@ddussaultce wrote:
I am trying to configure some of my yealink’s phone can receive only 2 calls at the same time, if a 3rd one comes in it gets the busy message in the voicemail.
I also have some extensions that I just want them to have only 1 call at a time and since they won’t have a voicemail, the call will be dropped
I didn’t find a place to configure this, is it possible ?
Posts: 1
Participants: 1
@bmayfield wrote:
Hello, We are having issues with getting our FreePBX Server Connected with ATT IP Flexible Reach and was needing advice on the issue. The instructions from att are to forward all Voice (Sip) Trafic to the address “1.1.2.1” and since att has no authentication there is no login and when we have our test and turn up att is say they are getting a 401 error.
Here is what att is sending up as an error log:
Here is what we have for out SIP Trunks
Outgoing:
qualify=2000
dtmfmode=rfc2833
host=1.1.2.1
insecure=invite,port
context=from-pstn
directmedia=no
nat=noIncoming:
qualify=2000
dtmfmode=rfc2833
host=1.1.2.1
insecure=invite,port
context=from-pstn
directmedia=no
nat=noAny help is greatly appreciated!
Posts: 1
Participants: 1
@leem wrote:
Hi, Was hoping someone could help. my system is currently stuck after a Framework upgrade to V14. Module Admin shows “Disabled; Pending Upgrade to 14.0.5.1” and Apply Config gives the following errors:
exit: 1
Unable to continue. The process has been signaled with signal “7”. in /var/www/html/admin/modules/recording_report/vendor/symfony/process/Process.php on line 434 #0 /var/www/html/admin/modules/recording_report/vendor/symfony/process/Process.php(212): Symfony\Component\Process\Process->wait() #1 /var/www/html/admin/libraries/media/Media/Driver/Drivers/FfmpegShell.php(46): Symfony\Component\Process\Process->run() #2 /var/www/html/admin/libraries/media/Media/Media.php(126): Media\Driver\Drivers\FfmpegShell::installed() #3 /var/www/html/admin/libraries/BMO/Media.class.php(80): Media\Media::getSupportedFormats() #4 /var/www/html/admin/libraries/BMO/Media.class.php(40): FreePBX\Media->getSupportedFormats() #5 /var/lib/asterisk/bin/retrieve_conf(77): FreePBX\Media->getSupportedHTML5Formats() #6 {main}1 error(s) occurred, you should view the notification log on the dashboard or main screen to check for more details.
I´ve tried upgrading via CLI with
- fwconsole ma upgrade framework
This returns
- No repos specified, using: [standard] from last GUI settings
framework is the same as the online version, unable to upgrade
Updating Hooks…DoneThen
fwconsole reload
This returns the following errors
Reloading FreePBX
Error(s) have occured, the following is the retrieve_conf output:
exit: 1
Unable to continue. The process has been signaled with signal “7”. in /var/www/h tml/admin/modules/recording_report/vendor/symfony/process/Process.php on line 43 4
#0 /var/www/html/admin/modules/recording_report/vendor/symfony/process/Process.p hp(212): Symfony\Component\Process\Process->wait()
#1 /var/www/html/admin/libraries/media/Media/Driver/Drivers/FfmpegShell.php(46): Symfony\Component\Process\Process->run()
#2 /var/www/html/admin/libraries/media/Media/Media.php(126): Media\Driver\Driver s\FfmpegShell::installed()
#3 /var/www/html/admin/libraries/BMO/Media.class.php(80): Media\Media::getSuppor tedFormats()
#4 /var/www/html/admin/libraries/BMO/Media.class.php(40): FreePBX\Media->getSupp ortedFormats()
#5 /var/lib/asterisk/bin/retrieve_conf(77): FreePBX\Media->getSupportedHTML5Form ats()
#6 {main}Phones are still working but GUI is all over the place.
All other standard modules are working and updated as they should be except Framework which is disabled pending upgrade.
Thank you in advance for any assistance someone can provide.
Lee
Posts: 1
Participants: 1
@becausewecan wrote:
Hi All,
I wonder if anyone can shed any light on this, we’ve setup a freepbx server on amazon ec2 - everything is working fine, we have a bunch of connections from our office to the pbx plus a few external extensions which work fine.
