#asterisk — Public Fediverse posts
Live and recent posts from across the Fediverse tagged #asterisk, aggregated by home.social.
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Help Desk с голосовым ассистентом при ограниченном бюджете
Help Desk с голосовым ассистентом. Для руководителей ИТ-подразделений, упор делается на потребность, внедрение и принцип работы.
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🚀 New #release · Asterisk certified-22.8-cert4
The Asterisk Development Team would like to announce
the release of Certified asterisk-22.8-cert4.
The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/certified-22.8-cert4
andFull release notes & setup → https://selfhost.directory/project/asterisk#update-9553901
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🚀 New #release · Asterisk certified-22.8-cert4
The Asterisk Development Team would like to announce
the release of Certified asterisk-22.8-cert4.
The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/certified-22.8-cert4
andFull release notes & setup → https://selfhost.directory/project/asterisk#update-9553901
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https://www.alojapan.com/1520931/asreaders-parent-company-asterisk-inc-celebrates-20th-anniversary-at-headquarters-in-osaka-japan/ AsReader’s parent company Asterisk, Inc. celebrates 20th anniversary at headquarters in Osaka, Japan #AsReader #Asterisk #Inc #LukeSuzukiwalker #news #Osaka #OsakaNews #大阪 #大阪府 North America VP and COO Paul Whitney talks to the crowd at 20th anniversary celebration Asterisk attendees of 20th celebration gather around founder and CEO Noriyuki Suzuki as YouTube personality Luke Suzukiwalker Yoko Sakai, DX Division Director and Executive Officer
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Интеграция Asterisk с веб‑интерфейсом и управление звонками через WebSocket: кейс имплементации под ключ
Привет! Меня зовут Андрей Нелюбин, я руководитель функции фронтенд-разработки. В рамках проекта для заказчика требовалось доработать сценарий работы колл-центра: связать телефонию на Asterisk с операторским интерфейсом, где рядом со звонком доступны данные клиента, заказы, статусы и действия управления. Для этого в 2022 году была реализована собственная прослойка поверх Asterisk: backend на Python/Django, обработка событий через Django Channels, WebSocket для передачи изменений в интерфейс и frontend на Vue. В статье расскажу, как устроили архитектуру решения, как обрабатывали события звонков, зачем понадобился BFF-слой на Django и почему в рабочей конфигурации аудио осталось в SIP-клиенте, а контекст и управление — в операторском интерфейсе
https://habr.com/ru/companies/gri/articles/1061970/
#Asterisk_rest_interface #asterisk #django #django_channels #python #websocket #sip #операторский_интерфейс
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48 #Debian LTS advisories were released in June fixing 231 CVEs for various packages. These include notable security updates for apache2, asterisk, chromium, firefox-esr, havaged, libinput, nginx, thunderbird, tomcat9 and many more.
In June, Debian 12 (#bookworm) was handed over to the LTS team and the relevant tools and documentation were updated. Debian LTS contributors also prepared updates for more recent releases, Debian 13 (#trixie) and Debian unstable.
Read the full report: https://www.freexian.com/blog/debian-lts-report-2026-06/?utm_source=mastodon&utm_medium=social
This work is funded by Freexian's Debian LTS offering. Become a sponsor of Debian LTS (https://www.freexian.com/lts/debian/?utm_source=mastodon&utm_medium=social) and enjoy the benefits (https://www.freexian.com/lts/debian/details/#benefits).
Debian 11 will reach the end of LTS period on August 31 and for continued security support after that, you can subscribe to Freexian's ELTS offer (https://www.freexian.com/lts/extended/?utm_source=mastodon&utm_medium=social).
#debianlts #freexian #asterisk #chromium #nginx #thunderbird
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48 #Debian LTS advisories were released in June fixing 231 CVEs for various packages. These include notable security updates for apache2, asterisk, chromium, firefox-esr, havaged, libinput, nginx, thunderbird, tomcat9 and many more.
In June, Debian 12 (#bookworm) was handed over to the LTS team and the relevant tools and documentation were updated. Debian LTS contributors also prepared updates for more recent releases, Debian 13 (#trixie) and Debian unstable.
Read the full report: https://www.freexian.com/blog/debian-lts-report-2026-06/?utm_source=mastodon&utm_medium=social
This work is funded by Freexian's Debian LTS offering. Become a sponsor of Debian LTS (https://www.freexian.com/lts/debian/?utm_source=mastodon&utm_medium=social) and enjoy the benefits (https://www.freexian.com/lts/debian/details/#benefits).
Debian 11 will reach the end of LTS period on August 31 and for continued security support after that, you can subscribe to Freexian's ELTS offer (https://www.freexian.com/lts/extended/?utm_source=mastodon&utm_medium=social).
