#asterisk — Public Fediverse posts
Live and recent posts from across the Fediverse tagged #asterisk, aggregated by home.social.
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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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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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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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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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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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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)
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Setting up incoming and outgoing calls using #Asterisk #AsteriskPBX #VOIP with #twilio and old #Cisco VOIP phones: http://vikaskumar.org/2026/02/27/cisco-voip-setup-inbound-outbound-calling-asterisk-sip.html
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Just over a month and I've tamed all the #trunks (with abundant snake heads at the end), made sure 1500+ #blighters are yeeted (with more trying every day) got inter #PBX #trunk working between on-premises and cloud #FreePBX - just waiting for porting of first analogue number to check this (and CLID presentation) works and then main office one can follow.
Took many late evenings, a lot of research of everything from old #BritishTelecom training manuals to some from Universities in India and the Indian telecom companies, and I've learned a lot more about #SIP even since 2008 when I built the first #VOIP #PBX used at work.
Thankfully #routers seem to handle #SIP over #NAT a lot better than they used to (even got an extension it working over #LTE with #Linphone)
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Just over a month and I've tamed all the #trunks (with abundant snake heads at the end), made sure 1500+ #blighters are yeeted (with more trying every day) got inter #PBX #trunk working between on-premises and cloud #FreePBX - just waiting for porting of first analogue number to check this (and CLID presentation) works and then main office one can follow.
Took many late evenings, a lot of research of everything from old #BritishTelecom training manuals to some from Universities in India and the Indian telecom companies, and I've learned a lot more about #SIP even since 2008 when I built the first #VOIP #PBX used at work.
Thankfully #routers seem to handle #SIP over #NAT a lot better than they used to (even got an extension it working over #LTE with #Linphone)
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Just over a month and I've tamed all the #trunks (with abundant snake heads at the end), made sure 1500+ #blighters are yeeted (with more trying every day) got inter #PBX #trunk working between on-premises and cloud #FreePBX - just waiting for porting of first analogue number to check this (and CLID presentation) works and then main office one can follow.
Took many late evenings, a lot of research of everything from old #BritishTelecom training manuals to some from Universities in India and the Indian telecom companies, and I've learned a lot more about #SIP even since 2008 when I built the first #VOIP #PBX used at work.
Thankfully #routers seem to handle #SIP over #NAT a lot better than they used to (even got an extension it working over #LTE with #Linphone)
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Just over a month and I've tamed all the #trunks (with abundant snake heads at the end), made sure 1500+ #blighters are yeeted (with more trying every day) got inter #PBX #trunk working between on-premises and cloud #FreePBX - just waiting for porting of first analogue number to check this (and CLID presentation) works and then main office one can follow.
Took many late evenings, a lot of research of everything from old #BritishTelecom training manuals to some from Universities in India and the Indian telecom companies, and I've learned a lot more about #SIP even since 2008 when I built the first #VOIP #PBX used at work.
Thankfully #routers seem to handle #SIP over #NAT a lot better than they used to (even got an extension it working over #LTE with #Linphone)
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Just over a month and I've tamed all the #trunks (with abundant snake heads at the end), made sure 1500+ #blighters are yeeted (with more trying every day) got inter #PBX #trunk working between on-premises and cloud #FreePBX - just waiting for porting of first analogue number to check this (and CLID presentation) works and then main office one can follow.
Took many late evenings, a lot of research of everything from old #BritishTelecom training manuals to some from Universities in India and the Indian telecom companies, and I've learned a lot more about #SIP even since 2008 when I built the first #VOIP #PBX used at work.
Thankfully #routers seem to handle #SIP over #NAT a lot better than they used to (even got an extension it working over #LTE with #Linphone)
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Surprised it took me so long to discover #SNGREP (which has been around for a while), its a game changer compared to looking at the #Asterisk console trying to pick out the #SIP traffic you want from all the other stuff.
It also intercepts packets at kernel level (before any firewall) giving you an insight as to how many #blighters are trying to exploit #VOIP phone systems (especially those where you *have* to open 5060 UDP to connect correctly to SIP trunk providers and/or other PBX or else no inbound calls arrive)
Even produces call flow diagrams which look like the 1980s era training manuals for ISDN from British Telecom I downloaded!
