@Sneffy9 wrote:
Hello everyone! can some one please tell me the name of any ATA that can support more than 2 FXS ports?
i have 12 analog phones that i want to use with one new freepbx. i already tested them on a Cisco spa112, and they worked fine.
Posts: 6
Participants: 3
@Bidhya wrote:
Hello everyone,
I want to keep the record of phone number of the caller. But the Caller number is not stored in cdr table, instead the User Id that I used to register the PSTN in Linksys Sipura ATA is stored in the Caller Number. I tried to set the caller number using asterisk dial plan also (like Set(CALLERID(name)=${CALLERID(num)})) but it also stored the UserId not the caller phone number. Any insight would be greatly appreciated.
Posts: 1
Participants: 1
@tty0744 wrote:
FreePBX 13.0.192.19
Asterisk 13.18.0
SHMZ release 6.6 (Final)I have a remote extension that registers successfully with our pbx. Internal users can contact the remote extension, leave vm’s, etc., with no issues at all. However, the remote extension can contact internal users, but as soon as they pick up, the call drops. The same happens when dialing external destinations (i.e. xxx-xxx-xxxx).
What am I missing here?
Posts: 3
Participants: 3
@mvogel4949 wrote:
I’m seeing the following error in my backup of the cdr:
mysqldump: Error 1194: Table ‘cel’ is marked as crashed and should be repaired when dumping table
celat row: 31525366I have gone into the system and run both
mysqlcheck -u root -p asteriskcdrdb --auto-repair -c -o
it reports the cel as ok but I’m still seeing the error in my backup. Any ideas?
Posts: 1
Participants: 1
@faisalkhan wrote:
Hi Guys,
I need to work a way around to add my own php agi script to interact with whole dialplan.
How can I add this script.
Posts: 2
Participants: 2
@chris43 wrote:
Hi
Running Freepbx 12
had a power failure today and now MYSQL wont start.
in the past below would fix my issue
rm /var/lib/mysql/mysql.sock
service mysqld start
mysqlcheck --repair --all-databasesafter removing mysql.sock and tring to start i get the following error
Timeout error occurred trying to start MySQL Daemon.
i have tried rebooting as well
any help appreacated
Posts: 2
Participants: 1
@mcisar wrote:
Is there any way of doing a consolidated tollfree pattern match for caller ID in an inbound route? Thusfar I’ve used _800XXXXXXX but then need another for 844, 855, 866, 877, 888 so that ends up being 5 routes… times each DID I want to apply it to. Obviously pretty cumbersome. Obviously _8[045678][045678]XXXXXXX is going to catch some that I don’t want to catch… is there anything similar to _8[00|44|55|66|77|88]XXXXXXX or is that just a pipe-dream?
Cheers!
Posts: 2
Participants: 2
@mcisar wrote:
Is there any way to programmatically terminate a call in progress?
Client has a situation where the nightshift guys with far separated offices will sometimes handsfree one another so they can chat while they work. This is no problem and is in fact encouraged by management because it not only keeps everybody from going insane but it actually does improve their workflow
The issue is when you’ve had a handsfree call going for 9 or 10 hours you are almost guaranteed to forget to terminate it at the end of your shift.
We’d just like to have a script that we can run at shift-change to drop the calls.
Cheers
Posts: 2
Participants: 2
@Ds5v50 wrote:
Is there any GUI login logs, there is more then 1 admin to a FreePBX install and I would like to be able to see which ones have logged in to the dashboard. I’ve done a bit of searching and have come up empty. If GUI dashboard logins are logged somewhere, where would that be. If it is not logged then how do I make a feature request?
TIA.
Posts: 3
Participants: 3
@matthewljensen wrote:
I installed the default certificate that was auto-generated into the https setup. I shouldn’t have messed around with this, because currently, my server is completely behind a firewall anyway. But that’s what I did. I can no longer access the server through chrome. It gives me the error NET::ERR_CERT_INVALID. I accessed the server through edge, and that works. I created a new self-signed certificate, and imported that, but I still can’t access through chrome. Right now, I just want to turn off https access completely, but I can’t figure out how to de-import that certificate. I can’t disable the https admin port in system admin port management. I don’t know how to disable apache. What should I do?
Posts: 4
Participants: 2
@Red wrote:
So here is my issue. I have a remote phone at home connected to openvpn to the phone server at work. it receives ip from the vpn server, but won’t register. This phone is attached to the internet by a netgear nighthawk(dhcp) and a comcast cable modem(bridge mode). Is there anything i need to do, to the netgear, in order to allow the phone to communicate with the phone server at work. All NAT issues have been addressed and appropriate ports are open on the work firewall.
