@l0b0 wrote:
Is there a way to have an agent announcement that plays you the selected language that customer chose without having to create two queues?
Posts: 1
Participants: 1
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Agent language announcement
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Sip trunk status rejected – no notification, no re-registraton
@4r7ur wrote:
I have a strange issue that I encounter once in a while. A similar thread was created 7 months ago, but this didn’t really help. In total I’ve set up 8 (pjsip) trunks, 6 with the same provider plus 2 with others. On the dashboard everything looks fine, trunks online: 8. In reports -> Summary status looks “quite” OK, PJSip Endpoints: available 8, unavailable 0, unknown 2 (BTW, what is unknown?). When I look into Reports -> Registries, I see that sometimes some trunks are in status rejected. Not always the same trunks go into rejected and not all trunks of the same provider are affected.
Now the things that bother me:
- Where in the logs can I find an information when and why the trunks switch to rejected? In none of the logfiles (Reports -> asterisk logfile) can I find something that helps
- Why don’t the trunks reconnect automatically? At the moment I help myself by manually disabling one trunk, and reenabling it, thus all other rejected trunks also switch back to registered.
- How can I set up an email notification if one of the trunks turn to rejected?
This really gives me a headache if trunks go offline unnoticed and without obvious reason.
Current Asterisk Version is 13.19.1 (I’ll run an upgrade later), currently no commercial modules are installed. Anyway, the above questions are independent of the asterisk version.
Other individual PJSIP settings:
Permanent Auth Rejection: enabled
Forbidden Retry Interval: 10
Fatal Retry Interval: 0
General Retry Interval: 60
Expiration: 120
Max Retries: 999Thanks for reading & your help,
Artur
Posts: 3
Participants: 2
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Polycom VVX PJSIP
@brianhess wrote:
I have migrated an old Freepbx system to a new one (version 14) and I am trying to switch the extensions from CHAN_SIP to PJSIP. All extensions are working using CHAN_SIP. However when I switch one I cannot get the phone to register. I am getting the following error:
Registration failed User: 116, Error Code:480 Temporarily not available
I am using the EPM and basically used the defaults for both EPM and the PJSIP Extension setup. Any ideas?
Thanks,
->brian
Posts: 2
Participants: 2
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Bulk delete Inbound routes
@tgross wrote:
We have a situation where we had to duplicate extension numbers and 10 digit numbers in the Inbound Route table for a tie-trunk during the rollout of our FreePBX phone system. Now we need to delete those 4 digit inbound routes (about 160 of them) but we have so many it brings the FreePBX to it’s knees.
Is there a way to bulk remove Inbound Routes? Maybe a remove all and bulk add or something like that?
We using version 13 FreePBX.
Posts: 1
Participants: 1
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Network Settings
@rstebih wrote:
When you change any of the network settings on the ‘System Admin -> Network Settings’ screen, where do they get written too?
I accidentally change the ‘Gateway’ ip address and need to manually reset it back as I am now unable to connect via the web interface.
