Search
331 results for “pbx”
-
Does anyone happen to have any documentation for a British Telecom "Tester 420A" ISDN BRI tester.
Also known as the T4200 - seemingly made by Fulcrum Communications in the early 1990s
It's throwing some error codes at me, and I'd really like to know what they mean!
#isdn #BritishTelecom #BT #telecoms #telecom #osmocom #pbx #telephoneExchange #telephonSystem
-
How to Install and Deploy #FusionPBX on #Debian VPS
This article provides a guide for how to install and deploy FusionPBX on Debian VPS. Installing FusionPBX on a Debian Virtual Private Server (VPS) provides a robust platform for managing Voice over IP (VoIP) communications. FusionPBX offers an intuitive web interface for managing FreeSWITCH, an open-source telephony platform.
This guide is designed for system administrators and IT ...
Continued 👉 https://blog.radwebhosting.com/how-to-install-and-deploy-fusionpbx-on-debian-vps/?utm_source=mastodon&utm_medium=social&utm_campaign=ReviveOldPost #pbx #installguide -
How to Install and Deploy #FusionPBX on #Debian VPS
This article provides a guide for how to install and deploy FusionPBX on Debian VPS. Installing FusionPBX on a Debian Virtual Private Server (VPS) provides a robust platform for managing Voice over IP (VoIP) communications. FusionPBX offers an intuitive web interface for managing FreeSWITCH, an open-source telephony platform.
This guide is designed for system administrators and IT ...
Keep reading 👉 https://blog.radwebhosting.com/how-to-install-and-deploy-fusionpbx-on-debian-vps/?utm_source=mastodon&utm_medium=social&utm_campaign=ReviveOldPost #pbx #installguide -
Does anyone know any accessible soft phone clients for windows? I'm trying to test out my asterisk system and apparently it's so difficult to make a soft phone client that's readable with a screen reader. I tried lin phone so far, all the other ones I saw have paid features or require you to have an account. All I want to do is enter my IP address and sip info and call it a day.
#accessibility #NVDA #PBX #soft phone #ipphone -
🔊 #NowPlaying on #BBC6Music's #CillianMurphysLimitedEdition
Grian Chatten:
🎵 All Of The Peoplehttps://open.spotify.com/track/2iXFX4oPBXFVIADlUuHIgq
https://grianchatten.bandcamp.com/track/all-of-the-people
#BandcampPlease 🔁 BOOST to share what you like
- your followers don't see if you ⭐ favourite a post -
Are there any #Telecoms folk out there who have a copy of the documentation for an ancient #ISDX DLI card, part No 871/2/06561/004
I want to check what signalling the 06561/004 variant supports
I've got the docs for the 06562/004 which I think is similar (but later?) card - but I'm wary of extrapolating...
I wouldn't normally ask, but please boost for reach?
-
How to Install and Deploy FusionPBX on Debian VPS #certbot #debian #freeswitch #fusionpbx #installguide #letsencrypt #nginx #opensource #pbx #php #postgresql #selfhosted #selfhosting #ufw #voip #vps #Cloud #Guides #VPS blog.radwebhosting.com/deploy-fusio...
How to Install and Deploy Fusi... -
Read the NEW Coach Bennett’s Newsletter now! Welcome Back To PBX is ready for your 👀 eyes. Cheers and thank you for reading! #coachbennett https://open.substack.com/pub/coach3s23a/p/welcome-back-to-pbx?r=1gdsu0&utm_campaign=post&utm_medium=email
-
También se puede ver en el fediverso (de hecho estuvo antes ahí) https://fediverse.tv/w/7sQJDSwaSrcBTTApBxKP9K #fediversetv #laTVdelFediverso
-
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)
-
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)
-
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)
-
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)
-
SIP Trunk, Explained
The SIP trunk carries all voice traffic between your PBX and the network. It scales up or down with load. VoipTower focuses on routing stability and metrics transparency.