Posts: 1
Participants: 1
@mvogel4949 wrote:
I have two systems connected using IAX2 trunks and the proper outbound routes. Different extension numbers. If my main queue is on system 1 can I have extensions on system 2 in the main queue as static agents?
Posts: 1
Participants: 1
@sentinelace wrote:
does anyone know if you can hit an option to call back the caller who left a voicemail?
Posts: 2
Participants: 2
@theitdept02 wrote:
The install.php file is empty for the Motif module at the Github site.
Should this not be empty? I tried installing the module, but it created no MySQL tables.
Posts: 2
Participants: 2
@danielfite wrote:
Hello
i have some call routing questions. Ill try to be as brief as possible
Current setup
business hours time group to time conditions, day ivr, press 2 tech queue, manually assigned agents
then i have the same setup for 7 Nights pushing to different IVRs for each night of the week pushing to different ques for each after weeknight.
IVR- after hours monday>>>>press 2 >>> queue after hours monday>>>static agent 702>>>follow me 702, 802 (zoiper android sip client)
the idea here is to be able to setup each of our after hours techs to receive calls on there mobiles through follow me with the client sitting in the queue until they get the call. i have so many queues and ivrs and time conditions because i just want to be able to set the queue agents at the beginning of the week and have the pbx route the calls all week ( and not have to depend on someone to remember to adjust the queue agents every day, especially on Saturday and Sunday this is an issue)so i get to setting this all up and i run into one issue. The incoming route is being sent to a time condition and that time condition, daytime, has a match of “IVR day” and non match of “IVR after hours monday.” with this setup i dont see a way to get the system to ever go to the IVR tuesday, wednesday, thursday, etc.
Is there a better way to set this up so i can route calls to a queue on different days with different agents or am i just missing something in the call flow??
Just to clarify if i just setup IVR DAY QUEUE DAY and IVR AFT HOURS, QUE AFT HOURS everything works but i have to switch out the QUE members daily based on who is on call that evening.
Posts: 1
Participants: 1
@jeff.wong wrote:
Is there a way to link voicemail email address (Applications -> Extensions -> Voicemail) in User Management?
I’m manage user with Active Directory, once enter the value for Extension Link Attribute, user “primary linked extension” will be linked automatic.
I would like to link user’s email address for voicemail, whether freePBX have this feature similar like linking extension?
Posts: 3
Participants: 2
@sentinelace wrote:
I normally use sonicwall which we have had great success. I have a customer who uses pfsense. Have you guys ever setup freepbx with this? Any issues? We enable consistent NAT on the sonicwalls and never had an issue. Just curious what challenges we may run into?
Posts: 2
Participants: 2
@nmarques wrote:
I was checking out my Asterisk logs and noticed hundreds of lines like the ones below… what do they mean?