Cheers,
Rudy
Posts: 1
Participants: 1
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Activation Error (see message)
@Stevevr wrote:
Hi
I’ve done a new install and when I try to activate the freepbx I get a red box at the top
"Unable to display activation page. Error returned was:
I received ‘’, I sent {“display”:“sysadmin”,“view”:“activation”,“extdisplay”:false,“persist”:"{“lastrun”:1510639560,“guid”:“05cb6c26-f027-4611-885e-377761768b7e”,“lastrun-timestamp”:1510639572,“sipstation-lastrun”:1501201557,“sstrial-timestamp”:1501201560,“restartoobe”:true,“lastrestart”:1523418586,“restartreason”:false}",“sysid”:"{“pbxid”:“19993232”,“zend”:[“M:SRXNE-D7493-FEB68-ASPZJ”,“M:6CPXE-F3BLX-DWJ9C-GRPLN”],“modules”:{“accountcodepreserve”:“13.0.2”,“announcement”:“13.0.7.1”,“asteriskinfo”:“13.0.7.1”,“backup”:“14.0.6”,“blacklist”:“13.0.14.8”,“builtin”:“2.3.0.2”,“bulkhandler”:“13.0.14.4”,“calendar”:“14.0.2.4”,“callforward”:“14.0.1.3”,“callrecording”:“13.0.11.5”,“callwaiting”:“14.0.1.1”,“campon”:“13.0.4.1”,“cdr”:“14.0.5.14”,“cel”:“14.0.2.4”,“certman”:“13.0.37.1”,“conferences”:“13.0.23.9”,“configedit”:“13.0.7.1”,“core”:“14.0.6”,“customappsreg”:“13.0.5.4”,“dashboard”:“14.0.3.3”,“daynight”:“14.0.1”,“donotdisturb”:“14.0.1.1”,“extensionsettings”:“13.0.4”,“featurecodeadmin”:“13.0.6.4”,“findmefollow”:“14.0.1.18”,“firewall”:“13.0.52”,“framework”:“14.0.2.14”,“fw_langpacks”:“14.0.1”,“infoservices”:“13.0.1.2”,“irc”:“2.11.0.7”,“ivr”:“13.0.27.7”,“languages”:“13.0.6”,“logfiles”:“13.0.10.4”,“manager”:“13.0.2.5”,“miscapps”:“13.0.3.1”,“miscdests”:“13.0.5”,“music”:“13.0.22.3”,“outroutemsg”:“13.0.2.1”,“paging”:“13.0.26.5”,“parking”:“13.0.19.8”,“phpinfo”:“13.0.2”,“pinsets”:“13.0.8”,“pm2”:“13.0.5”,“printextensions”:“13.0.3.1”,“queueprio”:“13.0.2”,“queues”:“14.0.2.14”,“recordings”:“13.0.30.12”,“ringgroups”:“14.0.1.4”,“setcid”:“13.0.6.2”,“sipsettings”:“14.0.27.1”,“soundlang”:“14.0.4.3”,“sysadmin”:“14.0.12.2”,“timeconditions”:“14.0.2.13”,“ucp”:“14.0.2.3”,“userman”:“14.0.3.40”,“voicemail”:“14.0.1.20”,“weakpasswords”:“13.0.2”},“lic”:false}"} to //oobe/sysadmin
I’ve tried in different browsers, different computers, all still shows the same. From the terminal on the server I can ping both IP and URL addresses so I believe DNS and the network is working correctly.
Any thoughts?
Posts: 1
Participants: 1
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Issue with update module
@Interlink wrote:
Hello, when trying to check for updates online I receive an error:
I tried running yum update in cli and get the following:
Loaded plugins: fastestmirror, versionlock
Could not retrieve mirrorlist http://mirrorlist.sangoma.net/?release=7&arch=x86_64&repo=os&dist=$dist error was
12: Timeout on http://mirrorlist.sangoma.net/?release=7&arch=x86_64&repo=os&dist=$dist: (28, ‘Connection timed out after 30001 milliseconds’)One of the configured repositories failed (Unknown),
and yum doesn’t have enough cached data to continue. At this point the only
safe thing yum can do is fail. There are a few ways to work “fix” this:1. Contact the upstream for the repository and get them to fix the problem. 2. Reconfigure the baseurl/etc. for the repository, to point to a working upstream. This is most often useful if you are using a newer distribution release than is supported by the repository (and the packages for the previous distribution release still work). 3. Run the command with the repository temporarily disabled yum --disablerepo=<repoid> ... 4. Disable the repository permanently, so yum won't use it by default. Yum will then just ignore the repository until you permanently enable it again or use --enablerepo for temporary usage: yum-config-manager --disable <repoid> or subscription-manager repos --disable=<repoid> 5. Configure the failing repository to be skipped, if it is unavailable. Note that yum will try to contact the repo. when it runs most commands, so will have to try and fail each time (and thus. yum will be be much slower). If it is a very temporary problem though, this is often a nice compromise: yum-config-manager --save --setopt=<repoid>.skip_if_unavailable=trueCannot find a valid baseurl for repo: sng-base/7/x86_64
Current PBX Version:
14.0.2.17
Current System Version:
12.7.4-1803-1.sng7Any help would be great!