(I clipped off part of the screenshot with my IP addresses at the top)
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Building new #FreePBX #Asterisk #VOIP server (about 10 years since I built the last one) was also an eye-opener of how much #tech world seems to have been deskilled with the rush to #cloud (even before AI) - it seems fewer folk want to build a server from bare metal or even VPS and are flocking to proprietary cloud #PBX (that nearly all run Asterisk under the hood anyway), it might be that #telephony is "uncool" but also remaining engineers have simply stopped helping one another, perhaps not wanting to aid the competition?
I didn't even get much AI slop when searching for info on community forums, as there is so little there and many unanswered threads..
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configuring #VPS that I'd had dormant for a while (was being used for experimental stuff) as proof of concept for #Cloud #VOIP #PBX using #FreePBX and #Asterisk with #Debian12 and the #Sangoma install script.
TIL: to get #PostFix to work correctly with #IONOS without paying for the commercial System Admin module you *must* install
>apt install libsasl2-modules
or else the emails get knocked back with message "no worthy mechs found" which sounds like some kind of robot battle 😁 🤖
also added a swapfile (VPS doesn't have one normally), knocked off the commercial modules (which I don't use) and managed to get a couple of eztensions and routes working (after some false starts as I'm using an unusual config linking to another virtual #pbx)
https://sangomakb.atlassian.net/wiki/spaces/PP/pages/73990871/PBX+Platforms+-+Setup+Postfix+Manually
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Gestern Abend hab ich zusammen mit einen Kumpel sein Asterisk Setup debugged, dabei kam die Frage was man denn als SIP <--> DECT Bridge und als SIP <--> POTS Bridge einsetzen mag. Geht um ein privates Setup ca 4 Endpoints), sollte also mäßig Admin-Aufwände erzeugen, sollte nicht super teuer sein, dafür mäßige Availability akzeptabel.
Da ich weniger Ahnung habe als ihr gebe ich die Frage gerne mal weiter
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Gestern Abend hab ich zusammen mit einen Kumpel sein Asterisk Setup debugged, dabei kam die Frage was man denn als SIP <--> DECT Bridge und als SIP <--> POTS Bridge einsetzen mag. Geht um ein privates Setup ca 4 Endpoints), sollte also mäßig Admin-Aufwände erzeugen, sollte nicht super teuer sein, dafür mäßige Availability akzeptabel.
Da ich weniger Ahnung habe als ihr gebe ich die Frage gerne mal weiter
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Gestern Abend hab ich zusammen mit einen Kumpel sein Asterisk Setup debugged, dabei kam die Frage was man denn als SIP <--> DECT Bridge und als SIP <--> POTS Bridge einsetzen mag. Geht um ein privates Setup ca 4 Endpoints), sollte also mäßig Admin-Aufwände erzeugen, sollte nicht super teuer sein, dafür mäßige Availability akzeptabel.
Da ich weniger Ahnung habe als ihr gebe ich die Frage gerne mal weiter
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Gestern Abend hab ich zusammen mit einen Kumpel sein Asterisk Setup debugged, dabei kam die Frage was man denn als SIP <--> DECT Bridge und als SIP <--> POTS Bridge einsetzen mag. Geht um ein privates Setup ca 4 Endpoints), sollte also mäßig Admin-Aufwände erzeugen, sollte nicht super teuer sein, dafür mäßige Availability akzeptabel.
Da ich weniger Ahnung habe als ihr gebe ich die Frage gerne mal weiter
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Gestern Abend hab ich zusammen mit einen Kumpel sein Asterisk Setup debugged, dabei kam die Frage was man denn als SIP <--> DECT Bridge und als SIP <--> POTS Bridge einsetzen mag. Geht um ein privates Setup ca 4 Endpoints), sollte also mäßig Admin-Aufwände erzeugen, sollte nicht super teuer sein, dafür mäßige Availability akzeptabel.
Da ich weniger Ahnung habe als ihr gebe ich die Frage gerne mal weiter
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Everything now seems to be working. Now I know SIP trunks work even on this old server and versions of #Asterisk / #FreePBX I can plan for when our main analogue lines have to be ceased and reprovided as SIP (and will be looking into using a cloud server / VPS as its hardware is getting old, and should stop a problem we have at a remote site where the ISP controls the router and won't open up firewall ports other than as chargeable work (which means I had to use a different cloud #PBX service for that site)
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Alas, log does not seem to get picked up by logrotate - changed filename to /var/log/asterisk/fail2ban (already in /etc/logrotate.d and previously working) to see if thats any better (as apparently #FreePBX can alter /etc/logrotate.d but its not clear exactly where this happens!)
it turns out maybe some regexes in fail2ban may have been fine, but the full log generated by #Asterisk didn't contain "security" events so it couldn't find any to catch). I've also added "notice" to the security log and the regex *now* seems to snag these!
