@covici wrote:
Hi. I just installed freepbx under DEbian buster with asterisk 16.6. After install I went to the html/admin URL and it asked me for admin username, password and email address and I said set up system. Now, there are two choices, ucp which when I click the browser says not found, and operator panel which waits indefinitely. I am using php 7.3. How do I even troubleshoot this one?
Thanks in advance for any suggestions.
Posts: 6
Participants: 2
@RobS wrote:
I can’t seem to get DTMF to work on calls the way I need to.
I have an A400 card in a PBXAct box with 8 FXS and 4 FXO. When on a call FreePBX not passing the DTMF on the FXS lines. I get a slight chirp sometimes but most of the time it’s just a “thunk” or click.
Our IP phones are Sangoma S705’s and S505’s. They show the same issue as on the FXO lines (old conference phones no one wants to give up) which can make dialing up a conference a royal pain when you can’t enter the conference ID#. Like I said I get a slight chirp sometimes but most of the time it’s just a “thunk” or click.
What setting(s) do I need to concern myself with to get DTMF to work while on call instead of, I’m guessing, being interpreted?
Right now I have “SIP DTMF Signaling” set to “auto” on the FreePBX Advanced Settings page.
Posts: 1
Participants: 1
@jsmith wrote:
Please let me first apologize for the impact that yesterday’s outage of Sangoma infrastructure had on your business. We understand the trust you have placed in us to perform reliably, and we did not live up to those expectations yesterday. For that we are truly sorry, and commit to using this event as a learning experience to improve our service to you.
In summary, yesterday we suffered a major outage of our internal database cluster that powers many of the public facing services we provide. This outage had many impacts on our customers including the Sangoma Portal, Support Ticketing systems, FreePBX and PBXact updates, and many other business transactions with Sangoma. While many of Sangoma’s Cloud Services customers were unaffected, the event did create a service disruption for some SIPStation customers, as well as an outage on our FAXStation, VPN and SMS services.
As to what specifically happened, at approximately 2:08PM CST yesterday, one of our production database clusters crashed, and indicated data corruption as a result of the crash. Subsequently we were unable to simply restore it to a working state. Our internal infrastructure teams worked through the night to restore the data, redeploy the database (in a manner that would not be susceptible to another similar problem), and return the services to normal operation. All services were fully restored by 6AM CST today.
In addition to the outage, there was also a lack of effective communication and support for our customers and partners during the event. While there was not a lot of definitive information we could provide during the early portion of the outage, we should have done a better job of keeping you informed about the scope of the outage, and our progress. Some of you have expressed your displeasure with our lack of communication, and we certainly heard you, and commit to doing better in the future. As we continue our investigation into the root cause of this event, we will provide more details over the next few days, along with a plan for corrective actions.
We know that you depend on Sangoma products and services to run your business, and our goal is to exceed your expectations. Yesterday, we fell short of that goal, and for that we apologize. We take the availability and reliability of our services and infrastructure very seriously, and will take steps to prevent this in the future. Thank you for your patience, understanding and continued support.
Posts: 4
Participants: 2
@tensi0n519 wrote:
Hello Everyone,
I am new to the FreePBX system and I plan on using it to replace a Cisco 2811 that I’ve been only using now as my pbx around my office. I would like to use a few on my existing 7975 phones in my shop area because they are pretty durable and cheap. I plan on getting some nicer (and probably way easy to configure) phones once I’m able to make sure the system works for me.Anyways, I have sort of been able to get one of the 7975 phones talking to FreePBX but it is still not working correctly. I have followed the guide posted here: github (dot) com/chan-sccp/chan-sccp/wiki/FreePBX
The issue I am having is that the display keeps alternating a message that states “Configuring Unified CM List” and then goes back to “Asterisk Connect.” It does this over and over and it makes the phone completely unusable. I have no idea what to do from here. I did have to manually input the TFTP IP address on the phone to get it to connect to the FreePBX. I’m not sure if that’s normal or not since with the Cisco 2811, the phones just connect automatically.
Below I will post my message log and the SCCP. Thank you
Here is a few of the message logs I am getting:
Feb 6 04:27:14 freepbx systemd-logind: New session 66 of user root.
