#freepbx — Public Fediverse posts

@buersten
Ich würde eine reine SIP-Lösung z.B. #FreePBX oder von #Unifi nehmen.

Bei uns läuft eine FreePBX auf einem #RaspberryPi und als Clients verwenden wir Geräte von #Grandstream.

@buersten
Ich würde eine reine SIP-Lösung z.B. #FreePBX oder von #Unifi nehmen.

Bei uns läuft eine FreePBX auf einem #RaspberryPi und als Clients verwenden wir Geräte von #Grandstream.

@buersten
Ich würde eine reine SIP-Lösung z.B. #FreePBX oder von #Unifi nehmen.

Bei uns läuft eine FreePBX auf einem #RaspberryPi und als Clients verwenden wir Geräte von #Grandstream.

@buersten
Ich würde eine reine SIP-Lösung z.B. #FreePBX oder von #Unifi nehmen.

Bei uns läuft eine FreePBX auf einem #RaspberryPi und als Clients verwenden wir Geräte von #Grandstream.

@buersten
Ich würde eine reine SIP-Lösung z.B. #FreePBX oder von #Unifi nehmen.

Bei uns läuft eine FreePBX auf einem #RaspberryPi und als Clients verwenden wir Geräte von #Grandstream.

todays #VOIP discovery- - found another snakehead at the end of a #trunk - this time I am using #Acrobits #Groundwire #SIP client on #Android for a mobile extension on #cloud #PBX

Works *unless* I use a wifi connection with same external IP address as on-site PBX connected to cloud PBX (registered as PJSIP interPBX trunk and IP authentication).

When Groundwire extension tries to register as #endpoint on #FreePBX, #AOR records get all confused and #Groundwire shows "error"

tried adding external IP address to "Match (Permit)" in FreePBX extension entry - alas - this allows Groundwire to work but hoses outbound calls from the on-site PBX so had to be reverted (not a complete disaster as I can use the other wifi connection or LTE for Groundwire)

todays #VOIP discovery- - found another snakehead at the end of a #trunk - this time I am using #Acrobits #Groundwire #SIP client on #Android for a mobile extension on #cloud #PBX

Works *unless* I use a wifi connection with same external IP address as on-site PBX connected to cloud PBX (registered as PJSIP interPBX trunk and IP authentication).

When Groundwire extension tries to register as #endpoint on #FreePBX, #AOR records get all confused and #Groundwire shows "error"

tried adding external IP address to "Match (Permit)" in FreePBX extension entry - alas - this allows Groundwire to work but hoses outbound calls from the on-site PBX so had to be reverted (not a complete disaster as I can use the other wifi connection or LTE for Groundwire)

todays #VOIP discovery- - found another snakehead at the end of a #trunk - this time I am using #Acrobits #Groundwire #SIP client on #Android for a mobile extension on #cloud #PBX

Works *unless* I use a wifi connection with same external IP address as on-site PBX connected to cloud PBX (registered as PJSIP interPBX trunk and IP authentication).

When Groundwire extension tries to register as #endpoint on #FreePBX, #AOR records get all confused and #Groundwire shows "error"

tried adding external IP address to "Match (Permit)" in FreePBX extension entry - alas - this allows Groundwire to work but hoses outbound calls from the on-site PBX so had to be reverted (not a complete disaster as I can use the other wifi connection or LTE for Groundwire)

todays #VOIP discovery- - found another snakehead at the end of a #trunk - this time I am using #Acrobits #Groundwire #SIP client on #Android for a mobile extension on #cloud #PBX

Works *unless* I use a wifi connection with same external IP address as on-site PBX connected to cloud PBX (registered as PJSIP interPBX trunk and IP authentication).

When Groundwire extension tries to register as #endpoint on #FreePBX, #AOR records get all confused and #Groundwire shows "error"

tried adding external IP address to "Match (Permit)" in FreePBX extension entry - alas - this allows Groundwire to work but hoses outbound calls from the on-site PBX so had to be reverted (not a complete disaster as I can use the other wifi connection or LTE for Groundwire)

todays #VOIP discovery- - found another snakehead at the end of a #trunk - this time I am using #Acrobits #Groundwire #SIP client on #Android for a mobile extension on #cloud #PBX

Works *unless* I use a wifi connection with same external IP address as on-site PBX connected to cloud PBX (registered as PJSIP interPBX trunk and IP authentication).

When Groundwire extension tries to register as #endpoint on #FreePBX, #AOR records get all confused and #Groundwire shows "error"

tried adding external IP address to "Match (Permit)" in FreePBX extension entry - alas - this allows Groundwire to work but hoses outbound calls from the on-site PBX so had to be reverted (not a complete disaster as I can use the other wifi connection or LTE for Groundwire)

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!

#VOIP #telephony

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!

#VOIP #telephony

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!

#VOIP #telephony

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!

#VOIP #telephony

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!

#VOIP #telephony

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)

#Telecoms #Telephony

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)

#Telephony #Asterisk

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)

#Telephony #Asterisk

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)

#Telephony #Asterisk

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)

#Telephony #Asterisk

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)

#Telephony #Asterisk

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..

#Telephone set #BritishTelecom 8746 now with replaced 21A microphone insert, working correctly with #Grandstream #HT802 v2 ATA linked to #FreePBX I built on cloud VPS - accepts both #MF (tone) and #LoopDisconnect #dialling (although dialling a full UK mobile number is quite a long process and I had to make sure the timer was at least 4 seconds (or the digits get sent to register before you've dialled any 0 and call fails due to wrong number being sent to #PBX !)

Ring voltage (set to 55V RMS) is strong enough to ring the 4k bell in 8746

So currently this corner of the office looks like its back in 1980s 😁

#VOIP #Telephony

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

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)

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)

#Telecoms

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

https://www.reddit.com/r/selfhosted/comments/1ovt7z7/how_to_trigger_a_phone_call_from_home_assistant/

You Already Have Our Data, Take Our Phone Calls Too (FreePBX CVE-2025-57819)

https://labs.watchtowr.com/you-already-have-our-personal-data-take-our-phone-calls-too-freepbx-cve-2025-57819/

#HackerNews #You #Already #Have #Our #Data #Take #Our #Phone #Calls #Too #FreePBX #CVE-2025-57819 #Cybersecurity #Vulnerability #DataPrivacy #TechNews

Experts warn of actively exploited FreePBX zero-day – Source: securityaffairs.com https://ciso2ciso.com/experts-warn-of-actively-exploited-freepbx-zero-day-source-securityaffairs-com/ #rssfeedpostgeneratorecho #informationsecuritynews #ITInformationSecurity #SecurityAffairscom #CyberSecurityNews #PierluigiPaganini #SecurityAffairs #SecurityAffairs #CVE-2025-57819 #BreakingNews #SecurityNews #hackingnews #Security #FreePBX #hacking #zeroday

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

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

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

Zero-Day Vulnerabilities Compromise All FreePBX Systems https://nerdvittles.com/zero-day-vulnerabilities-compromise-all-freepbx-systems/ #asterisk #freepbx #IncrediblePBX #Defcon31 #voip