Turned off FreePBX software #firewall as fighting with #fail2ban #iptables rules (never worked straight anyway and didn't guard #SIP traffic), checking if config persist across reboots and services start correctly.. #VOIP
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Had to open 5060 inbound to get one providers trunk to signal inbound calls (either #STUN isn't working there or some #NAT issues), with predictable results..
Got older version of #fail2ban on this box to yeet all blighters trying to get in - by turning on security logging in /etc/asterisk/logfiles_custom.conf (add entry security_log => security), updating regexes in /etc/fail2ban/filter.d and pointing failt2ban jail to check /var/log/asterisk/security_log (main Asterisk log is in wrong format and I don't know enough regex to fix that)
Also registered a #Voipfone virtual PBX extension to use as an extra trunk (needs contact-user and from-user set in #PJSIP config)
The picture @alex drew a few months back sums up exactly what dealing with these #VOIP #trunks is like
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Just spent half day reconfiguring #SIP trunks on #FreePBX and #Asterisk (using chan_pjsip) as one of our providers silently yoinked #IAX2 support - #VOIP is as cursed if not more so than #PSTN / #ISDN circuits, except you don't have to crawl around as much in corners and roofspaces amongst spiders, mouse-like rodents and possibly snakes (if you have them in your country), and there's less chance of ending up on the wrong side of 100-120 volts (either AC or DC, depending on whether its ringing voltage or the strong DC voltage that British Telecom and others used to send down certain ISDN lines in 90s/00s)
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Kích hoạt cuộc gọi từ Home Assistant sử dụng FreePBX? Cần thiết lập Asterisk integration và AMI Manager User.
#HomeAssistant #FreePBX #Asterisk #TựĐộngHóa #Automate #CuộcGọi #PhoneCall #IoT #Smarthome #TíchHợp #Integration -
Aktywnie wykorzystywana krytyczna podatność w wirtualnej centrali FreePBX
Ataki na rozwiązania używane do świadczenia usług VoIP nie są niczym nowym. Wielokrotnie pisaliśmy też na łamach sekuraka na temat (nie)bezpieczeństwa wszelakich interfejsów webowych. Tym razem mamy połączenie jednego i drugiego. TLDR: – Wykryto krytyczną (CVSS 10.0) podatność w FreePBX, interfejsie webowym dla Asterisk – Podatność jest aktywnie wykorzystywana przez...
#WBiegu #Asterisk #Freepbx #Podatność #SiP #VoIP
https://sekurak.pl/aktywnie-wykorzystywana-krytyczna-podatnosc-w-wirtualnej-centrali-freepbx/
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Finally got around to upgrading my company's #Asterisk box to #PJSIP last night! So glad I did.
We now have proper SNI support, better NAT traversal, better header management, #IPv6 support in progress, and perhaps most importantly, are now using something under active development, so security issues will be fixed sooner.
Please move away from chan_sip. It's been deprecated since 2019, and completely removed from Asterisk as of 2023.
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Интеграция Битрикс24 и Asterisk: что это такое и для чего нужно
Современные компании всё чаще сталкиваются с необходимостью интеграции IP-телефонии и CRM-систем для улучшения управления клиентскими коммуникациями. Такая интеграция позволяет автоматизировать ключевые процессы, минимизировать человеческий фактор и ускорить обработку обращений. Модуль интеграции
https://habr.com/ru/articles/909268/
#телефония #asterisk #автоматизация #sipтелефония #ipатс #битрикс24 #bitrix24 #интеграция_с_crm #интеграция_с_атс #crm
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Imagine you work in the food delivery service. Making pizzas for example.
Are you accepting a call of a robot ordering a pizza for someone?
.
Asking for sascha my phone assistant. -
Had an issue yesterday with an #Asterisk install not seeing a key I have for connections to another via #IAX
Debugged it for a few hours after work and gave up. Went to bed and dreamt a possible solution. Woke up, went back to work and tried said solution and that was it!I simply forgot to install openssl-dev and compile Asterisk --with-ssl ..
