Disaster has struck, rebuilt my phone system, need help getting functionality back

FreePBX Configuration
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As the title says, disaster struck, my original phone system I hosted earlier has failed… I learned the hard way that LVMs are NOT my friend… but that’s not relevant here…

I rebuilt my phone system, and got basic functionality back with the UCM patched Asterisk 20.18, using the latest FreePBX 17… In/Out local calls works fine, BLF works to monitor my 7941, but I need BLF working for my 7965 to monitor my Polycom phone, which is REQUIRED to be PJSIP, I guess… here’s a pastebin with the various files: Various configs (see file) - FreePBX Pastebin

Yes, I had to modify “Extensions_Override_FreePBX.conf”, there was no custom file to get around it, yes I accept the risks of modifying those files… Gemini was very little help, so I’m REALLY hoping someone HERE would be able to help… unless everyone’s done with me…

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Good, I’ve already gone over it though, and it doesn’t help me do what I need to do, if anything, the instructions are pretty unclear…

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Just chucked your message into Claude and asked it to check over. Used Fable and it’s given this response:

In EXTENSIONS_OVERRIDE_FREEPBX.CONF, change just the 2861 lines to use PJSIP:

[extensions]
exten => 2861,0,Goto(call-extension,${EXTEN},1)
same => hint,PJSIP/2861,PJSIP/2861

[ext-local]
exten => 2861,hint,PJSIP/2861,PJSIP/2861

(Leave 2001/2452/2271 on SIP/ since those are genuinely chan_sip Cisco phones.)

Then remove the orphaned [2861](+,polycom) block from SIP_CUSTOM_POST.CONF since 2861 shouldn’t have a chan_sip peer at all.

Give it a try if it errors let me know. Seems like a really annoying issue though :rofl:

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AI’s not a good tool to use, sadly… I already tried using AI, but it’s not been helpful for this, I also already tried the PJSIP thing from Gemini, and it’s not worked… I’d like to ask that people refrain from using AI to try to solve any issues… but still worth a try, I guess…

Update: Didn’t work…

Update 2: It sort of works now, I put the glob [ext-local] in sip_custom.conf, and now the extension can be monitored, but the BLF hints aren’t being sent to the Poly phone, it doesn’t matter too terribly much? Would like for it to work, and for the Cisco phone to be able to understand the Poly hints, but it’s more or less optional, I’m sure I’m missing a subscribe somewhere for the phones to register when 2452 goes in-use, I’m sure I’ll still need help down the road, I just need to get my wits about me right now…

Update 3: Oh yes, I need paging to work… here’s what I have in the same sip custom file:

[paging]
; Only allow one page at a time
exten => 7255,1,Set(GROUP()=paging)
same => next,ExecIf($[${GROUP_COUNT(paging)} != 1]?Busy(10))
; Multicast audio mode is enabled using the m() option
same => next,SIPCiscoPage(2452&2001,oad(From ${CALLERID(number)}))
same => next,Hangup(normal_clearing)

This is the same configuration (I believe) that worked before, I wish I knew, but all the files from the previous build are gone…

Update 4: Now my Polycom phone won’t register, getting a 401 from SIP/2.0, all the values in the advanced and other pages in the extension are the defaults, and all the values in the phone’s web config match… The exact error I’m getting:

