#freepbx — Public Fediverse posts

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

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

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

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

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..

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..

#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

#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

#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

#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

#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

Had to open 5060 inbound to get one providers trunk to signal inbound calls (either #STUN isn't working there or some #NAT issues), with predictable results..

Got older version of #fail2ban on this box to yeet all blighters trying to get in - by turning on security logging in /etc/asterisk/logfiles_custom.conf (add entry security_log => security), updating regexes in /etc/fail2ban/filter.d and pointing failt2ban jail to check /var/log/asterisk/security_log (main Asterisk log is in wrong format and I don't know enough regex to fix that)

Also registered a #Voipfone virtual PBX extension to use as an extra trunk (needs contact-user and from-user set in #PJSIP config)

The picture @alex drew a few months back sums up exactly what dealing with these #VOIP #trunks is like

#Asterisk #FreePBX

Had to open 5060 inbound to get one providers trunk to signal inbound calls (either #STUN isn't working there or some #NAT issues), with predictable results..

Got older version of #fail2ban on this box to yeet all blighters trying to get in - by turning on security logging in /etc/asterisk/logfiles_custom.conf (add entry security_log => security), updating regexes in /etc/fail2ban/filter.d and pointing failt2ban jail to check /var/log/asterisk/security_log (main Asterisk log is in wrong format and I don't know enough regex to fix that)

Also registered a #Voipfone virtual PBX extension to use as an extra trunk (needs contact-user and from-user set in #PJSIP config)

The picture @alex drew a few months back sums up exactly what dealing with these #VOIP #trunks is like

#Asterisk #FreePBX

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/

A day of #telephones - reconfigured #FreePBX #trunks that link on-premises #PBX with #cloud PBX to use #PJSIP rather than chan_sip - hopefully they will stay registered more reliably, repaired handset of desktop #VOIP phone with broken RJ9 socket using another harvested from defective analogue set (not prettiest repair, and I had to check the wiring colours as they are *different* between original and new socket), but it works #Repair #Maintenance #Telecoms

A day of #telephones - reconfigured #FreePBX #trunks that link on-premises #PBX with #cloud PBX to use #PJSIP rather than chan_sip - hopefully they will stay registered more reliably, repaired handset of desktop #VOIP phone with broken RJ9 socket using another harvested from defective analogue set (not prettiest repair, and I had to check the wiring colours as they are *different* between original and new socket), but it works #Repair #Maintenance #Telecoms