SIP2SIP service is not trivial" It

[asterisk-users] Can someone please help with this sip2sip pjsip_wizard "no matching endpoint" issue?

Using pjsip 2.7.2 on Asterisk 15.5 Really struggling to make sense of translating these old 1.8 SIP instructions into a neat pjsip_wizard conf suitable for 2018 http://wiki.sip2sip.info/projects/sip2sip/wiki/SipDevicesAsterisk#Version-18 In pjsip_wizard.conf, I have the following, which seems to

[asterisk-users] dialplan reload not showing debug info even with debug on (ast 15.5)

I've not needed to do a dialplan reload for a while, so I don't know exactly which version is stopped working, but on 15.5, I'm not seeing ANY debug info at any debug level. So I'm not really sure how to find mistakes in the dialplan. This is all I get... how do I enable this debug mode to see

Re: [asterisk-users] dialplan reload not showing debug info even with debug on (ast 15.5)

debug is still 4. But it always respects "core set debug" in whichever direction of verbosity is required. Thanks again! On Sun, 29 Jul 2018 at 13:14, Richard Mudgett wrote: > > > > On Sat, Jul 28, 2018 at 1:10 PM, Jonathan H wrote: >> >> I've not needed to do a

[asterisk-users] Struggling to make sense of sending DTMF and why DIAL is trying to make multiple calls?

Hi there; I'm trying to dial into a Zoom conference, send some digits, wait, send a name, and be "in the room", as it were. I thought this would work: same => n,Dial(PJSIP/02036950088@voipfone-205,12,r(callWaiting)D(WWW12345W#WW::)) But it didn't, so I tried all of these: same =>

Re: [asterisk-users] G729

Hmmm, again, this conversation has just faded out. I wondered why no response from Digium? So I found this discussion - https://community.asterisk.org/t/why-does-g729-still-require-licensing/71920/8 - seems very clear that G729 is patent free, but still no response from Digium. Also, the link to

Re: [asterisk-users] Handling a long-running agi on hangup-handler?

atch from the controlling terminal so we don't become a > zombie when we die. > if (posix_setsid() == -1) { > die("could not detach from terminal"); > > } > > > On 01/18/2018 12:27 PM, Jonathan H wrote: > > I know that hangup handlers (ht

[asterisk-users] Can anyone help with a quick app_record.c module improvement and can explain over-riding modules?

Hello, I want to start recording with a prompt of "press or say 1 to 5". If no DMTF is pressed, I want to send the recording to Google Speech to get the number back (got that part working already). If any dtmf key is pressed while Application_Record is running with option y, then the recording

Re: [asterisk-users] Can anyone help with a quick app_record.c module improvement and can explain over-riding modules?

t; n,AGI(agi-ruvoip-net.php,speeddial-voice,${ARG1},${tmp_recor > d_file}-in.wav16) > same => n,NoOp(Voice recognition result: "${agi_result}") > same => n,Gotoif($[ "${agi_result}" != "found" ]?end) > same => n,Return(${agi_call_exten}) >

[asterisk-users] What's the best way of extracting call data which has been written to flat files?

I'm dealing with a blind charity phone information system which writes its logs to two flat csv files (Although the log COULD actually now be written to dynamoDB or sqlite3, too if needed). The first file contains basic call information, one line per call and a unique call ID (distinct from

Re: [asterisk-users] Spontaneous reboot due to MySQL lookups ?

Or better still, skip straight to the current LTS version 16 which was release a few days ago, supported right through until 10-2023! If you're going to do a big upgrade, might as well leap onto the current release :) https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions On Fri, 12 Oct

[asterisk-users] Connecting an existing conference via PJSIP?

Let's say I have a conference room of 8 users. At some point in the evening, we need to hook up with a Zoom conference. That means hooking up that existing pool of users to a new PJSIP channel. An admin would dial in, enter a pin, and initiate that connection. Sounds really simple, but I've

[asterisk-users] After updating to 16 "Some non-required modules failed to load"

I just noticed this upon startup since updating from 15.6.1 to 16.0.0 - do any of these matter? [Oct 18 12:12:18] WARNING[4489]: loader.c:2228 load_modules: Some non-required modules failed to load. [Oct 18 12:12:18] ERROR[4489]: loader.c:2243 load_modules: res_pjsip_transport_websocket declined

Re: [asterisk-users] Struggling to make sense of sending DTMF and why DIAL is trying to make multiple calls?

-to-make-multiple-calls/75556/2 Many thanks On Thu, 2 Aug 2018 at 12:44, Jonathan H wrote: > > Hi there; I'm trying to dial into a Zoom conference, send some digits, > wait, send a name, and be "in the room", as it were. > > I thought this would work: > > same

[asterisk-users] How can I connect an existing Confbridge to a new SIP channel when DIALEDPEERNAME is empty?

