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331 results for “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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been checking last few days #blighters who are kicked out trying to get at our #VOIP #PBX - they don't even seem to be wanting to use the #SIP #trunks for getting free #telephone calls for actually talking to people (even spam/coldcalls), but appear to be edgelords attempting to use them for their own private vendetta and DDOS some individual/business phone as INVITE attempts are all to the same USA number(s)
I'm assuming its not someone trying to call their *own* phone to find when someone *has* left a trunk open, as that would surely create a data trail authorities could pick up on?
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I made up a folder for all my #telecoms tech info (for two #PBX I built, general #SIP #VOIP and #ATA configurations and other useful stuff as much is easier to read in paper form than on the screen!)
I thought the "Trunks" cartoon from @alex was a good choice for the cover, along with the "ELEPHANT EXCHONGE" lettering from a Telephone Exchange in SE England (after #Openreach engineers moved the individual letters around) 😁
There's other stuff in the folder than the Ofcom and ITU documents, about internal configs but I can't share it for cybersecurity reasons (I will put the sanitised info on my blog at some point to help others setting up VOIP systems)
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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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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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How difficult would it be to set up a #PBX at home and make it callable from the PSTN? :3c -
#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 😁
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I think my other 21A (hard to get nowadays, and sell for silly prices, more than the telephones themselves) is in the 8746 I built a few years back with a flaky hookswitch, I will try that in this phone and then link it to a Grandstream ATA I got which is supposed to work with loop disconnect dialling, and connect it to the cloud #VOIP #PBX I recently built 😁 ☎️
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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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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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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
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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
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Progress has been slow but now I have the first revision of my modular analog PBX back plane PCB in my hands! ☎️
This board has eight line card slots for either FXS or FXO interfaces and three generic expansion slots, ring generator using an H-bridge plus external step-up transformer, progress tones, DTMF decoder (and pulse dialing of course), four simultaneous calls with balanced audio paths. Now I can finally hook up my modems without messing with VoIP! #pbx #retrocomputing #pcbway
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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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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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Send an SMS to confirm you're a human? That's strange. How about dozens of SMS, to locations all over the world? That sounds more like a hot take on International Revenue Share Fraud (IRSF). Infoblox Threat Intel has come across an operation that defrauds both individuals and telecoms by way of social engineering victims through the use of a fake CAPTCHA process.
With IRSF, fraudsters generate their revenue by driving call or SMS traffic to numbers to which they have revenue sharing agreements with the local telecoms. Historically, this has been done by methods like hacking an organization's PBX system, or using bots to abuse services that generate one-time-passwords, and directing that call or SMS traffic to numbers under their control.
This operation, however, takes advantage of individuals' familiarity with the CAPTCHA process, by adding a multi-stage requirement to send bulk SMS to get access to games, videos, or adult content - because of course, these things are so hard to access online otherwise.
In this case, the victims are two-fold. First, it impacts the people who get unexpected international SMS charges on their bill, and then the telecoms who both pay termination fees to the international destinations telecom, and who also possibly absorb the cost of the chargeback.Read more about our investigation into this new flavour of scam, including the specific domains and infrastructure we uncovered, here: https://www.infoblox.com/blog/threat-intelligence/hold-the-phone-international-revenue-share-fraud-driven-by-fake-captchas/
#threatintel #cybercrime #threatintelligence #cybersecurity #infosec #infoblox #infobloxthreatintel #scam #phishing #IRSF #telecom #captcha
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Send an SMS to confirm you're a human? That's strange. How about dozens of SMS, to locations all over the world? That sounds more like a hot take on International Revenue Share Fraud (IRSF). Infoblox Threat Intel has come across an operation that defrauds both individuals and telecoms by way of social engineering victims through the use of a fake CAPTCHA process.
