@Slim1717 wrote:
When I try to connect a new SIP trunk, this unknown error comes out, the old 3 trunks work fine, but if I want to add a new one then the error comes out, this error began to appear after reconfiguring our router.
I did not find duplicate trunks
My version: FreePBX 13
Posts: 1
Participants: 1
@uvais wrote:
Hello,
I have installed a new let’s encrypt certificate in certificate manager and have made it default. However, when I come back to system admin–https setup – I try to install the SSL that I created from the drop down and I get the following error. This error comes for any SSL certificate I select and try to install.Apache Config: Error parsing /etc/pki/tls/certs/localhost.crt
Freepbx version: FreePBX 14.0.13.24
Any help would be highly appreciated.
Posts: 1
Participants: 1
@Adalbikerboy wrote:
Hola buena tarde.
Saben tengo el tema curioso de que si no contesto las llamadas a los 2 tonos se cortan, Si las contesto antes de que terminen esos 2 tonos la llamada anda sin problema.
Saben que puede ser o por donde le puedo buscar?
Posts: 1
Participants: 1
@4allbusiness wrote:
Why does module ‘update all’ install diverse ‘not installed’ modules, commercial and not commercial.
I have uninstalled and removed all unwanted modules for me.
Now if I choose ‘check online’ I have 1 update.If I choose ‘Update all’, suddenly several ‘uninstalled modules’ are set as ‘update and enable’
That would be unwanted behavior of the update modules.
Why is this, and can it be fixed, or for sure, set more info on this, that ‘uninstalled modules’ suddenly be installed and enabled.
See screenshot
Posts: 2
Participants: 2
@sthostetler wrote:
Im about to pull my hair out. I have a New Fresh Install of Freepbx 14.0.13.23
Extension 100 Shows Display name plus (DND). Example Karen Reed(DND)
DND is not activated on phone or system. Phone accepts phone calls and dials out. Where is this coming from. When this user calls another extension this shows.
Thanks
Posts: 1
Participants: 1
@CraigT wrote:
Hi all,
Any thoughts on an Intel NUC7PJYH4 as a FreePBX server. I know that dicko has stated elsewhere that he only uses i5’s in this format, but the above box is a quad core Pentium silver cpu(J5005) 1.5 to 2.3 GHz and it would be fitted with 8Gb ram and 120Gb SSD. It is to replace a Celeron box that is struggling.
Setup would be for <10 extns, 1 or 2 voicemail and call flow with a couple of announcements. Small office setup. A true i5(NUC7i5BNH) is double the price wholesale and could be too much for a volunteer site.Comments please…
Posts: 4
Participants: 3
@sentinelace wrote:
WE have a twilio trunk setup as well as 2 other carrier trunks. No matter what, for incoming calls I get call can not be completed. I checked with wireshark and I have a 401 unauthorized error. ACL is set and username and password are correct. I renamed the trunks so that twilio would be first by putting an A in front. Still no go.
Posts: 4
Participants: 2
@Azkey wrote:
Hello Everyone,
I am a beginner in freepbx. i encounter this problem
" Reload failed because retrieve_conf encountered an error: 1"
wheni am clicking the apply button.
Please help me.
Posts: 2
Participants: 2
@brian4 wrote:
I want to direct an unanswered incoming call (office number) to a recorded voicemail message (sorry we are unavailable right now) with the incoming message shared/sent to all extensions.
- I dont want to use an existing extension as that would be a personal message. Virtual/Custom Ext???
- How do I send 1 voice message to multiple extensions? or is it even possible?
Any help would be appreciated
Thanks
Posts: 20
Participants: 8
@vespino wrote:
Yesterday I got a mail from a client saying they couldn’t reach me. I checked this morning and indeed the phones didn’t ring. I noticed an update, so I did an update on my system and reloaded it after which I could be phoned again. But after an hour or so I checked to see if this was still the case, but no. So I thought “this must have to do with changes I had just done”, reverted the changes and reloaded the system. Again some time went by and again calls weren’t coming in. So it seems like something is wrong, but I can’t seem to find what. I only just now noticed this warning:
[2020-01-24 21:28:23] WARNING[17849] res_pjsip_pubsub.c: No registered subscribe handler for event as-feature-event
It’s repeating every 20 seconds or so. Could this be the case, because I’m not sure what it means. Any other way to find out what is causing the issue?
Posts: 4
Participants: 3
@vespino wrote:
When I’m called, people do not hear me while I do hear them. When calling internally I have a similar problem, so I guess this can’t be firewall related? Just to be sure I have checked my firewall settings and the netwerk settings in FreePBX, all seems correct.
Any idea how to troubleshoot?
Posts: 1
Participants: 1
@boubadebf wrote:
the memory of my freepbx fills up quickly I would like a solution to solve this.