Feb 6 04:27:14 freepbx in.tftpd[5589]: RRQ from 10.10.20.102 filename SEP0023EB51E80E.cnf.xml
Feb 6 04:27:14 freepbx in.tftpd[5589]: Client 10.10.20.102 finished SEP0023EB51E80E.cnf.xml
Feb 6 04:27:24 freepbx in.tftpd[5638]: RRQ from 10.10.20.102 filename CTLSEP0023EB51E80E.tlv
Feb 6 04:27:24 freepbx in.tftpd[5638]: Client 10.10.20.102 File not found CTLSEP0023EB51E80E.tlv
Feb 6 04:27:24 freepbx in.tftpd[5639]: RRQ from 10.10.20.102 filename ITLSEP0023EB51E80E.tlv
Feb 6 04:27:24 freepbx in.tftpd[5639]: Client 10.10.20.102 File not found ITLSEP0023EB51E80E.tlv
Feb 6 04:27:24 freepbx in.tftpd[5640]: RRQ from 10.10.20.102 filename ITLFile.tlv
Feb 6 04:27:24 freepbx in.tftpd[5640]: Client 10.10.20.102 File not found ITLFile.tlv
Feb 6 04:27:25 freepbx in.tftpd[5642]: RRQ from 10.10.20.102 filename SEP0023EB51E80E.cnf.xml
Feb 6 04:27:25 freepbx in.tftpd[5642]: Client 10.10.20.102 finished SEP0023EB51E80E.cnf.xmlSCCP Config:
;=========================================================================================
;
; general definitions
;
;=========================================================================================
[general]
debug = core, config, action, socket, device, line, channel
servername = Asterisk
keepalive = 60
context = default
dateformat = D.M.Y
bindaddr = 0.0.0.0
port = 2000
disallow=all
allow=alaw
allow=ulaw
allow=g729
firstdigittimeout = 16
digittimeout = 8
autoanswer_ring_time = 1
autoanswer_tone = Zip
remotehangup_tone = Zip
transfer=on
transfer_tone = 0
transfer_on_hangup = off
dnd_tone = 0x0
callwaiting_tone = Call Waiting Tone
musicclass=default
language=en
deny=0.0.0.0/0.0.0.0
permit=internal ; ‘internal’ is automatically converted to these private cidr address:
; 127.0.0.0/255.0.0.0, 10.0.0.0/255.0.0.0,
; 172.0.0.0/255.224.0.0, 192.168.0.0/255.255.0.0
;permit=10.10.20.0/255.255.255.0
localnet = internal ; (MULTI-ENTRY) All RFC 1918 addresses are local networks, example ‘192.168.1.0/255.255.255.0’
;externip = 77.44.22.33 ; External IP Address of the firewall, required in case the PBX is running on a seperate host behind it. IP Address that we’re going to notify in RTP media stream as the pbx source address.