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Had an issue yesterday with an #Asterisk install not seeing a key I have for connections to another via #IAX
Debugged it for a few hours after work and gave up. Went to bed and dreamt a possible solution. Woke up, went back to work and tried said solution and that was it!I simply forgot to install openssl-dev and compile Asterisk --with-ssl ..
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Had an issue yesterday with an #Asterisk install not seeing a key I have for connections to another via #IAX
Debugged it for a few hours after work and gave up. Went to bed and dreamt a possible solution. Woke up, went back to work and tried said solution and that was it!I simply forgot to install openssl-dev and compile Asterisk --with-ssl ..
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Had an issue yesterday with an #Asterisk install not seeing a key I have for connections to another via #IAX
Debugged it for a few hours after work and gave up. Went to bed and dreamt a possible solution. Woke up, went back to work and tried said solution and that was it!I simply forgot to install openssl-dev and compile Asterisk --with-ssl ..
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Had an issue yesterday with an #Asterisk install not seeing a key I have for connections to another via #IAX
Debugged it for a few hours after work and gave up. Went to bed and dreamt a possible solution. Woke up, went back to work and tried said solution and that was it!I simply forgot to install openssl-dev and compile Asterisk --with-ssl ..
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> "They were so aggressive in their text!"
< "Why, what'd they do?"
> "They used an ellipsis!"
Real conversation. I'm looking around for Alan Funt. I can't even imagine their reaction if it had been an octothorpe. [1]
#aggressive #punctuation #trigger #ellipsis #octothorpe #ampersand #hyphen #caret #parentheses #brackets #braces #asterisk
[1] Why yes, Firefox spellchecker, when I typed "octothorpe" I clearly really meant to type "#clotheshorse".
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> "They were so aggressive in their text!"
< "Why, what'd they do?"
> "They used an ellipsis!"
Real conversation. I'm looking around for Alan Funt. I can't even imagine their reaction if it had been an octothorpe. [1]
#aggressive #punctuation #trigger #ellipsis #octothorpe #ampersand #hyphen #caret #parentheses #brackets #braces #asterisk
[1] Why yes, Firefox spellchecker, when I typed "octothorpe" I clearly really meant to type "#clotheshorse".
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> "They were so aggressive in their text!"
< "Why, what'd they do?"
> "They used an ellipsis!"
Real conversation. I'm looking around for Alan Funt. I can't even imagine their reaction if it had been an octothorpe. [1]
#aggressive #punctuation #trigger #ellipsis #octothorpe #ampersand #hyphen #caret #parentheses #brackets #braces #asterisk
[1] Why yes, Firefox spellchecker, when I typed "octothorpe" I clearly really meant to type "#clotheshorse".
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> "They were so aggressive in their text!"
< "Why, what'd they do?"
> "They used an ellipsis!"
Real conversation. I'm looking around for Alan Funt. I can't even imagine their reaction if it had been an octothorpe. [1]
#aggressive #punctuation #trigger #ellipsis #octothorpe #ampersand #hyphen #caret #parentheses #brackets #braces #asterisk
[1] Why yes, Firefox spellchecker, when I typed "octothorpe" I clearly really meant to type "#clotheshorse".
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> "They were so aggressive in their text!"
< "Why, what'd they do?"
> "They used an ellipsis!"
Real conversation. I'm looking around for Alan Funt. I can't even imagine their reaction if it had been an octothorpe. [1]
#aggressive #punctuation #trigger #ellipsis #octothorpe #ampersand #hyphen #caret #parentheses #brackets #braces #asterisk
[1] Why yes, Firefox spellchecker, when I typed "octothorpe" I clearly really meant to type "#clotheshorse".
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Настройка Anycast-адреса в рамках бюджетного тестового стенда
В рамках IPv4 и IPv6 есть понятие Anycast-адресов . Если упрощать, то это IP-адреса выглядящие как обычные «серые» или «белые» адреса, но которые одновременно могут работать как на одном сервере, так и на множестве. Есть мнение, что это сложно настраивается, требует много дополнительных слоев маршрутизирующего оборудования и т.д. Но в данной статье я попробую описать настройку Anycast-адреса где угодно и с минимальными затратами.
https://habr.com/ru/companies/skbkontur/articles/848676/
#mikrotik #anycast #udp #voip #asterisk #opensips #bird #debian #ospf #ospfv2
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Настройка Anycast-адреса в рамках бюджетного тестового стенда
В рамках IPv4 и IPv6 есть понятие Anycast-адресов . Если упрощать, то это IP-адреса выглядящие как обычные «серые» или «белые» адреса, но которые одновременно могут работать как на одном сервере, так и на множестве. Есть мнение, что это сложно настраивается, требует много дополнительных слоев маршрутизирующего оборудования и т.д. Но в данной статье я попробую описать настройку Anycast-адреса где угодно и с минимальными затратами.