Asterisk 16.0, PJSIP For the first caller to a conference, I want to dial out and bridge that conference to a new PJSIP external call. For the next callers, I just want them to join the local Asterisk conference. After the last caller leaves the conference, I want to hangup the call it

Re: [asterisk-users] After updating to 16 "Some non-required modules failed to load"

Thanks Richard - any idea if these matter? And how to stop the errors: cdr_sqlite3_custom declined to load. cel_sqlite3_custom declined to load pbx_ael declined to load Standard 16.0 build, just updated a 15.4; nothing fiddled with in menuselect. On Tue, 23 Oct 2018 at 23:02, Richard Mudgett

[asterisk-users] Is order of channels shown by Function_CHANNELS consistently newest first?

After originating a PJSIP call, I need to get the channel for that call, so I can end it later in a hangup handler. So I use this: https://wiki.asterisk.org/wiki/display/AST/Function_CHANNELS In this bit of dialplan: same => n,Originate(PJSIP/0203123456@voipfone-205

Re: [asterisk-users] How can I connect an existing Confbridge to a new SIP channel when DIALEDPEERNAME is empty?

-an-existing-conference-to-a-new-call/76806/7 Many thanks in advance. On Wed, 24 Oct 2018 at 17:17, Jonathan H wrote: > Asterisk 16.0, PJSIP > > For the first caller to a conference, I want to dial out and bridge that > conference to a new PJSIP external call. > > For the next ca

Re: [asterisk-users] Spontaneous reboot due to MySQL lookups ?

On Thu, 4 Oct 2018 at 20:36, Jonas Kellens wrote: > I stick to 1.8 because it just works. Well, clearly it doesn't because you're posting here! In a few days time, the *8-year-old* Asterisk 1.8 line will be *three years past EOL.* That means End of Life. Do not use. No more support. Now, if

[asterisk-users] Simple one-word offline free speech recognition in Asterisk (or as an AGI)?

When I last looked into this a couple of years ago, simple one-word speech recognition was rather complex and slow. At the moment, I use Google Speech Recognition which uses no local processing power, and is very accurate and fast, allowing me to run on a very low end VPS. However, with the

Re: [asterisk-users] Multiple readfile oddities, newlines etc

_publish_varset: Error creating message -- Executing [s@root:42] Set("Local/s@root-0011;2", "feature=1") in new stack -- Executing [s@root:43] Verbose("Local/s@root-0011;2", "1,feature is 1 unfilteredfeat is ▒=") in new stack feature is 1 unfilteredf

[asterisk-users] Simple, fast single-word offline free speech recognition in Asterisk (or as an AGI)?

e changed :) On Wed, 16 Jan 2019 at 17:42, Jonathan H wrote: > When I last looked into this a couple of years ago, simple one-word speech > recognition was rather complex and slow. > > At the moment, I use Google Speech Recognition which uses no local > processing power, and is

[asterisk-users] AGI: "Get variable" returns variable VALUE vs "Get full variable" returns variable NAME - bug or my misunderstanding?

Just trying out a node agi package (https://github.com/sergey12313/ts-agi/ , and it wasn't behaving as I expected, but when turning on agi debug, it looks like it might be Asterisk (using 17.1.0) This works as expected AGI Rx << SET VARIABLE myVar "Hello World!!!" AGI Tx >> 200 result=1 AGI Rx

Re: [asterisk-users] AGI: "Get variable" returns variable VALUE vs "Get full variable" returns variable NAME - bug or my misunderstanding?

, Sean Bright wrote: > On 12/27/2019 2:24 PM, Jonathan H wrote: > > AGI Rx << SET VARIABLE myVar "Hello > > World!!!" > > AGI Tx >> 200 result=1 > > AGI Rx << GET FULL VARIABLE myVar > > AGI Tx >> 200 result=1 (myVar) > > >

[asterisk-users] Tip/Question about encoding temporary data for storage in Asterisk variable to use in AGI

I wanted to store a JSON object between agi requests for the duration of a call. Turns out asterisk does NOT like a stringified JSON object! AGI complains of "520-Invalid command syntax" So, I just base64 encode/decode it. Assuming I don't need to manipulate the JSON object within Asterisk

Re: [asterisk-users] Better way of streaming radio than "musiconhold" for Asterisk 17.4 ?

w - you make > your own RTP server and provide your own RTP back to asterisk > > On Sun, 3 May 2020, 13:07 Jonathan H, wrote: > >> Way back in 2016 the only way to allow callers to listen in to a stream >> "at will" was to do the following: >> >> moh.