Posts: 6
Participants: 4
@amontefiore wrote:
Hey, I’ve successfully added and configured the CRM module to FreePBX (14.0.13.23) and under User Management I’ve related some users with the ‘Link to CRM User’ option. However, I’m not getting anything showing up in Salesforce. Does anyone have any insight into anything I might be missing. E.g. do phone numbers have to match? etc.
Any help would be greatly appreciated.
Thanks
Alan M
Posts: 1
Participants: 1
@MrXirtam wrote:
FreePBX 15.0.16.42
Asterisk Version: 16.6.2I have a VPS hosted FreePBX server, and I am testing out a different SIP trunk provider. Doing so has forced me to change the SIP port to the standard 5060. I started seeing call attempts from non-existent extensions to international numbers. The calls weren’t able to complete, but they are filling my logs and CDR’s with junk. I have disabled my responsive firewall for my PJSIP traffic. I have added my local static IP to the networks tab as trusted for my phones. They register and work just fine. However, watching my SIP logs, I constantly keep seeing invites trying to place calls from extensions I do not have on my PBX. I added the source IP’s to the blacklist in the firewall settings and I have restarted asterisk, and it seems like it hasn’t made a difference. Those same IP’s I have blocked still persist with attempts. Under services, I have changed my SIP protocol to Local only as well. Under SIP settings, I have disabled both Allow Anonymous Inbound SIP Calls and Allow SIP Guests. I’m not sure if I am missing something in the firewall to prevent these attempts.I have my intrusion detection [fail2ban] running and I increased those restrictions too.
Can someone help me identify what I am missing to help tighten security?
Posts: 1
Participants: 1
@RobS wrote:
I have a bunch of remaining 90xxx extensions even after removing Zulu, WebRTC, etc (even reinstalling and making sure it was disabled for all users, then removing)… They will not go away.
How do I finally get rid of them and keep them from polluting my logs?
Posts: 3
Participants: 3
@eekaiser wrote:
Hello Community,
I am new to FreePBX, trying to set up our first server.
We have been trying to solve an issue with Dahdi for a few days now.
Trying to configure Dahdi with Digium wildcard T235. Showing us red alarm no matter what we try.FreePBX distro 14.
[root@freepbx ~]# dahdi_scan
[1]
active=yes
alarms=RED
description=WCTE23X (PCI) Card 0 Span 1
name=WCTE2/0/1
manufacturer=Digium
devicetype=Wildcard TE235 (VPMOCT064)
location=PCI Bus 08 Slot 01
basechan=1
totchans=24
irq=0
type=digital-T1
syncsrc=0
lbo=0 db (CSU)/0-133 feet (DSX-1)
coding_opts=B8ZS,AMI
framing_opts=ESF,D4
coding=B8ZS
framing=ESFSpan 1: WCTE2/0/1 “WCTE23X (PCI) Card 0 Span 1” (MASTER) ESF/B8ZS RED
1 unknown Clear (In use) (EC: VPMOCT064 - INACTIVE)
2 unknown Clear (In use) (EC: VPMOCT064 - INACTIVE)
3 unknown Clear (In use) (EC: VPMOCT064 - INACTIVE)
4 unknown Clear (In use) (EC: VPMOCT064 - INACTIVE)
5 unknown Clear (In use) (EC: VPMOCT064 - INACTIVE)
6 unknown Clear (In use) (EC: VPMOCT064 - INACTIVE)
7 unknown Clear (In use) (EC: VPMOCT064 - INACTIVE)
8 unknown Clear (In use) (EC: VPMOCT064 - INACTIVE)
9 unknown Clear (In use) (EC: VPMOCT064 - INACTIVE)
10 unknown Clear (In use) (EC: VPMOCT064 - INACTIVE)
11 unknown Clear (In use) (EC: VPMOCT064 - INACTIVE)
12 unknown Clear (In use) (EC: VPMOCT064 - INACTIVE)
13 unknown Clear (In use) (EC: VPMOCT064 - INACTIVE)
14 unknown Clear (In use) (EC: VPMOCT064 - INACTIVE)
15 unknown Clear (In use) (EC: VPMOCT064 - INACTIVE)
16 unknown Clear (In use) (EC: VPMOCT064 - INACTIVE)
17 unknown Clear (In use) (EC: VPMOCT064 - INACTIVE)
18 unknown Clear (In use) (EC: VPMOCT064 - INACTIVE)
19 unknown Clear (In use) (EC: VPMOCT064 - INACTIVE)
20 unknown Clear (In use) (EC: VPMOCT064 - INACTIVE)
21 unknown Clear (In use) (EC: VPMOCT064 - INACTIVE)
22 unknown Clear (In use) (EC: VPMOCT064 - INACTIVE)
23 unknown Clear (In use) (EC: VPMOCT064 - INACTIVE)
24 unknown HDLCFCS (In use) (EC: VPMOCT064 - INACTIVE)Can anyone point us in the right direction please.