https://habr.com/ru/companies/skbkontur/articles/848676/
#mikrotik #anycast #udp #voip #asterisk #opensips #bird #debian #ospf #ospfv2
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Настройка Anycast-адреса в рамках бюджетного тестового стенда
В рамках IPv4 и IPv6 есть понятие Anycast-адресов . Если упрощать, то это IP-адреса выглядящие как обычные «серые» или «белые» адреса, но которые одновременно могут работать как на одном сервере, так и на множестве. Есть мнение, что это сложно настраивается, требует много дополнительных слоев маршрутизирующего оборудования и т.д. Но в данной статье я попробую описать настройку Anycast-адреса где угодно и с минимальными затратами.
https://habr.com/ru/companies/skbkontur/articles/848676/
#mikrotik #anycast #udp #voip #asterisk #opensips #bird #debian #ospf #ospfv2
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MГТС GPON: SFP ONU + Mikrotik + Asterisk
Очень кратенько, в дополнение к следующим статьям: https://habr.com/ru/companies/ruvds/articles/547442/ https://habr.com/ru/articles/724566/ https://habr.com/ru/articles/553118/
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Как сделать виртуальную АТС на базе VPS
Несмотря на популярность мессенджеров и телеконференций, ни один офис ещё не отказался от телефонной связи. Люди такие существа, что иногда предпочитают общаться голосом. В каждом офисе установлена мини-АТС, которая коммутирует внутренние звонки. Телефоны сотрудников подключаются к коммуникационному шкафу или коробочке с Asterisk (как на КДПВ), а она подключена к телефонной сети общего пользования (PSTN или ТСОП). Таким образом, сотню офисных телефонов можно повесить на один внешний номер. В общем, мини-АТС — совершенно необходимая вещь. Виртуальная или облачная АТС (hosted PBX) — это услуга для компаний, которая заменяет им обычную офисную АТС. Вместо того, чтобы покупать специализированное телекоммуникационное оборудование или выделять отдельный компьютер с Asterisk , они заказывают услугу на удалённом хостинге. И этот компьютер с Asterisk (IP-АТС) физически размещается у провайдера. Таким образом, виртуализация добралась и до АТС, всё в русле современных тенденций.
https://habr.com/ru/companies/ruvds/articles/814083/
#ruvds_статьи #АТС #виртуальная_АТС #Asterisk #PBX #IPАТС #SIP #Prometheus #аудиоскремблер #распознавание_речи #IPтелефония #Linphone #VoIP #PSTN #ТСОП #FreePBX #FreePBX_Distro #InterAsterisk_eXchange #IAX #RFC_5456
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Как сделать виртуальную АТС на базе VPS
Несмотря на популярность мессенджеров и телеконференций, ни один офис ещё не отказался от телефонной связи. Люди такие существа, что иногда предпочитают общаться голосом. В каждом офисе установлена мини-АТС, которая коммутирует внутренние звонки. Телефоны сотрудников подключаются к коммуникационному шкафу или коробочке с Asterisk (как на КДПВ), а она подключена к телефонной сети общего пользования (PSTN или ТСОП). Таким образом, сотню офисных телефонов можно повесить на один внешний номер. В общем, мини-АТС — совершенно необходимая вещь. Виртуальная или облачная АТС (hosted PBX) — это услуга для компаний, которая заменяет им обычную офисную АТС. Вместо того, чтобы покупать специализированное телекоммуникационное оборудование или выделять отдельный компьютер с Asterisk , они заказывают услугу на удалённом хостинге. И этот компьютер с Asterisk (IP-АТС) физически размещается у провайдера. Таким образом, виртуализация добралась и до АТС, всё в русле современных тенденций.
https://habr.com/ru/companies/ruvds/articles/814083/
#ruvds_статьи #АТС #виртуальная_АТС #Asterisk #PBX #IPАТС #SIP #Prometheus #аудиоскремблер #распознавание_речи #IPтелефония #Linphone #VoIP #PSTN #ТСОП #FreePBX #FreePBX_Distro #InterAsterisk_eXchange #IAX #RFC_5456
