Mastodawn

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

Alex@rtnVFRmedia Suffolk UK Feb 24

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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Alex@rtnVFRmedia Suffolk UK Jan 15

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

Josh Anders Aug 29, 2025

Finally got around to upgrading my company's #Asterisk box to #PJSIP last night! So glad I did.

We now have proper SNI support, better NAT traversal, better header management, #IPv6 support in progress, and perhaps most importantly, are now using something under active development, so security issues will be fixed sooner.

Please move away from chan_sip. It's been deprecated since 2019, and completely removed from Asterisk as of 2023.

#voip

Alex@rtnVFRmedia Suffolk UK Dec 3, 2024

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

Julian Beach Mar 28, 2023

After many hours of updating config files, I have finally retired my old #asterisk server, which has been doing sterling service since 2015. Asterisk is now running on #ubuntu rather than #centos and my BT landline number is about to transfer to @aaisp so I no longer need to cater for analogue lines or deal with #wanpipe and #dahdi! Converting everything from #sip to #pjsip was a major undertaking.

benasse Jan 18, 2023

asterisk 21 will no longer ship with chan_sip !
#asterisk #pjsip
https://www.asterisk.org/asterisk-21-module-removal-chan_sip/

Asterisk 21 Module Removal - chan_sip ⋆ Asterisk

chan_sip will no longer be included with Asterisk as of the release of version 21. Deprecated in version 17, chan_sip has been scheduled for removal for

Asterisk