Posts: 1
Participants: 1
@darkpixel wrote:
We had a perfectly working FreePBX server a few weeks ago, except it was starting to get overloaded.
We purchased a new, larger server, backed everything up, and then restored to the new phone server.We found out very quickly that a “full backup” apparently doesn’t mean a “full backup”.
A lot of settings weren’t restored properly. One example is incoming routes. We had a handful set to ‘detect faxes’. These were magically flipped into ‘legacy mode’ so faxes were routed incorrectly. We had to manually change ~200 phone lines back to ‘yes’ instead of ‘legacy’…so this problem may be related, but I can’t seem to find it.When people call in, it goes to an IVR that asks them to press 1 if they are a new customer or 2 if they are an existing customer. The options are simply for tracking purposes on our end.
Pressing 1 sends them to a queue–and for the purposes of this issue I’m ignoring this option.
Pressing 2 sends them to a ring group.
The ring group consists of a handful of Digium D60 phones.
Dialing that ring group from my internal phone always works. It always rings the phones, and someone always answers.Calling in from a number of different sources (cell phone, home phone, Google Voice, etc…) will sometimes cause the phones to ring, and other times it immediately jumps to the ring group failover destination which is a time condition. If we are ‘in hours’ it goes to the queue I mentioned earlier. If we are ‘outside of hours’ it goes to voicemail.
It seems to be hit-or-miss, but when the phones in the ring group don’t ring, I will sometimes get dumped into the queue, and other times I will get dumped to voicemail.
Looking at the logs, I see:
== Spawn extension (from-internal, 4001, 1) exited non-zero on 'PJSIP/4002-00005f23' == Spawn extension (from-internal, 4001, 1) exited non-zero on 'PJSIP/4003-00005f24' == Spawn extension (from-internal, 4001, 1) exited non-zero on 'PJSIP/4010-00005f25' == Spawn extension (from-internal, 4001, 1) exited non-zero on 'PJSIP/4020-00005f26'If I immediately dial those extensions or the ring group on my desk phone, the call connects and the phones ring.
Each office has a firewall that is NOT running SIP ALG. UDP timeouts are set to 300 seconds, and nothing firewall-related has changed in months. It’s the same config that was working from before move to a new server. Each office connects out over the internet to our phone server. There is no VPN or ‘internal network’ involved in the phones communicating. The only phones that are on the same network as the phone server is the call center. The call center users are the ones in the queue I referred to previously.
Rebooting phones involves does appear to clear up the issue for a few days, then it comes back…but it still doesn’t explain why the queue will frequently send users directly to voicemail when there are agents signed in and ready to take calls…and calls from other desk phones are able to get through to the queue.
Posts: 18
Participants: 6
@hgsupport wrote:
Got a bunch of phones in a ring group for incoming calls. The issue i’m having is if some of those people have a conference call going with 3 or more people. When an new incoming call happens the conference call loses audio. The conference call is still going, but no one can hear anyone.
Posts: 1
Participants: 1
@Area51 wrote:
We are running FreePBX 13.0.197.21 and Asterisk 13.30.0 with spandsp 0.0.6, IAXmodem, Hylafax+ 7.0.0 and Faxpro 13.0.44 on CentOS 6. Long time running machine that does well on iaxmodem/hylafax.
The client wants to eliminate POTS and so added fax boards to their Canon Imagerunner MFP. Setup seems easy enough and it will send a few pages fine but when it comes to 30 page documents, it’s rare to get past 20. I’ve checked the bandwidth, latency and jitter which all well within specs required. Things are so bad I installed a SPA112 and while it does better, it trains down to 12000 baud before it reliably sends data. note the canon fax board is pretty bad and trains down to 4800 most of the time.
I’ve changed the error correction from redundant, fec, and adjusted maxdatagrams trying to find a difference and/or sweet spot, no luck. The vendor got a Canon tech online and proceeded to spend 8 hours “testing” faxes in and out. They found the same thing I’ve been telling them all along. The SPA appears to work better than the Canon Fax Board. Big help!
The last test I performed was to connect an SPA112 to the switch the FreePBX box is running on and viola, it ran perfect with 14400 connects and fast page transfers. So it appears the switch network is faulty somewhere, somehow. I don’t manage that part but they tell me they have prioritized for VoIP… it seems pretty obvious to me.
My question is when you have a “perfect” path, latency under 1ms, zero jitter and 124Mb bandwidth (iperf), can the switch interfere that much? There are four hops between the FreePBX and the problem printer.
Posts: 1
Participants: 1