#debianlts #freexian #asterisk #chromium #nginx #thunderbird
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I have a Telekom fiber connection and the landline phone is managed by a FreePBX instance behind an OPNSense Firewall.
From time to time, it happens that the landline is not working and I haven't figured out how to fix this.
I'm guessing its connected with changing IP addresses from the provider and/or not yet available internet on server reboots when the FreePBX is trying to establish the trunk connections.
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I have a Telekom fiber connection and the landline phone is managed by a FreePBX instance behind an OPNSense Firewall.
From time to time, it happens that the landline is not working and I haven't figured out how to fix this.
I'm guessing its connected with changing IP addresses from the provider and/or not yet available internet on server reboots when the FreePBX is trying to establish the trunk connections.
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The word you're looking for is "asterisk." It contains two instances of the letter "s." It's the symbol generated when you press Shift + 8 on a standard PC keyboard.
"Asterik" is not a word in English.
THANK YOU FOR YOUR ATTENTION TO THIS MATTER! #Asterisk
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Self-hosted voice AI agent for Asterisk/FreePBX
https://github.com/hkjarral/AVA-AI-Voice-Agent-for-Asterisk
Comments: https://news.ycombinator.com/item?id=48887276
#HackerNews #SelfHosted #AI #Voice #Asterisk #FreePBX #VoiceAgent
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Self-hosted voice AI agent for Asterisk/FreePBX
https://github.com/hkjarral/AVA-AI-Voice-Agent-for-Asterisk
Comments: https://news.ycombinator.com/item?id=48887276
#HackerNews #SelfHosted #AI #Voice #Asterisk #FreePBX #VoiceAgent
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☎️ I Connected a Desktop Phone to a FreeBSD Server, so Now I Can Call It // Alexander Deplov
https://interfacecraft.online/blog/2026/desktop-phone-connected-to-freebsd-server/
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☎️ I Connected a Desktop Phone to a FreeBSD Server, so Now I Can Call It // Alexander Deplov
https://interfacecraft.online/blog/2026/desktop-phone-connected-to-freebsd-server/
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i could get a wonky ubuntu touch based phone - probably not but i may run both iphone and android - another thing on the list #contacts #crm #pbxinaflash #asterisk
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i could get a wonky ubuntu touch based phone - probably not but i may run both iphone and android - another thing on the list #contacts #crm #pbxinaflash #asterisk
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Updated my #Asterisk config to support receiving faxes. Not sure if I’ll ever need it, but I live in Germany, so who knows.
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Updated my #Asterisk config to support receiving faxes. Not sure if I’ll ever need it, but I live in Germany, so who knows.
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Как голосовой ИИ-агент врал клиентам, путал звонящих и подделывал собственный голос — и как это чинится
За три месяца наш голосовой ИИ-агент успел соврать клиенту про несуществующего администратора, принять всех звонящих за одного человека и месяц выдавать обычный синтез за "клонированный голос". Разбираю, почему это лечится структурой кода, а не промптом — на полностью российском стеке.
https://habr.com/ru/articles/1053502/
#голосовой_ИИагент #ИИресепшионист #Asterisk #YandexGPT #Yandex_SpeechKit #function_calling #диалоговый_движок #152ФЗ #голосовой_бо #sipтелефония
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Asterisk's MASTER_CHANNEL() dialplan function is slept on. Lets a channel read variables from its master channel - super useful when you're in a local channel or an AGI subprocess and need context from the call that spawned you, without any hacky variable passing.
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Asterisk's MASTER_CHANNEL() dialplan function is slept on. Lets a channel read variables from its master channel - super useful when you're in a local channel or an AGI subprocess and need context from the call that spawned you, without any hacky variable passing.
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The php8.2-fpm patch now has been pushed out for #IncrediblePBX 2026 platforms. Here's a live (unenhanced) demo (about 30 seconds) of the performance improvement compared to #FreePBX old clunkers running #Asterisk 22 with #Debian 13:
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The php8.2-fpm patch now has been pushed out for #IncrediblePBX 2026 platforms. Here's a live (unenhanced) demo (about 30 seconds) of the performance improvement compared to #FreePBX old clunkers running #Asterisk 22 with #Debian 13:
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Successfully migrated my rotary phones over from a FRITZ!Box 7490 to a Grandstream HT802 and an Asterisk server on my #NixOS server.
Finally I’m able to assign any internal numbers I want and don’t have to go with the Fritzbox’s stupid defaults.
My dial plan now also automatically distinguishes between internal and external numbers. Everything with 3 digits or more is treated as external and routed through Telekom.
What a relief!
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Successfully migrated my rotary phones over from a FRITZ!Box 7490 to a Grandstream HT802 and an Asterisk server on my #NixOS server.
Finally I’m able to assign any internal numbers I want and don’t have to go with the Fritzbox’s stupid defaults.