Posts: 1
Participants: 1
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Asterisk CLI login error after freepbx server restart
@RZ1 wrote:
Hi,
Why do I get this error at CLI for 3 minutes (at freepbx 13 I didn’t get this error) after freepbx 14 sangoma 7 restart If I try to log into asterisk with asterisk -rv:
[root@freepbx ~]# asterisk -rv
Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?)
[root@freepbx ~]#
Posts: 3
Participants: 2
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Call forwarding on SIP trunk | Original CallerID not being transmitted
@sipsepp wrote:
Hi FreePBX users and admins!
Running FreePBX with Asterisk 13.19.1.
Enabling call forwarding on my extension 123 by *72<CF_destination_number># is being accepted by the PBX, but both internal and external calls to the extension 123 are not being forwarded.
The Asterisk CLI tells me the following:
#######################################################
[2018-04-11 15:42:27] NOTICE[20580][C-0000017b]: app_dial.c:1000 do_forward: Not accepting call completion offers from call-forward recipient Local/<CF_destination_number>@from-internal-0000036b;1
#######################################################Google doesn’t help searching with these info.
Anyone an idea?
Thanks in advance!
Best Regards,
Sepp
Posts: 1
Participants: 1
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Network RX error after freepbx 14 upgrade
@RZ1 wrote:
Hi,
I recognised that I’m getting RX dropped packets after freepbx 14 upgrade:
eth0: flags=4163<UP,BROADCAST,RUNNING,MULTICAST> mtu 1500
RX packets 289879 bytes 44472747 (42.4 MiB)
RX errors 0 dropped 16965 overruns 0 frame 0Is there any fix on it?
regards,
RZ
Posts: 1
Participants: 1
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Cannot reload EPM after update to 14
@Interlink wrote:
I am trying to save and rebuild configs in EPM and I keep getting an error:
I am not sure what “hpro” is to enable it in the sysadmin module.
Posts: 1
Participants: 1
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Fresh install on KVM won't boot
@gdi2k wrote:
I have a freshly installed FreePBX distro which won’t boot. I installed to KVM on Proxmox VE 5.1. I’m using SNG7-FPBX-64bit-1712-2.iso and chose the recommended Asterisk version (13).
Install went smoothly, but during boot I see lots of failures. It eventually stops at:
This is the second install - the first time it behaved the same way.
I’ve been using FreePBX on KVM / Proxmox for years without any issues like this.
Any tips?
Posts: 1
Participants: 1
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Why is my announcement play even before the call is answered?
@rizkips wrote:
hi everyone
i am really new with freepbx environment, and i really need help about it
i am using FreePBX 13.0.194.2 and i want to build an outbond IVR system in my small office
i integrated it with loway wombatdialer. it actually works as an IVR system but i have an issue , why the IVR start play even when the customer hasnt answered the call ?? how can fix that ?
Posts: 1
Participants: 1
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Freepbx 14 error after upgrade
@RZ1 wrote:
Hi,
I upgraded freepbx 13 to freepbx 14 with Sangoma 7. If I reboot the Sangoma 7 at login I get the below error message. What does it causes this error? How can I eliminate it? I ran mysqlcheck -A. the command found no inconsistency in mariadb.
Exception: SQLSTATE[HY000] [2002] No such file or directory::SQLSTATE[H Y000] [2002] No such file or directory in file /var/www/html/admin/libraries/utility.functions.php on line 204
Stack trace:
- Exception->() /var/www/html/admin/libraries/utility.functions.php:204
- die_freepbx() /var/www/html/admin/libraries/BMO/Database.class.php:142
- PDOException->() /var/www/html/admin/libraries/BMO/Database.class.php:137
- PDO->__construct() /var/www/html/admin/libraries/BMO/Database.class.php:137
- FreePBX\Database->__construct() /var/www/html/admin/libraries/BMO/FreePBX.class.php:69
- FreePBX->__construct() /var/www/html/admin/bootstrap.php:153
- require_once() /etc/freepbx.conf:9
- include_once() /var/lib/asterisk/bin/fwconsole:12
Posts: 1
Participants: 1
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Freepbx 14 restore backup taken in freepbx 13
@antonis77 wrote:
Hi,
Is it possible to restore a backup which was taken in FreePBX 13 version to a system running FreePBX 14 ?
