I have been developing directly using pjsip c stack For almost 12 years Building IVRs Sip proxies Sip media b2b ua recorders And much more Using this stack.

It would be nice to have someone doing this without the thick heavy accents - We need CLEAR unbroken *ENGLISH* All I hear from this guy is : EEEEEHEHH EEEEEHHHHH. DIFFICULT TO SIT THROUGH THIS, I could only imagine how everyone feels having to listen to this guy at this seminar. *SO DISTRACTING i LOST THE CONTENT VERY ANNOYING*

Excellent

nice except it was 10 ys ago but still fun to watch

I have two pbxact units. How Do I implement this call transcription like what is scene in the browser example you showed?

okay good

Great presentation Thomas although I would point out that Threads was conceived (and much to Meta's annoyance trademarked) 12 years ago, not 17 years ago.

Congratulations 🎉. This demo on github?

Great video

Is good to see Astricon videos again.

Many thanks Sangoma 😊

For those unable to get the files the correct bitly link ends with 3l015c0 (Three Lima Zero One Five Charlie Zero)

Most awaited and demanded feature. I think Live transcription like MS Teams do will be more valuable.

❤

This presentation is gold! Sadly, as a consultant, when you try to tell clients about these things they rarely listen and usually insist on using a “cheap” developer who has “experience with Sipjs/Linphone/etc.”

Awesome!

how can I use Asterisk as SIP server ? any tutorial please ?

can you please guide how we can use monitor() for leg recording in asterisk 20.5.0 version...?

I didn’t know it is pronounced Asterix. I always had that German sk pronounciation before.

“Very mature code for a teenager” was wild😂

useless when u cant see the slides

crappy video, no slides

Thanks a lot!

if dont use proxy External or CNI, RTP so issues. LB only support for Singnal Port.

En el minuto 17:42 entiendo hay un archivo que tiene un texto como "hello-word" y "good bye" el asterisk de forma automática realiza el TTS. Osea se le puede enviar texto para que el asterisk lo convierta a voz.?

Well, here's a little feedback on how well you are doing your job. I am a FreePBX user for only about the last almost 20 years in businesses and running it on a daily basis in my home and after trying to upgrade using the Sangoma distro/FreePBX package I am totally and permanently done with both. First, the crap didn't work. Installation seemed OK but the Asterisk underlying the FreePBX web interface was broken because of a failure of the underlying SQL system with error after error displayed on the CLI. When looking for information on how to troubleshoot/fix the issue there is none. I tried installing Asterisk onto Mint Vera, outside the Sangoma distro which is a sucky version of Centos, and found that is practically impossible now because FreePBX is based on an unsupported and insecure version PHP (7.4). Sangoma's response to users wanting to get help with migrating to version 8 is go ahead and let us know what happens. There appears to be no plan by Sangoma to bring FreePBX up to date. I won't waste my time on crappy unsupported software which seems to me is what I FreePBX has become. I'll use Asterisk from the CLI and say goodbye to FreePBX until it is free from Sangoma. Please, respond and tell me where I am wrong.

Rookie instructor, she may be knowledgeable as hell but, that doesn't make an instructor. I have taught adult education technical products and services for 26 years. Some people don't have it, some do. She was probably chosen due to her knowledge but Digium wasn't paying attention after the fact.

As of 2023 Which Asterisk is best working with Vicidial? and where would I be able to download it?

Don't do anything to me I taught myself

I started asterik 16 on my ununtu in 2020 then i ahift to isabel gui before that i domanal c9nfig that was so fun

Muy buen tutorial muchas gracias, me queda una pregunta: esto tambien se puede hacer con chansip ? o es solo para pjsip? gracias

Awesome presentation.

super that you did it, but I cannot. Looking for solution since 5 days...

how to do balance check in free pbx by giving url

No hay forma de hacerlo en localhost ?

I want to learn how to build the Asterisk switch from scratch.. Please teach me

Are the k8s files available somewhere?

phpari is defunct

phpari is beyond defunct now

IS HANDSET IS O.K.

Hola, muy bueno tu tutorial, tengo una consulta, para la agi, crees que podria utilizarze IBM Watson ? un saludo,y gracias!

very good explanation, thanks

I agree with the previous comment. The audio makes this video unusable. Moving on.

hi

Can we call on local sim phones?

Too bad it didn't go anywhere.

Thanks

I need a couse that wil;l teach me step by step on building PBX

This is great lectuer, I really love it but i am yet to understand if they are developing the call center App with asterisk from scrach or they are setting existing call center app ??

It's about time that I'm on my journey into voip world more specifically Asterisk. This will probably give me a good starting point b4 I get my hands thirty with the actual installation and setup. Thanks alot.