@matthewljensen wrote:
Is there any way to track how many minutes a certain user has talked from any extension? Can a user be forced to ‘log in’ to a phone before making calls so that the number of minutes they use can be tracked?
Imagine a school environment where students are trying to make outgoing calls, but since most sip providers charge for outgoing calls, we can’t implement a phone card system. Is there any way to deal with this?
Posts: 1
Participants: 1
@papelini wrote:
Hello,
Can someone help me to setup dialer ID that each user can dial and show their number when dialing externally?
Regards
Posts: 3
Participants: 2
@felymania wrote:
Good evening
I am new to the forum but I follow you for some time.
Use FreePBX for about 2 years with all devices SCCPs, problems with a Cisco model CP 6921, do not can’t activate the call transfer key, someone could help me?ITALIAN VERSION
Buonasera
son nuovo del forum ma vi seguo da tempo.
Utilizzo freepbx da circa 2 anni con tutti apparati sccp, riscontro problemi con un modello Cisco CP 6921, non riesco ad attivare il tasto di trasferimento chiamata, qualcuno potrebbe aiutarmi?
Posts: 1
Participants: 1
@SamannSam wrote:
Hello, i have some questions about FreePBX. Does PBX can alert to some account in social media such as Facebook? If possible, how can i do with it ?
Posts: 6
Participants: 5
@edlentz wrote:
Is there any way to elevate the playback volume on custom audio files when they are played back?
Thanks
Posts: 5
Participants: 3
@avayax wrote:
My plan is to do a yum upgrade, distro upgrades and a major Asterisk version switch on my High Availability machines.
To avoid much downtime I am thinking of doing it all at once.
So perform distro, yum and asterisk major version upgrade on first node FreePBX-A while Node B is in standby, then bring B online, swap and do upgrades on B.
Or should I rather separate the different upgrades and swap nodes after each?
Posts: 1
Participants: 1
@mrmrmrmr1 wrote:
Hi,
I’ve successfully paired my phone with my Asterisk server using chan_mobile and now I can see it connected with “mobile show devices” command:
raspbx*CLI> mobile show devices
ID Address Group Adapter Connected State SMS
Myphone DC:74:XX:07:XX:XX 1 pabx Yes Free NoNow I want to define a trunk using this channel. But I could not figure out how.
My chan_mobile.conf is like:
[general] interval=30 ; Number of seconds between trying to connect to devices. ;;;;;;;;;MyConfig [adapter] id=pabx address=B8:27:XX:7C:XX:XX ;just hidden real MAC [Myphone] address=DC:74:XX:07:XX:XX port=3 context=from-MySipphone adapter=pabx group=1how can I define the trunk on Freepbx GUI ?
as I read in some instructions , I need something similar to below for the conf files but I don’t want to modify conf files on a FReepbx installation. So the better way is to define the trunk on GUI. Am I correct ?[test] exten => _X.,1,Dial(Mobile/Myphone/${EXTEN},45) _X.,n,Hangup [incoming-mobile] exten => s,1,Noop(Accepting mobile call from ${DID}) exten => s,n,Dial(SIP/test)
Posts: 1
Participants: 1
@AIC2000 wrote:
Hello
I’ve spent all day trying to get a new install of Debian 8, with Kamailio and Siremis. I’ve installed from source, tried different versions of everything and although I can install both in under 5 minutes now I’m having trouble getting them just to work out of the box - let alone figuring out how to configure it.
Debian 9 wouldn’t work due to PHP compatability errors in PHP7, so I’m using Debian 8 with php5.
I’ve read not only the official documentation but many other articles too.
What I’m asking is:
Is it possible to setup Kamailio using Siremis so that I can have multiple FreePBX’s sharing one outbound trunk, with Kamaiolio acting as the router?
Is there any relatively clear guide out there that achieves this? I’ve not found anything for literally just simple forwarding of calls to a central outbound trunk from all 3 different FreePBX’s.
Thanks in advance for any pointers!
Posts: 26
Participants: 4
@CPORich wrote:
server running:
freepbx 14.0.1.24
asterisk 13.18.5I cannot remove a local network from the asterisk sip settings page. GUI says blank fields will be ignored, however hitting submit while blank fields are present returns error and cannot save changes.
Posts: 2
Participants: 2
@KnightRider wrote:
I’m currently running FreePBX with Chan-SCCP on an old computer. I would like to move it to our Vmware server. What would the easiest way to go about this be? My main concern is that Chan-SCCP was a lot of work to install and I’d prefer not to have to do it from scratch.
Thanks!
Posts: 1
Participants: 1
@bksales wrote:
Follow Me is set for 0 initial ring time and then to follow for 20 seconds and then hangup. It is only following itself. I see it rings 20 seconds, the pbx claims its hanging up but then the ringing sound changes and it appears to start another call. I let this loop several times and it appears to never hangup.
I think its silly to want the phones to do this but thats what they want and I’m trying to make it work. Any ideas?
Here is what I see in putty:
-- SIP/253-00000516 is ringing -- Nobody picked up in 20000 ms -- SIP/253-00000516 Internal Gosub(crm-hangup,s,1) startit goes through the motions of hanging up and then I see this but the caller hears ringing still
-- Goto (app-blackhole,hangup,1)== Extension Changed 253[ext-local] new state Idle for Notify User 209
== Extension Changed 253[ext-local] new state Idle for Notify User 251
== Extension Changed 253[ext-local] new state Idle for Notify User 206
== Extension Changed 253[ext-local] new state Idle for Notify User 203It rings for another 20 seconds and then checks time conditions and such again, changes the state back to ringing, rings for another 20 seconds, and then repeats the above message(s) and continues to ring. I didn’t go any further than this but i suspect it will ring forever.