[2017-12-21 21:01:41] NOTICE[5198] res_pjsip_exten_state.c: Endpoint '110' state subscription failed: Extension '*992*1*110' does not exist in context 'from-internal' or has no associated hint [2017-12-21 21:03:00] NOTICE[9720] res_pjsip/pjsip_distributor.c: Request 'INVITE' from '"1" <sip:1@192.168.10.5>' failed for '66.70.186.99:5079' (callid: c85271006800dd6cfd0caee471179f7f) - No matching endpoint found [2017-12-21 21:03:51] WARNING[18271] chan_sip.c: Purely numeric hostname (200), and not a peer--rejecting! [2017-12-21 21:03:51] NOTICE[22160] res_pjsip_exten_state.c: Endpoint '200' state subscription failed: Extension '*992*1*200' does not exist in context 'from-internal' or has no associated hint [2017-12-21 21:04:00] WARNING[18271] chan_sip.c: Purely numeric hostname (108), and not a peer--rejecting! [2017-12-21 21:04:00] WARNING[18271] chan_sip.c: Purely numeric hostname (108), and not a peer--rejecting! [2017-12-21 21:04:00] NOTICE[5198] res_pjsip_exten_state.c: Endpoint '108' state subscription failed: Extension '*992*1*108' does not exist in context 'from-internal' or has no associated hint [2017-12-21 21:04:39] NOTICE[27343] res_pjsip_exten_state.c: Endpoint '107' state subscription failed: Extension '*992*1*107' does not exist in context 'from-internal' or has no associated hint [2017-12-21 21:04:48] NOTICE[22160] res_pjsip_exten_state.c: Endpoint '100' state subscription failed: Extension '*992*1*100-1' does not exist in context 'from-internal' or has no associated hint [2017-12-21 21:05:19] NOTICE[22160] res_pjsip_exten_state.c: Endpoint '106' state subscription failed: Extension '*992*1*106' does not exist in context 'from-internal' or has no associated hint [2017-12-21 21:07:01] NOTICE[9720] res_pjsip/pjsip_distributor.c: Request 'OPTIONS' from '<sip:n@n>' failed for '2.105.13.142:21281' (callid: 5) - No matching endpoint found
Posts: 1
Participants: 1
@mvogel4949 wrote:
Asterisks 13.18
FreePBX 13-22System was working perfectly then asterisk stopped. I go command line and find this:
/usr/sbin/safe_asterisk: line 171: 22259 Segmentation fault (core dumped) nice -n $PRIORITY “${ASTSBINDIR}/asterisk” -f ${CLIARGS} ${ASTARGS} > /dev/${TTY} 2>&1 < /dev/${TTY}
/usr/sbin/safe_asterisk: line 171: 22526 Segmentation fault (core dumped) nice -n $PRIORITY “${ASTSBINDIR}/asterisk” -f ${CLIARGS} ${ASTARGS} > /dev/${TTY} 2>&1 < /dev/${TTY}
/usr/sbin/safe_asterisk: line 171: 22830 Segmentation fault (core dumped) nice -n $PRIORITY “${ASTSBINDIR}/asterisk” -f ${CLIARGS} ${ASTARGS} > /dev/${TTY} 2>&1 < /dev/${TTY}
/usr/sbin/safe_asterisk: line 171: 23081 Segmentation fault (core dumped) nice -n $PRIORITY “${ASTSBINDIR}/asterisk” -f ${CLIARGS} ${ASTARGS} > /dev/${TTY} 2>&1 < /dev/${TTY}
/usr/sbin/safe_asterisk: line 171: 23383 Segmentation fault (core dumped) nice -n $PRIORITY “${ASTSBINDIR}/asterisk” -f ${CLIARGS} ${ASTARGS} > /dev/${TTY} 2>&1 < /dev/${TTY}
/usr/sbin/safe_asterisk: line 171: 23660 Segmentation fault (core dumped) nice -n $PRIORITY “${ASTSBINDIR}/asterisk” -f ${CLIARGS} ${ASTARGS} > /dev/${TTY} 2>&1 < /dev/${TTY}
/usr/sbin/safe_asterisk: line 171: 23956 Segmentation fault (core dumped) nice -n $PRIORITY “${ASTSBINDIR}/asterisk” -f ${CLIARGS} ${ASTARGS} > /dev/${TTY} 2>&1 < /dev/${TTY}
/usr/sbin/safe_asterisk: line 171: 24222 Segmentation fault (core dumped) nice -n $PRIORITY “${ASTSBINDIR}/asterisk” -f ${CLIARGS} ${ASTARGS} > /dev/${TTY} 2>&1 < /dev/${TTY}
/usr/sbin/safe_asterisk: line 171: 24545 Segmentation fault (core dumped) nice -n $PRIORITY “${ASTSBINDIR}/asterisk” -f ${CLIARGS} ${ASTARGS} > /dev/${TTY} 2>&1 < /dev/${TTY}
/usr/sbin/safe_asterisk: line 171: 24816 Segmentation fault (core dumped) nice -n $PRIORITY “${ASTSBINDIR}/asterisk” -f ${CLIARGS} ${ASTARGS} > /dev/${TTY} 2>&1 < /dev/${TTY}
/usr/sbin/safe_asterisk: line 171: 25092 Segmentation fault (core dumped) nice -n $PRIORITY “${ASTSBINDIR}/asterisk” -f ${CLIARGS} ${ASTARGS} > /dev/${TTY} 2>&1 < /dev/${TTY}
/usr/sbin/safe_asterisk: line 171: 25371 Segmentation fault (core dumped) nice -n $PRIORITY “${ASTSBINDIR}/asterisk” -f ${CLIARGS} ${ASTARGS} > /dev/${TTY} 2>&1 < /dev/${TTY}
/usr/sbin/safe_asterisk: line 171: 25635 Segmentation fault (core dumped) nice -n $PRIORITY “${ASTSBINDIR}/asterisk” -f ${CLIARGS} ${ASTARGS} > /dev/${TTY} 2>&1 < /dev/${TTY}
