#freepbx — Public Fediverse posts
Live and recent posts from across the Fediverse tagged #freepbx, aggregated by home.social.
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@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.
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@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.
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@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.
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@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.
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@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.
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this is just cursed behaviour - whather #Groundwire registers when something else is using same external IP depends on what order SIP registrations arrive!
In this case its not a disaster as if I am on site I have far better devices available than my mobile phone to make SIP calls with (or can just use its LTE data), but its something worth being wary about as it could bite you in the arse if you are setting this up in somewhere where you don't have access to a different IP address to that shared by a #FreePBX inter-PBX #trunk
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this is just cursed behaviour - whather #Groundwire registers when something else is using same external IP depends on what order SIP registrations arrive!
In this case its not a disaster as if I am on site I have far better devices available than my mobile phone to make SIP calls with (or can just use its LTE data), but its something worth being wary about as it could bite you in the arse if you are setting this up in somewhere where you don't have access to a different IP address to that shared by a #FreePBX inter-PBX #trunk
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this is just cursed behaviour - whather #Groundwire registers when something else is using same external IP depends on what order SIP registrations arrive!
In this case its not a disaster as if I am on site I have far better devices available than my mobile phone to make SIP calls with (or can just use its LTE data), but its something worth being wary about as it could bite you in the arse if you are setting this up in somewhere where you don't have access to a different IP address to that shared by a #FreePBX inter-PBX #trunk
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this is just cursed behaviour - whather #Groundwire registers when something else is using same external IP depends on what order SIP registrations arrive!
In this case its not a disaster as if I am on site I have far better devices available than my mobile phone to make SIP calls with (or can just use its LTE data), but its something worth being wary about as it could bite you in the arse if you are setting this up in somewhere where you don't have access to a different IP address to that shared by a #FreePBX inter-PBX #trunk
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this is just cursed behaviour - whather #Groundwire registers when something else is using same external IP depends on what order SIP registrations arrive!
In this case its not a disaster as if I am on site I have far better devices available than my mobile phone to make SIP calls with (or can just use its LTE data), but its something worth being wary about as it could bite you in the arse if you are setting this up in somewhere where you don't have access to a different IP address to that shared by a #FreePBX inter-PBX #trunk
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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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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!
-
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!
-
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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CVE Alert: CVE-2026-40520 - FreePBX - api - https://www.redpacketsecurity.com/cve-alert-cve-2026-40520-freepbx-api/
#OSINT #ThreatIntel #CyberSecurity #cve-2026-40520 #freepbx #api
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Ce téléphone est maintenant au monde des hackerspaces et est validé par les Blåhaj locaux ! #voip #freepbx #analogphone
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Ce téléphone est maintenant au monde des hackerspaces et est validé par les Blåhaj locaux ! #voip #freepbx #analogphone
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Ce téléphone est maintenant au monde des hackerspaces et est validé par les Blåhaj locaux ! #voip #freepbx #analogphone
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Ce téléphone est maintenant au monde des hackerspaces et est validé par les Blåhaj locaux ! #voip #freepbx #analogphone
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Ce téléphone est maintenant au monde des hackerspaces et est validé par les Blåhaj locaux ! #voip #freepbx #analogphone
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So, my initial review of the Yaelink SIP-T42S before I have the in hand:
- The documentation exists, but can be a pain in the ass to track down exactly what you need.
- Looks like I've got everything already setup to serve the boot configurations and should work with my FreePBX installation. I'll probably get these setup as backups in case I lose a couple of Digium D40 phones during the competition.
- Firmware is still available for the phone, but it is EoL.
- English isn't the greatest in the documentation, but it gets the job done.
- Cheap as hell as I bought 15 of these for $34 which is absolutely insane. -
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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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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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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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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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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Finished writing the docs for my FreePBX-17-Container Repository.
Took around an hour, felt like an eternity.
Hope they help folks understand the ~magic~ behind it better.
Repo can be found here: https://codeberg.org/Spoljarevic/FreePBX-17-Container/
#docs #documentation #wiki #freepbx #freepbx17 #container #containerization #docker #dockercompose
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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)
-
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)
-
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)
-
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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As an experiment, I asked #MS365 #Copilot to explain how to set up #SIP trunk on #FreePBX using #PJSIP (consider that I have already successfully set up several of these, to external providers and an inter-PBX line between two servers).
Results it returned were horribly mangled and mixed up from various providers sites, if you followed them the trunk likely won't work at all, and even if it did it would end up in completely wrong context/dialplan.
It didn't mention such things as fromuser and took a few prompts to point out potential firewall issues.
You need to know (or learn) the basics of #telephony before starting, or else it will all go tits up very quickly - #AI is still no substitute for "boots on the ground" who have put in research for what they are trying to achieve..
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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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#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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#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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#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 😁
-
#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 😁
-
#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 😁
-
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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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