dndFeature = on
sccp_tos = 0x68
sccp_cos = 4
audio_tos = 0xB8
audio_cos = 6
video_tos = 0x88
video_cos = 5
echocancel = on
silencesuppression = off
private = on
callanswerorder=oldestfirst
pickup_modeanswer = on
hotline_enabled=yes ;can devices without configuration register
hotline_context=default ; context for hotline
hotline_extension=111 ; extension will be dialed on offHook;=========================================================================================
;
; actual definitions
;
;=========================================================================================
[SEP0023EB51E80E]
description = Phone Number One
addon = 7914
devicetype = 7975
park = off
button = speeddial,Helpdesk, 98112, 98112@hints ; Add SpeedDial to Helpdesk
button = line, 98011,default ; Assign Line 98011 to Device and use this as default line
button = empty ; Assign an Empty Button
button = line, 98012 ; Assign Line 98012 to Device
button = speeddial,Phone 2 Line 1, 98021, 98021@hints ; Add SpeedDial to Phone Number Two Line 1 (button labels allow special characters like ‘é’)
cfwdall = off
type = device
keepalive = 60
;tzoffset = +2
transfer = on
park = on
cfwdall = off
cfwdbusy = off
cfwdnoanswer = off
deny=0.0.0.0/0.0.0.0
permit=10.10.20.102/255.255.255.255
dndFeature = on
dnd = off
directrtp=off
earlyrtp = progress
private = on
mwilamp = on
mwioncall = off
setvar=testvar=value
cfwdall = on[98011]
id = 2102
type = line
pin = 2102
label = Phone 1 Line 1
description = Line 98011
mailbox = 2102
cid_name = MY CID
cid_num = 98011
accountcode=79011
callgroup=1,3-4
pickupgroup=1,3-5
directed_pickup = on
directed_pickup_context = “”
pickup_modeanswer = on
;amaflags =
context = default
incominglimit = 2
transfer = on
vmnum = 600
meetme = on
meetmeopts = qxd
meetmenum = 700
trnsfvm = 1000
secondary_dialtone_digits = 9
secondary_dialtone_tone = Outside Dial Tone
musicclass=default
language=en
echocancel = on
silencesuppression = off
setvar=testvar2=my value
dnd = reject
parkinglot = myparkspace[98012]
id = 1001
type = line
pin = 4356
label = Phone 1 Line 2
description = Line 98012
mailbox = 10012
cid_name = MY LINE 2
cid_num = 98012
accountcode=79002
callgroup=1,4-9
pickupgroup=1,3-9
directed_pickup = on
directed_pickup_context = “another”
pickup_modeanswer = on
echocancel = off
context = default
incominglimit = 2
transfer = on
vmnum = 600
trnsfvm = 1000
secondary_dialtone_digits = 9
secondary_dialtone_tone = Outside Dial Tone
musicclass=default
language=en
echocancel = on
silencesuppression = off
silencesuppression = on
dnd = silent
Posts: 2
Participants: 2
@Croissantisokay wrote:
Hello gurus,
i’m trying to log with telnet into my PBX serveur.
Manager.conf :
i try to connect this way :
telnet MY.SERVEUR.IP 5038
or
telnet MY.SERVEUR.DOMAIN 5038
but no luck i got timedout.
Any ideas?
Posts: 2
Participants: 2
@MCJ wrote:
We currently run Cisco’s call manager across 3 buildings and roughly 300 phones. I hate Cisco because it is so convoluted and expensive. Mitel was so much better. Besides the point…
I have used FreePBX in the past in a semi-enterprise setting and it worked fine. The part that I am having issues with is, FreePBX would be new to our techs and myself and it seems like support is not overly available, especially in the rural area that this would be deployed in. I think trying to figure out e911, trunk lines, general configuration would be a nightmare without someone initially on-site.
I had a Digium/Switchvox rep come out and size a system for us and the initial and ongoing cost was similar to Cisco. Expensive. The reason I am looking at something else is that our Cisco system is dated.
Posts: 3
Participants: 3
@Cooltronics wrote:
I am having an issue where it tells me UCP, RESTful API, and LetsEncrypt under Port Management are all disabled. When I try to enable them to their defaults and click “update now” nothing at all happens. Tried 3 different web browsers, both http & https and different Pc’s just in case.
Any ideas?
Posts: 1
Participants: 1
@carlyle705 wrote:
Really need help on this.
Call forwarding setup:
- Misc Destination to my cellphone
- In my extension advanced tab, scroll all the way down, set Optional Destinations to Misc Destination
- I have a inbound call route for my phone number to directly call my extension.
When I call my number, the call will be transferred to my cellphone, but no audio.
Any help would be appreciated.
Thanks!!
Posts: 3
Participants: 2
@AmidouFlorian92 wrote:
Hi all
I’m searching a way to configure an SMS gateway on my freepbx server.
I want to be able to send and receive automatic SMS on my server.
Is there a solution to do It using the CLI or what solution can I install to have an SMS module working with freepbx ?
Thanks
Posts: 3
Participants: 2
@sentinelace wrote:
I am trying to upload the current FW. I use slot 1, 0.00 and it doens’t update. What is the proper process for custom FW with yealink?
Posts: 2
Participants: 2
@PYoung wrote:
I’m having a strange issue which I can replicate in any FreePBX easily.