/usr/sbin/safe_asterisk: line 171: 25920 Segmentation fault (core dumped) nice -n $PRIORITY “${ASTSBINDIR}/asterisk” -f ${CLIARGS} ${ASTARGS} > /dev/${TTY} 2>&1 < /dev/${TTY}
/usr/sbin/safe_asterisk: line 171: 26193 Segmentation fault (core dumped) nice -n $PRIORITY “${ASTSBINDIR}/asterisk” -f ${CLIARGS} ${ASTARGS} > /dev/${TTY} 2>&1 < /dev/${TTY}
/usr/sbin/safe_asterisk: line 171: 26487 Segmentation fault (core dumped) nice -n $PRIORITY “${ASTSBINDIR}/asterisk” -f ${CLIARGS} ${ASTARGS} > /dev/${TTY} 2>&1 < /dev/${TTY}
/usr/sbin/safe_asterisk: line 171: 26757 Segmentation fault (core dumped) nice -n $PRIORITY “${ASTSBINDIR}/asterisk” -f ${CLIARGS} ${ASTARGS} > /dev/${TTY} 2>&1 < /dev/${TTY}
/usr/sbin/safe_asterisk: line 171: 27070 Segmentation fault (core dumped) nice -n $PRIORITY “${ASTSBINDIR}/asterisk” -f ${CLIARGS} ${ASTARGS} > /dev/${TTY} 2>&1 < /dev/${TTY}
/usr/sbin/safe_asterisk: line 171: 27345 Segmentation fault (core dumped) nice -n $PRIORITY “${ASTSBINDIR}/asterisk” -f ${CLIARGS} ${ASTARGS} > /dev/${TTY} 2>&1 < /dev/${TTY}
/usr/sbin/safe_asterisk: line 171: 27969 Segmentation fault (core dumped) nice -n $PRIORITY “${ASTSBINDIR}/asterisk” -f ${CLIARGS} ${ASTARGS} > /dev/${TTY} 2>&1 < /dev/${TTY}
/usr/sbin/safe_asterisk: line 171: 28240 Segmentation fault (core dumped) nice -n $PRIORITY “${ASTSBINDIR}/asterisk” -f ${CLIARGS} ${ASTARGS} > /dev/${TTY} 2>&1 < /dev/${TTY}
/usr/sbin/safe_asterisk: line 171: 28595 Segmentation fault (core dumped) nice -n $PRIORITY “${ASTSBINDIR}/asterisk” -f ${CLIARGS} ${ASTARGS} > /dev/${TTY} 2>&1 < /dev/${TTY}
/usr/sbin/safe_asterisk: line 171: 28861 Segmentation fault (core dumped) nice -n $PRIORITY “${ASTSBINDIR}/asterisk” -f ${CLIARGS} ${ASTARGS} > /dev/${TTY} 2>&1 < /dev/${TTY}
Posts: 3
Participants: 2
@Mric wrote:
Hello,
There is my problem: i have to set up a dev VM to my ipbx centrex. So i need to setup a VM and put all content of “centrex-prod” in “centrex-dev”.
I know this is dirty. But i can’t do otherway.
So here is the centrex-prod :
Linux Centrex-prov 3.2.0-4-amd64 #1 SMP Debian 3.2.54-2 x86_64 GNU/LinuxAnd the centrex-dev:
Linux centrex-dev 4.9.0-4-amd64 #1 SMP Debian 4.9.65-3 x86_64 GNU/LinuxI did setup last version of asterisk into centrex-dev, php7, apache2, then i dowloaded all files from /etc/asterisk/ (prod) and put them into centrex-dev. Same with DB and /var/www/*.
(Centrex-prod have php5, i do not know if that can mess up things)
i did face a lot of access permission and path error, but now i got this error on centrex-dev:
FATAL ERROR
fatal error reading freepbx_settings
Trace Back
/var/www/html/admin/libraries/freepbx_conf.class.php:219 die_freepbx()
[0]: fatal error reading freepbx_settings/var/www/html/admin/bootstrap.php:103 freepbx_conf->__construct()
/etc/asterisk/freepbx.conf:9 require_once()
[0]: /var/www/html/admin/bootstrap.php/var/www/html/administrateur/includes/bootstrap.php:305 include_once()
[0]: /etc/asterisk/freepbx.conf/var/www/html/administrateur/index.php:13 include_once()
[0]: /var/www/html/administrateur/includes/bootstrap.phpAnd im stuck here. Any hint about this error? Is it even possible to achiv what i want with this setup ?
additionnals info :
no log into /var/log/apache2/error.logcat /var/log/asterisk/freepbx_debug give me :
2017-Dec-28 17:08:02 /var/www/html/admin/libraries/utility.functions.php:203
[NOTICE]: Undefined index: DIE_FREEPBX_VERBOSETell me if you need more info.
Thx for help.
Mric
Posts: 1
Participants: 1
The issue is when you’ve had a handsfree call going for 9 or 10 hours you are almost guaranteed to forget to terminate it at the end of your shift.