I ve tried and my system crashed immediately.
Posts: 2
Participants: 2
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Time to voicemail 18 seconds for internal calls
@wnell wrote:
I am using the latest SNG7. I have follow me turned off on my extensions. Everything else is pretty much at their defaults.
If an external user (say cellphone) calls the system, the IVR picks up correctly. If I enter an extension, say 10, the line rings. After the ring time duration it correctly plays the user’s unavailable voicemail greeting. All good.
However, if I make an internal call, say extension 20 calling extension 10, the system rings the extension for the ring duration however after that duration I see the timer on my phone start counting (i.e. call answered) yet I have to listen to complete silence for 18 seconds before the unavailable voicemail greeting plays. Additionally, the system default voicemail plays and not the users recorded message like when an external user calls that extension.
What can I look for? PS: All endpoints are using chan_pjsip.
Posts: 3
Participants: 2
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DigitTimeout
@ibapah wrote:
Is there a way to set DigitTimeout with the GUI? I prefer to have to have to hit the dial button to launch the call. Somehow I got it to be that way one one system but on a different system it has a very short timeout and sends the call before I have completed dialing.
Posts: 2
Participants: 2
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Customer Support Line with Time Restrictions
@KBliven wrote:
I have a client that we installed FreePBX 13 on and everything has been working great. They have about 10 or so extensions. Most in house but a few are virtual extensions forwarding to salesmen cell phones in the field. We are also using SIPStation for the trunks and mostly Aastra 6757i phones on the desks.
They now want to add a customer support line to field calls from the public. The issue though is with limited staff they want to redirect this line to various other staff based on the time of day. In addition, these redirections may need to be altered from the schedule once or twice a day, so they need to be able to modify the time table with little notice.
So I guess what I am asking is what is the best way to go about this? To summarize the requirements :
- Set up a basic schedule to route calls from this customer support extension to various other extensions (or go to VM off hours)
- Give a few employees the ability to modify this schedule if needed (preferably keeping them out of the full blown GUI if possible).
- We will probably need to implement queues for this number as well. I haven’t done that before but wanted to mention it in case it alters the solution.
If there is a commercial module that would accomplish this we would be happy to purchase it. They are also looking into third party solutions outside of FreePBX but if we can accomplish everything internally without an additional ongoing cost I would prefer to do so.
Thanks for any suggestions,
Kent
Posts: 1
Participants: 1
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Changing recordings location to ReadyNAS and limiting incoming calls to one per time
@BlackMagic wrote:
Hi, I’m pretty new in all the PBX stuff since I’ve changed my job recently
The thing is, I am trying to reconfigure some things in the PBX. First of all I’ve enabled call recording in one extention to check if it will work - it does. So the thing is that I want to store the recordings on the ReadyNAS server - is it possible? Should I somehow try to mount ReadyNAS smb on the PBX server and then try to setup the path in the web console?
Second thing is - how to limit incoming calls in the extension settings? Right now, when there are few incoming calls at one time, everyone has the normal dialing signal, just like the line is not used at the moment.
I want them to hear “busy” signal or sound message saying “all the lines are busy ATM” or something similiar. How can I do it? I’d love to get a step by step guide if possible.Cheers!
Posts: 7
Participants: 3
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FreePBX 14 + Custom Context
@DIMMittriy wrote:
Hello!
Need help in solving the problem:
on version of FreePBX 12 enjoyed module Custom Context and here had to to to move on a new server and set on him the latest FreePBX Distro. Installed it the module that created it and the rules was that the rules in this module do not apply to Extension. When saving settings the numbers in freepbx logs:
[[2018-Apr-13 09:26:25] [WARNING] (core/functions.inc.php:6136) is a Deprecated Function core_devices_get detected in /var/www/html/admin/modules/customcontexts/views/extensions_hook.php on line 3 [[2018-Apr-13 09:26:25] [WARNING] (libraries/module functions.legacy.php:7) is a Deprecated Function module_getinfo detected in /var/www/html/admin/modules/customcontexts/functions.inc.php on line 13
Dancing with a tambourine did not lead to anything good, except that freepbx stopped working at all. Can someone tell me a specific solution?
Posts: 3
Participants: 2
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