My dial plan now also automatically distinguishes between internal and external numbers. Everything with 3 digits or more is treated as external and routed through Telekom.
What a relief!
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Dear Fedi,
I would like to have a simple SIP Client, connected to EPVPN, that plays a simple Audio-File.
Other than https://docs.asterisk.org/Getting-Started/Hello-World, I didn't find any good Resources. The Asterisk Documentation just talks about it being an SIP Server.Do any of you know, how to do it?
Any Help would be appreciated#asterisk #eventphone #sip :BoostOK:
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Dear Fedi,
I would like to have a simple SIP Client, connected to EPVPN, that plays a simple Audio-File.
Other than https://docs.asterisk.org/Getting-Started/Hello-World, I didn't find any good Resources. The Asterisk Documentation just talks about it being an SIP Server.Do any of you know, how to do it?
Any Help would be appreciated#asterisk #eventphone #sip :BoostOK:
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Back in 2007/2008, when I was doing my civilian service driving for the Göttinger Tafel, I sometimes had a coworker who would tell me about his #Asterisk setup nonstop, down to the tiniest details.
Now, 19 years later, I’m finally setting up my own Asterisk installation. So far, I have it running in a #NixOS container, and I can call it from a SIP client on my iPhone to get a spoken “Hello World” reply.
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Back in 2007/2008, when I was doing my civilian service driving for the Göttinger Tafel, I sometimes had a coworker who would tell me about his #Asterisk setup nonstop, down to the tiniest details.
Now, 19 years later, I’m finally setting up my own Asterisk installation. So far, I have it running in a #NixOS container, and I can call it from a SIP client on my iPhone to get a spoken “Hello World” reply.
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When setting #FreePBX with special #SIP Alert-Info on inbound route on DDI number, for distinctive ringing on #extensions linked to a ring group, beware that Alert-Info does *not* get transferred to extensions across intercompany #trunks (those with # at end of number)
Workaround to achieve this without custom #Asterisk dialplan (assuming you have outbound route with escape digits with direct route to inter-PBX #trunk)
* Set up ring group on remote #PBX with desired Alert-Info and extensions to be rung
* On ring group on original PBX, add local extensions list and at end route to remote ring group (for instance if ring group 610 and your escape code is 77 for remote PBX put 77610# in ring group extension list)
now distinctive ringing works across both PBX!
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When setting #FreePBX with special #SIP Alert-Info on inbound route on DDI number, for distinctive ringing on #extensions linked to a ring group, beware that Alert-Info does *not* get transferred to extensions across intercompany #trunks (those with # at end of number)
Workaround to achieve this without custom #Asterisk dialplan (assuming you have outbound route with escape digits with direct route to inter-PBX #trunk)
* Set up ring group on remote #PBX with desired Alert-Info and extensions to be rung
* On ring group on original PBX, add local extensions list and at end route to remote ring group (for instance if ring group 610 and your escape code is 77 for remote PBX put 77610# in ring group extension list)
now distinctive ringing works across both PBX!
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New Post: 🚀 How to Deploy Issabel PBX on Rocky Linux VPS #asterisk #fail2ban #issabel #pbx #rockylinux #unifiedcommunications #voip #vps #Cloud #Guides #VPS
🚀 How to Deploy Issabel PBX on... -
I've finished setting up my #voip system at home. I've found all parts in the dumpster, including the four Cisco 7821 phones and a Cisco Catalyst switch with PoE. The phones are now connecting to an #asterisk server on a linux box in the basement. And I've purchased both a German and Canadian dial in number which costs about $1/month. Everything works really well! There were probably close to a hundred phones in the bin at my university. Now I wish I'd taken more of them!
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Welp, figured having some spares would be a good idea so I found an eBay seller selling a lot of 15 Yaelink phones...should be pretty "interesting" to set up and this totally is a good idea and I won't be taking any opinions that don't also think it's awesome...lol
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Letters to Sports: Bam Adebayo’s 83-point game needs asterisk https://www.rawchili.com/nba/658061/ #83PointGame #asterisk #BamAdebayo #Basketball #BillPlaschke #DominantCollegeBasketballTeam #Ducks #heat #Kobe #letter #LosAngelesTimes #Miami #MiamiHeat #MiamiHeat #NBA #NBADraft #RamsFan #SpecialTeam #Sport #TeamUSA #UclaWoman
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Now the other two are ported - all the #analogue #trunks are ceased and all traffic to and from the #PSTN from this #PBX is now #VOIP (using #PJSIP trunks on #FreePBX )
I disconnected all the analogue lines as #Openreach leave battery on them, that way they show as RED alarm on #Asterisk and won't be selected for any calls (worst case is they return a busy trunk status and the route will go to the next trunk, but I've taken away all the analogue circuits from every PSTN route)