Re: [asterisk-users] Old Asterisk forums not working

olp wrote: > On Mon, Mar 23, 2020 at 9:30 AM Jonathan H wrote: > >> Hope you're all well. >> >> I know we should be using https://community.asterisk.org/ but until >> someone lets Google know that it's moved, all the search results (and >> Asterisk's own search r

Re: [asterisk-users] Old Asterisk forums not working

nathan On Fri, 15 May 2020 at 15:52, Joshua C. Colp wrote: > On Fri, May 15, 2020 at 9:39 AM Jonathan H wrote: > >> Hello! >> >> https://forums.asterisk.org/ is doing it again - "Content Encoding >> Error. An error occurred during a connection to forums.asterisk.

[asterisk-users] Old Asterisk forums not working

Hope you're all well. I know we should be using https://community.asterisk.org/ but until someone lets Google know that it's moved, all the search results (and Asterisk's own search results) come from https://forums.asterisk.org/ In most browsers, it's not displaying; in Firefox, it says:

[asterisk-users] Better way of streaming radio than "musiconhold" for Asterisk 17.4 ?

Way back in 2016 the only way to allow callers to listen in to a stream "at will" was to do the following: moh.conf [radio] mode=custom application=/usr/bin/mplayer https://example.com/stream.mp3 -quiet -ao pcm:file=/dev/stdout -af volume=5,resample=8000,channels=1,format=alaw extensions.conf

[asterisk-users] Any api (agi/ari/ami) equivalent of "core show calls"?

I'm parsing ` sudo asterisk -rx "core show calls" | grep active | head -c 1 ` as an external call from within the Asterisk dialplan then passing it to agi, but this seems really hacky and ugly. However, I cannot find any ARI/AGI/AMI function (or global variable I can get with agi) which shows me

Re: [asterisk-users] Any api (agi/ari/ami) equivalent of "core show calls"?

Thank you... but "just update the database" - hmm, what database? Did you mean ARI? I still can't find the command! The asterisk wiki is somewhat, um... spread around! On Sat, 13 Jun 2020 at 16:56, Steve Edwards wrote: > > On Sat, 13 Jun 2020, Jonathan H wrote: > &

Re: [asterisk-users] Any api (agi/ari/ami) equivalent of "core show calls"?

g me sound ungrateful. I don't mean to be! On Sun, 14 Jun 2020, 22:39 Steve Edwards, wrote: > On Sun, 14 Jun 2020, Jonathan H wrote: > > > Thank you... but "just update the database" - hmm, what database? > > I used MySQL. > > > Did you mean ARI? I

[asterisk-users] serverless fastagi/ARI via AWS lambda and a question about dialplan curl for variables

I want a very basic Asterisk install on a pi to play two mp3 files, which change location, one after another. I have a tiny node script which reads an rss feed and returns the first 2 episodes. In this case, running a fast-agi server seems like overkill and as it's simply 2 or 3 variables, I can

Re: [asterisk-users] Playing MP3's in Asterisk

for music on > hold. I simply had asterisk call ffmpeg to play the files (to get around > all the issues). Where I have local files I convert them over to wav or gsm > and call it a day. > > > On Wed, Dec 23, 2020 at 4:33 AM Jonathan H wrote: > >> Hi all, >> >> R

Re: [asterisk-users] Playing MP3's in Asterisk

Hi all, Returning to the issue of mp3 support in Asterisk, it seems it is using a build from 1997?! http://svn.digium.com/svn/thirdparty/mp3/trunk/layer3.c I have the same problems as everyone with the mp3 add-on, but now a new one: - "mp3/interface.c: Junk at the beginning of frame

Re: [asterisk-users] CURLOPT(useragent) fails with Set requires an '=' to be a valid assignment

Of course! Thank you. I had not thought about escaping it because ";" is not a character I've normally had to escape. Thanks again for this - so obvious now you mention it. On Mon, 14 Dec 2020 at 18:54, Joshua C. Colp wrote: > > On Mon, Dec 14, 2020 at 2:51 PM Jonathan H wrote:

Re: [asterisk-users] CURLOPT(useragent) fails with Set requires an '=' to be a valid assignment

, 14 Dec 2020 at 18:34, Richard Mudgett wrote: > > There are semicolons in the useragent string you are trying to set. If that > is the exact dialplan line then > those semicolons are being seen as a start of a comment. > > Richard > > On Mon, Dec 14, 2020 at 12:25 PM Jonatha

[asterisk-users] CURLOPT(useragent) fails with Set requires an '=' to be a valid assignment