Callers is sent to a Queue.
Call is answered by the agent.The agent then transfers the call to a ring group.
This ring group has 2 extensions with # at the end so the follow-me can work. (Ex : 225# and 228#)
The no answer destination of this ring Group is Voicemail/Unavailable Message Ext 307.
If either 225 or 228 have their follow-me enabled, everything works as expected.
If one of these extensions has their follow-me disabled, the call will end up in the voicemail of that extension instead of Voicemail 307.I have no idea why!
I have recreated a Queue with default settings or changing one setting at a time, I get this behaviour everytime. When I call the Ring group directly I cannot replicate this issue! So there is something between the queue/ring group and # option that I can’t seem to put my finger on.
Posts: 1
Participants: 1
@arjones5 wrote:
Migrating from 14.0.13.24 directly to current on new hardware due to hardware failure. I’m left to migrate it all by hand piece by piece for a small office (3 trunks, 10 phones). I, or so I thought, was complete with all the settings, firewall changes, etc. Outbound calls work just fine, inbound route directly to voicemail every time. The logs are completely useless or they’re buried.
This is the output during 4 inbound call attempts and one outbound success:
https://pastebin.freepbx.org/view/e98144efPlease halp.
Posts: 12
Participants: 3
@tensi0n519 wrote:
I’ve just set up SIPSTATION on my freepbx system and I am able to send and receive calls. I don’t have any prefixes setup and I am able to just dial the number which ultimately connects whoever I’m calling; in this case my cell phone. The issue I am having is that after I dial my outgoing number (my cell phone), I just hear silence. There is no tone as the phone connects to the outside, just silence. Hopefully what I am saying makes sense. Thank you.
Posts: 10
Participants: 3
@mrjoe wrote:
I’m using 15.0.16.42 with Asterisk 16.6.2
I get the above error when trying to “Apply Config”
When I run that I get the following error: [Exception (404)]
Unable to locate the FreePBX BMO Class 'Pjsip’A required module might be disabled or uninstalled. Recommended steps (run from the CLI): 1) fwconsole ma install pjsip 2) fwconsole ma enable pjsipWhen I run the first one I get the following error: Unable to install module pjsip:
- Cannot find module
When trying to run the second I get: The following error(s) occured:
- Module pjsip cannot be enabled
Does anyone know what could be the problem?
It started after I updated the Modules.
Posts: 1
Participants: 1
@assos40 wrote:
Hi,
Using FreePBX 14.0.13.23
I am trying to add a new system recording.
The filename is ‘+2105897854’ but a get an error on submiting
“Undefined index: playback
File:/var/www/html/admin/modules/recordings/Recordings.class.php:330”
If i remove the ‘+’ sign then everything is working properly.
I need the ‘+’ sign in front of the filename.
Any idea how to fix this.
Thank you
Posts: 1
Participants: 1
@SiP1701 wrote:
I have 2 options for full backup on my PBXact appliance
Full backup
Pbxact full backupComparing the 2 PBXact seems to be backing up much more.
Will choosing PBXact be the best choice when running a full backup prior to making any updates?
Posts: 1
Participants: 1
@faisalkhan wrote:
Hi guys,
I need little help in configuring callback
how to setup callback and how it works and what are the best locations where we use this feature
Posts: 1
Participants: 1
@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: 6
Participants: 3
@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
@cal wrote:
Hi all,
I am running asterisk 13.29.2 and Current PBX Version:14.0.13.24.
I have it so when a call comes through it goes to a ring group which the clients would normally put on park.
I currently have it set so ring group MOH is set to ring and the parked calls should have a stream going. Ive double and triple checked the stream which is fine. i even set it to default and still no music.
i saw in another thread that parking uses the same channel as where the call came from i.e ring group. Is this my problem? Do i have to stream the music through the ring group first in order to get MOH in parking?
The clients want the ring groups to have ringing though not music which is a dilema.
No good a custom configurations so if anyone has a solution would be so greatful.
Im new to Freepbx and asteriskThankyou in advance
Posts: 1
Participants: 1