[2026-08-19 12:39:02] NOTICE[2228]: res_pjsip/pjsip_distributor.c:688 log_failed_request: Request 'REGISTER' from '"Polycom" <sip:[email protected]>' failed for '192.168.0.41:5060' (callid: 1756d90914d884ce70f429cadfc1db42) - Failed to authenticate
<--- Transmitting SIP response (501 bytes) to UDP:192.168.0.41:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.41;rport=5060;received=192.168.0.41;branch=z9hG4bKd99ac5efD90625E0
Call-ID: 1756d90914d884ce70f429cadfc1db42
From: "Polycom" <sip:[email protected]>;tag=F1D8D6DF-5A7436F8
To: <sip:[email protected]>;tag=z9hG4bKd99ac5efD90625E0
CSeq: 4 REGISTER
WWW-Authenticate: Digest realm="asterisk",nonce="1787168342/3351fa7e53697506b06b657f28e1da11",opaque="2c2ad73351395198",algorithm=MD5,qop="auth"
Server: FPBX-17.0.32(20.18.0)
Content-Length:  0
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It would seem I’m always having the weird issues that there’s NO information about on Google… this is becoming quite stressful, this is why I didn’t want to rebuild my phone system and why I prayed that it’d be recoverable… but I’d be doing myself and a friend of mine a disservice by giving up now… but given the hesitance of others in regards to helping me… I think I might just HAVE to give up and let this thread expire…

Update about Polycom 401: It was an arbitrary issue, after many reboots/restarts and fiddling and removing and re-adding the configuration stuff for hints in the .conf files, it randomly started working, dunno what did it, but it’s all back now, so now back to getting the Polycom to show extension 2452 as in use when it goes off hook rather than when a call is places, and the other stuff…

Update 2: My <transportLayerProtocol> was set to 2 (UDP), needed to be 1 (TCP), so now my Polycom shows the phone as busy when off-hook, properly… may still need help down the road, but I’m getting closer to where I was… dunno how the TLP option got set wrong, but… cool, I guess… Oh yeah, still need help with the SIPCiscoPage thing…

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Sorry, what is LVM ?

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Logical Volume Manager, Linux’s junky system to expand the file system size to the entire partition… If I recall, to make the system run better…

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Oh, yea that. I had to fight with that when I moved a Windows server from a VM cloud platform over to a ProxMox VM platform and expand the drive space.

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Yeah, it’s painful to deal with… but that situation’s been “handled”, by which I mean I just rebuilt the phone system from scratch and just abandoned all efforts on trying to recover the files, given the chances of doing so were Slim to none, and Slim just drowned…

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You have to patch Asterisk for BLF to work on your Cisco. Cisco uses a non-standard code…

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I’ve already handled that, now I just need to get things working by faffing about with the configs, but it’s horribly unclear what goes where… among other things…

Update: And I’m back to having to force the Polycom phone and the FPBX server to cooperate and quit fighting each other, but now nothing I’m doing is working…

Update 2:

[2861](+)
subscribe=2452@ext-local

This was my gremlin, I guess… Well, on the bright side, I don’t need it, given that the phone monitors busy stuff out of the box without any fancy code or whatnot, so… shrugs I will still need support down the road, I know I need to get the paging to work, since Cisco phones don’t support the native paging from FPBX, and the patch aims to fix that, but I’d like to know how to get it to work again…

Update 3: Editing this message instead of making new posts such that I don’t clog up notifications for those subscribed to this thread. Codec lock didn’t work earlier because the string for the feature code wasn’t constructed correctly; string changed to

; ----- Feature codes, lock codec to prevent video
exten => _*.,1,Set(__SIP_CODEC=g722)
same => next,Goto(from-internal,${EXTEN},3)

so the codec lock actually works, outbound codec lock disabled as I want the best audio quality that it can get, if possible, else re-enable codec lock to support high quality calls with “G.722” codec, will take suggestions on codecs to use to encourage high quality calls where possible; i.e. between HQ-enabled devices like an iPhone and the phone system. Assistance still requested for the SIPCiscoPage application (not explicitly listed anywhere in FreePBX, but exists, feature doesn’t presently work

Update 4: Heed this warning to all who create a FreePBX system with the UCM patch: Make sure everything is under the right context. The paging thing MUST be under the
[from-internal-custom] context… AND NOWHERE ELSE… So let this be a lesson… put things in the right place… It is now late tonight, I can’t codec lock specific extensions, which is suckage, but… oh well, I’ll just do it another way…

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Well, I guess my system’s back to the way it should be, so assistance on restoring it no longer needed, though I still need help with other things, but… I’ll have to make separate topics for those things…