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Change the path for the web with FreePBX 14.0.13.4
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FreePBX as a Gatekeeper
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FreePBX 14 Asterisk 13 Fail2ban issue
fail2ban-regex --help
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What do you like or not about FreePBX?
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You can already batch uninstall. In Admin -> Module Admin select all modules you don’t want and uninstall them. You can also do the same with
fwconsole ma uninstall xxxxwhere you can specify multiple module raw names. -
This is not a FreePBX complaint, but an Asterisk complaint. And frankly, 50 lines is nothing. I’ve investigated queue issues on busy systems that can run to the 10’s of thousands for a single inbound call. In order for logs to be useful they must be detailed.
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Agreed, it would be nice, but I fear not practical to incorporate in the GUI. If you have provider settings you wish to share with the community, post them here.
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Duplicate PJSIP contacts in CLI
Each PJSIP phone connection is duplicated in ‘pjsip list contacts’ from CLI, two entries per phone. Voice all working OK, web admin Asterisk -> Info -> Chan_PJSip Endpoints shows no duplicates, is this normal?
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Duplicate PJSIP contacts in CLI
Known issue, and it’s cosmetic only. You can ignore until a future Asterisk version resolves.
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What do you like or not about FreePBX?
#1 is already there
#2 i agree with , we may look at something mr gaetz actually proposed at one time…
ill speak with lorne when hes back from vaca last week; imagine a feature code that could toggle some debug file being created - only its relevant data from logs and tcpdump …
#3 i like as well but maybe not a template that needs to integrate with trunks directly but perhaps separate notes module that could hook into other modules and let you know there was a note related to a page and note template that was general and maybe one specific for trunking ?
i’ll wait to see what else comes in on the thread - good dialogue … thanks for starting it 
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FreePBX Gui blank after running updates
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Cisco PAP2 with EndPoint Manager
I have answered my own question. Basefile edit is actually very easy to figure out. Just select your device that you will be doing, and the parameter will be exactly how it is shown in your device. ie. Ring Waveform is inputted into parameters as Ring_Waveform then set your value to whatever is available as an option, ie. Sinusoid
It was able to recognize the parameter and change it to my preferred setup. Next step is to see if I can have the server force the profile rule on to the ATA so that they don’t even need logged in to, but I feel that might not be possible as the ATA needs to be able to talk back to the server. More updates to follow as I progress along.
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FreePBX 14 Asterisk 13 Fail2ban issue
Ya, I’m aware of that. I’m not the best with writing regex so I was hoping somebody had a working file for Asterisk 13. I tried a few online with no success.
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I finally gave up chan_sip for chan_pjsip and it was fine
Confirmed; rtp_ipv6 does not need to be set.
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I finally gave up chan_sip for chan_pjsip and it was fine
I’ve gone ahead and updated the wiki page to state nothing additional is required for RTP.
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FreePBX 14 Asterisk 13 Fail2ban issue
The asterisk jail that comes with Fail2ban .10 works fine for me.
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What do you like or not about FreePBX?
You can’t get away from a debug with low log lines. Just impossible. A call hitting an Inbound Route alone is going to generate at least 25+ lines in the call log and that’s just with almost nothing being enabled on the route. Due to all the dialplan a call can traverse in the PBX the log is going to be at least 50+ lines or more for a single call. Having something like a ring group or followme being checked/used is going to generate even more.
Turning on the pjsip or chan_sip debugs to get SIP messaging for debugs is going to create even more lines.
It doesn’t matter how the debug logging is triggered, CLI, feature code, GUI click. Having less logged for a call makes troubleshooting it or reviewing it more of a PITA then having to go through all the lines.
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FreePBX 14 Asterisk 13 Fail2ban issue
I am currently on Fail2Ban v0.8.14 that comes with the install and having no luck. Can you give me the full version you are running?
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Cisco 7960 Phones - No Text CID on Internal Extensions
I remember trying to get SCCP working when I first got the system, but never did figure out how to do so. (Would have saved a LOT of time/effort had I been able to simply use the SCCP phones already in place rather than flash new phones & then run around like crazy swapping phones when we migrated.) Do you know if it works with the PBXact system/hardware? If so, are there any HOWTOs or detailed directions on how to do so? Finally, would there be any issues with installing it on the system and running both in parallel? (We do have a bunch of Sangoma phones in our system as well, which only work with SIP AFAIK.)
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S500 PC Port issues
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FreePBX 14 Asterisk 13 Fail2ban issue
as I said , .10 
You can get it yourself from
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Cisco PAP2 with EndPoint Manager
So the server cant force provision the rule, which I was pretty much expecting. The profile rule needs to be set so that it can talk back to the server. But other than that, with editing the basefiles, I am able to fully provision the ATA to my standard, with the only change being to the profile rule.
Thank you for the help @dolesec very much appreciated!
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NOTICE[15420]: manager.c:3504 authenticate: 127.0.0.1 failed to authenticate as 'admin'
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