All my other CURLOPT settings like timeout work fine. But this: same => n,Set(CURLOPT(useragent)="Mozilla/5.0 (Windows NT 10.0; Win64; x64) AppleWebKit/537.36 (KHTML, like Gecko) Chrome/88.0.4324.41 Safari/537.36") give the following warning on dialplan reload, with and without quotes around

[asterisk-users] Help needed with ARI RTP externalMedia bridging please

Very simply, I want to pipe some external audio into a channel (bridge) using the externalMedia channel option. Running Asterisk 18 on ubuntu, here's what I did to try and test things out: open a console tab vlc -vvv https://media-ssl.musicradio.com/LBCUK --sout

Re: [asterisk-users] Help needed with ARI RTP externalMedia bridging please

etc! Thanks again. On Mon, 4 Jan 2021 at 17:03, Joshua C. Colp wrote: > On Sun, Jan 3, 2021 at 4:14 PM Jonathan H wrote: > >> Very simply, I want to pipe some external audio into a channel (bridge) >> using the externalMedia channel option. >> Running Asterisk 18 on ubu

Re: [asterisk-users] How do I extract CPLAYBACKOFFSET from ARI playback?

On Mon, 4 Jan 2021 at 10:17, Joshua C. Colp wrote: > On Mon, Jan 4, 2021 at 6:14 AM Jonathan H wrote: > >> Following the playback.js ari-client example, I now need to store the >> current playback offsetms, either when it was skipped or hung up on. >> >> The inform

[asterisk-users] How do I extract CPLAYBACKOFFSET from ARI playback?

Following the playback.js ari-client example, I now need to store the current playback offsetms, either when it was skipped or hung up on. But I can't seem to find it. I know that https://wiki.asterisk.org/wiki/display/AST/Asterisk+18+Application_ControlPlayback sets CPLAYBACKOFFSET but that

Re: [asterisk-users] S3 Bucket support for playing sound files

d it might be if Asterisk were to offer a way of using ControlPlayback etc with an external library? Good luck! -- Forwarded message ----- From: Jonathan H Date: Wed, 23 Dec 2020 at 09:33 Subject: Re: [asterisk-users] Playing MP3's in Asterisk To: Asterisk Users Mailing List - Non-

Re: [asterisk-users] S3 Bucket support for playing sound files

What authentication? I just point to the bucket URL. On Thu, 6 May 2021, 21:28 Dovid Bender, wrote: > Jonathan, > > How do you get around the authentication part? In my case I am using GSM > files so there are no issues there. > > > > On Thu, May 6, 2021 at 4:11 AM J

Re: [asterisk-users] AGI: Why is stream file and wait for digit result ASCII, but get data is "normal"?

On Mon, 24 May 2021 at 18:41, Steve Edwards wrote: > > If you're not using a library, you may want to consider it. 'Comma' is not a valid 'digit' so this the same as '#*0123456789' > I'm using ts-agi which has served me well. In the docs, it suggests phonekeys is an array:

Re: [asterisk-users] HELP! AGI AUTOHANGUP does not seem to hangup the channel.

, 26 May 2021 at 17:22, Jonathan H wrote: > > https://wiki.asterisk.org/wiki/display/AST/Asterisk+18+AGICommand_set+autohangup > > "Cause the channel to automatically hangup at time seconds in the future" > > SET AUTOHANGUP TIME > > Looks great. Except... it doesn't

[asterisk-users] HELP! AGI AUTOHANGUP does not seem to hangup the channel.

https://wiki.asterisk.org/wiki/display/AST/Asterisk+18+AGICommand_set+autohangup "Cause the channel to automatically hangup at time seconds in the future" SET AUTOHANGUP TIME Looks great. Except... it doesn't. It just causes AGI to send "HANGUP" and any audio to stop playing. It does NOT hangup

Re: [asterisk-users] HELP! AGI AUTOHANGUP does not seem to hangup the channel.

8:01, Joshua C. Colp wrote: > > On Wed, May 26, 2021 at 1:58 PM Jonathan H wrote: >> >> I have also tried configuring pjsip wizard like this. >> >> endpoint/rtp_timeout=5 >> >> And I see this shortly after the "hangup" command has be

[asterisk-users] AGI: Why is stream file and wait for digit result ASCII, but get data is "normal"?

Having been scratching my head the whole morning to find a bug, I now have an A4 poster on the wall (not joking!) saying: "get data" = *number* "wait for digit" and "stream file" = *ascii !!!* As you can see here: AGI Rx << STREAM FILE "hello-world" "1,2,3,4,5,6,7,8,9,*,0,#" AGI Tx >> 200

<    1   2