In simple terms:
A “voice lane” that grows with your call volume.
Key takeaway:
Scale capacity, keep control.
Closing line:
Need specifics on codecs, capacity, or failover? Ask below.
#did #voipfone #SipPro #voiptower -
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
-
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
-
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
-
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
-
Web application security has DVWA and WebGoat. VoIP and WebRTC security hasn't had anything like it ... until now.
We built DVRTC (Damn Vulnerable Real-Time Communications): a hands-on lab for learning VoIP/WebRTC attack techniques. Full dockerized stack with Kamailio, Asterisk, rtpengine, and coturn — each configured to exhibit specific vulnerable behaviors.
7 exercises covering SIP extension enumeration, RTP bleed, SIP digest leaks, credential cracking (online and offline), TURN relay abuse, and traffic analysis. There's a live instance at pbx1.dvrtc.net you can test against right now.
https://www.enablesecurity.com/blog/introducing-dvrtc-damn-vulnerable-real-time-communications/
GitHub: https://github.com/EnableSecurity/DVRTC/
#infosec #webrtc #voipsecurity #sipsecurity #penetrationtesting #training #TURN
-
@CIO @spaetz Webseite der EU Ausschreibung wurde korrigiert.
-
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!
-
About XMPP universe-ality, Jingle is an extension adding peer-to-peer #P2P signaling, such as in #VoIP and video-conferencing. Some software using XMPP Jingle. From wikipedia.org/wiki/Jingle_(protocol):
#Asterisk PBX #telephony
#Coccinella
#Conversations
#Empathy
#FreeSWITCH
#Gajim
GoogleTalk for Gmail, Android
#iChat
#Jitsi
#KDE #Telepathy
#Kopete
#Miranda NG
#Monal IM-Client
#Pidgin
#Psi
#QIP Infium
#Yate
#RemoteVNC
-
Also #DinoIM https://dino.im -
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)
-
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 😁
-
CW: lots of nerd talk about dreamcast and phone lines with dubious delivery
In the service of totally overdoing some #Dreamcast #RetroGaming, I've successfully patched the #DreamPi gateway to be less useful. Buckle up, because this is a kinda long post with little in the way of delivery. :bec_smug:
Normally the DreamPi throws its modem into voice mode, plays a dial tone to fake out the dreamcast, listens to the digits dialed, and then switches to data mode and kicks off a handshake. Stays off-hook the entire time.
This makes sense if you're using an electrically-simple Line Voltage Inducer and a straight connection between the Dreamcast and the modem for minimum part count get-up-and-go. It's actually a really clever workaround.
My tweak adjusts the modem behavior such that it stays on-hook until it detects an incoming ring, then answers and jumps straight into data mode.
This change was made because presently we're using a Valcom DLE-200B phone line simulator which does provide 'local' tone, ringing voltage, and ringdown.
This necessarily means that we lose out on capturing the dialed digits, but it turns out we don't need those at all for pretty much any of the PPP-based games with central servers, and I'm not interested in playing anything outside of that scope.
So now we have a dreampi that can only work in PPP mode and it requires a line simulator to work correctly. Why?
All, ultimately, so that we can hook the DreamPi and two dreamcasts up to a TDM-based #PBX instead of the line simulator, wherein either of the latter can dial the former, thinking they're dialing Dreamcast Online Services. It goes through the whole ring-in process, which terminates at DreamPi, handshakes, and tunnels to Dreamcast Live -- getting us the goods.
And there's lots of Friday left to go :3
#RetroGaming #OtherNetworks #PhoneLab #RetroLab #POTS #Modem
-
Hnngh, talk about an anticlimax. Been hunting down this IP Office chassis to install in the telecoms lab and it *seems* to have a dead internal PSU.
It's gonna get repaired, of course, but it just totally killed all immediate forward progress.
#VoiceLab #OtherNetworks #HomeLab #RetroLab #Avaya #PhoneSystems #PBX #IPOffice
-
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
-
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)