With IRSF, fraudsters generate their revenue by driving call or SMS traffic to numbers to which they have revenue sharing agreements with the local telecoms. Historically, this has been done by methods like hacking an organization's PBX system, or using bots to abuse services that generate one-time-passwords, and directing that call or SMS traffic to numbers under their control.
This operation, however, takes advantage of individuals' familiarity with the CAPTCHA process, by adding a multi-stage requirement to send bulk SMS to get access to games, videos, or adult content - because of course, these things are so hard to access online otherwise.
In this case, the victims are two-fold. First, it impacts the people who get unexpected international SMS charges on their bill, and then the telecoms who both pay termination fees to the international destinations telecom, and who also possibly absorb the cost of the chargeback.Read more about our investigation into this new flavour of scam, including the specific domains and infrastructure we uncovered, here: https://www.infoblox.com/blog/threat-intelligence/hold-the-phone-international-revenue-share-fraud-driven-by-fake-captchas/
#threatintel #cybercrime #threatintelligence #cybersecurity #infosec #infoblox #infobloxthreatintel #scam #phishing #IRSF #telecom #captcha
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Send an SMS to confirm you're a human? That's strange. How about dozens of SMS, to locations all over the world? That sounds more like a hot take on International Revenue Share Fraud (IRSF). Infoblox Threat Intel has come across an operation that defrauds both individuals and telecoms by way of social engineering victims through the use of a fake CAPTCHA process.
With IRSF, fraudsters generate their revenue by driving call or SMS traffic to numbers to which they have revenue sharing agreements with the local telecoms. Historically, this has been done by methods like hacking an organization's PBX system, or using bots to abuse services that generate one-time-passwords, and directing that call or SMS traffic to numbers under their control.
This operation, however, takes advantage of individuals' familiarity with the CAPTCHA process, by adding a multi-stage requirement to send bulk SMS to get access to games, videos, or adult content - because of course, these things are so hard to access online otherwise.
In this case, the victims are two-fold. First, it impacts the people who get unexpected international SMS charges on their bill, and then the telecoms who both pay termination fees to the international destinations telecom, and who also possibly absorb the cost of the chargeback.Read more about our investigation into this new flavour of scam, including the specific domains and infrastructure we uncovered, here: https://www.infoblox.com/blog/threat-intelligence/hold-the-phone-international-revenue-share-fraud-driven-by-fake-captchas/
#threatintel #cybercrime #threatintelligence #cybersecurity #infosec #infoblox #infobloxthreatintel #scam #phishing #IRSF #telecom #captcha
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Send an SMS to confirm you're a human? That's strange. How about dozens of SMS, to locations all over the world? That sounds more like a hot take on International Revenue Share Fraud (IRSF). Infoblox Threat Intel has come across an operation that defrauds both individuals and telecoms by way of social engineering victims through the use of a fake CAPTCHA process.
With IRSF, fraudsters generate their revenue by driving call or SMS traffic to numbers to which they have revenue sharing agreements with the local telecoms. Historically, this has been done by methods like hacking an organization's PBX system, or using bots to abuse services that generate one-time-passwords, and directing that call or SMS traffic to numbers under their control.
This operation, however, takes advantage of individuals' familiarity with the CAPTCHA process, by adding a multi-stage requirement to send bulk SMS to get access to games, videos, or adult content - because of course, these things are so hard to access online otherwise.
In this case, the victims are two-fold. First, it impacts the people who get unexpected international SMS charges on their bill, and then the telecoms who both pay termination fees to the international destinations telecom, and who also possibly absorb the cost of the chargeback.Read more about our investigation into this new flavour of scam, including the specific domains and infrastructure we uncovered, here: https://www.infoblox.com/blog/threat-intelligence/hold-the-phone-international-revenue-share-fraud-driven-by-fake-captchas/
#threatintel #cybercrime #threatintelligence #cybersecurity #infosec #infoblox #infobloxthreatintel #scam #phishing #IRSF #telecom #captcha
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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)
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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)
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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)
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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)
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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!
-
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!