Posts: 11
Participants: 2
@aster888 wrote:
Callcounter is set yes in [general] and in peer sections.
I have FreePBX 13.0.192.19 and Asterisk 13.11.2 and default freepbx setup.When peer is online hints work correct. I can see hint change. But when peer goes offline I see hint like it is idle.
What’s wrong and where?`
Posts: 1
Participants: 1
@cdsJerryw wrote:
I just did the update from 10.13.66-17 following the instructions at https://wiki.freepbx.org/display/PPS/Upgrading+from+FreePBX+10.13.66+to+SNG7
When it finally completed however the GUI opens with the “Apply config” button lit up and a critical error "retrieve_conf failed, config not applied
Reload failed because retrieve_conf encountered an error: 255 "Based on another thread I read I ran the following commands in case that helps:
cd /var/lib/asterisk/bin
./retrieve_confHANDLED-ERROR: faxpro250_get_config should exist, but it doesn’t. This is a bug in faxpro250
HANDLED-ERROR: faxpro600_get_config should exist, but it doesn’t. This is a bug in faxpro600
PHP Fatal error: Call to undefined function FreePBX\modules\endpoint_apiApp() i n /var/www/html/admin/modules/restapps/Restapps.class.php on line 1281
Unable to continue. Call to undefined function FreePBX\modules\endpoint_apiApp() in /var/www/html/admin/modules/restapps/Restapps.class.php on line 1281
#0 /var/www/html/admin/libraries/Composer/vendor/filp/whoops/src/Whoops/Run.php( 383): Whoops\Run->handleError(1, ‘Call to undefin…’, ‘/var/www/html/a…’, 128 1)
#1 [internal function]: Whoops\Run->handleShutdown()
#2 {main}Unfortunately, I don’t know what to do next.
Posts: 21
Participants: 3
@psetti1 wrote:
I had a few problems with PHP but seem to have resolved them all, but following this I am not able to log into the GUI. Apache is running, but nothing registers in access_log or error_log after trying to load the webpage. Anyone have an idea on what I’m missing here?
Posts: 2
Participants: 1
@cdsJerryw wrote:
I did the update to 14.0.1.31 on Friday. This morning however I’ve discovered that none of the remote extensions are working. They are Sangoma phones so I went to Endpoint manager to see if I could tell what’s going on. As soon as I select the menu I get:
Exception
Can not write to the LESS cache folder at /var/www/html/admin/modules/endpoint/assets/less/cache. Please run (from the CLI): fwconsole chown::However that didn’t do anything but create another error. > fwconsole chown
No such command ’ fwconsole chown’ Not sure what that was all about. I started a new SSH connection and was able to run fwconsole chown but none of the remote phones are connecting. I can now get into Endpoint manager, but the status is empty.It does tell me to click and submit a ticket, which I did.
Update: The GUI appears to be working now so I’ll call this “solved”. I’m unable to get any remote phones to connect but I’ll put that is a different thread.
Posts: 7
Participants: 2
@munozj wrote:
I have an old Radio Shack Phone Flasher that I’m using as a visual notification of ringing.
And a Ring Group with 3 members and Ring All strategy, ignore busy agents and terminate to a busy signal.
- endpoint 1
- endpoint 2
- flasher
That ring group is assigned to a DID
When someone calls the DID, the flasher indicates the call and they can answer the call on either endpoint. This works great for 1 and 2 simultaneous calls however on the 3rd simultaneous call the flasher rings but we have no way of answering that 3rd call.
Is there a better way of triggering the flasher? Can I limit the Ring Group to only 2 concurrent calls?
Posts: 1
Participants: 1
@luxartsystems wrote:
Hey there,
I am attempting to install chan-sccp-b, and the guide tells me to use git clone to download the file. However, when I use the term “git clone” it spits out “git: command not found.” I am not big on the command line, so I have no clue what to do at this point. I am using the latest FreePBX directly from the FreePBX website (the premade image) if that helps at all.
Thanks,
Scotty
Posts: 3
Participants: 2
@therobust wrote:
This doesnt do trick for me … sorry to reopen this issue.
Here’s the error i receive after upgrading queues module and disabling some commercial modules which were not required:exit: 255
Unable to continue. symlink(): Permission denied in /shared_folded/var/lib/asterisk/bin/retrieve_conf on line 293#0 [internal function]: Whoops\Run->handleError(2, ‘symlink(): Perm…’, ‘/shared_folded/…’, 293, Array)
#1 /shared_folded/var/lib/asterisk/bin/retrieve_conf(293): symlink(’/var/www/html/a…’, ‘/etc/asterisk/s…’)
#2 /shared_folded/var/lib/asterisk/bin/retrieve_conf(213): connectdirs->do_symlink(’/var/www/html/a…’, ‘/etc/asterisk/s…’, ‘etc’, ‘/var/www/html/a…’)
#3 /shared_folded/var/lib/asterisk/bin/retrieve_conf(535): connectdirs->symlink_subdirs(’/var/www/html/a…’)
#4 {main}Kindly help!
Posts: 1
Participants: 1
@Bradbpw wrote:
I have call recording on for all external and internal calls on our system. I want to record all my phone calls automatically, except for phone calls between myself and my business partner. Our assistant has access to our call recordings and we would prefer if she couldn’t listen to recorded calls between us. I know I can manually turn off the recording while on the phone, but it would be ideal if I could list his extension and cell phone number on the PBX and tell it to disable call recording on those calls. Is this possible?
Posts: 4
Participants: 2