I have an odd issue with 1 of the external guys, who can connect and register no problem, but theres no audio either way - i can see the reason why when i run a tcpdump - it seems Freepbx is sending the UDP data to his internal address (192.168.1.x) instead of to his router’s external addres (81.123.123.123) - any ideas?
Hi Setup is exactly the same as mine, and mine works fine - same isp same router, same phone
Posts: 4
Participants: 2
@omairoja wrote:
Good afternoon all, when i run “dahdi show status” i get the following below. could anyone tell me of provide some reading material on what CRC is. I am also experening drop calls from time to time. I need to report this to my telco but i am trying to arm myself with as much information as possible.
Description Alarms IRQ bpviol CRC Fra Codi Options LBO
WCTE23X (PCI) Card 0 Span 1 OK 0 0 661 ESF B8ZS 0 db (CSU)/0-133 feet (DSX-1)
WCTE23X (PCI) Card 0 Span 2 OK 0 0 113986 ESF B8ZS 0 db (CSU)/0-133 feet (DSX-1)
Posts: 2
Participants: 2
@JohnChingo wrote:
Our FreePBX version is 13.0.195.4
We have an IVR where one of the Digit Entries is “Play System Recording.”
The dialplan generated in extensions_additional.conf has “Hangup” after playing that recording, even though the “Return” option in the GUI is set to “Yes” for that entry. Call is therefore hung-up instead of returning to the IVR after playing.
Instead of the currently generated
exten => 12,1,Answer
exten => 12,n,Playback(custom/parkinginfo)
exten => 12,n,Hangupshouldn’t it be
exten => 12,1,Answer
exten => 12,n,Playback(custom/parkinginfo)
exten => 12,n,ReturnThe System Recordings version is 13.0.30.12
Thank you,
-John
Posts: 3
Participants: 2
@sentinelace wrote:
I did the update and it won’t remove from my dash
framework has been automatically upgraded to fix security issues: SEC-2018-001
Posts: 2
Participants: 2
@BluePukeko wrote:
My experience is with web servers. I have a reasonable understanding of FreePBX.
I understand that its possible to write a module that would do an external HTML enquiry.What I what is assistance in creating the module and integrating it in to FreePBX.
The module needs to take input from the users phone and then pass that number via HTML to an external server which would then respond with another number which could be one of an internal extension, Ring Group or an external number.
Any assistance would be appreciated.
Posts: 4
Participants: 2
@crodseth wrote:
I’m attempting to connect an FXS port to our existing analog 1 zone paging system.
Here’s where I’m at…
FXS port out to amplifier. Created a DAHDI extension 500 that uses that FXS port. Also created a paging group 502 that includes only extension 500. When I call extension 500, it just rings. The intercom actually plays the ringing, followed by some old AOL type computer speak. When I call 502, I get a beep as if you can make your announcement. Ringing plays over the intercom still.
Do I need a paging gateway? I was hoping I could set the 500 extension to auto-answer, and then the paging system would magically work. However, auto-answer doesn’t do anything…just ringing, which then plays over the intercom. Please advise. If I need a paging gateway, I’ll get one, but I’m surprised the ringing plays over the intercom.
Thank you for any and all assistance.
Posts: 2
Participants: 2
@ddussaultce wrote:
I’m trying to update the firmware for my Yealink phones and it seems I am doing something wrong but can’t figure out what.
Added the 0.00 firmware to slot 2
I downloaded the latest firmware files that I need for my types of phones (t40 and t42s) in the slot 2
Renamed the file to match the ones in slot 1
Made sure asterisk was the owner of the files
Change in my Yealink template to use Firmware Slot 2Told the system to update phone, still running old version
What am I missing to get the phones updated ?
Posts: 2
Participants: 2
@faisalkhan wrote:
Hi all,
I want to setup an IVR with only option 1 for voicemail and want to ring the queues during the IVR announcement. so I want the callers to be answered by the Queue agents during the IVR or if I can give them the option to skip the announcement and directly goes to the agents ringing.
Posts: 3
Participants: 2
@bajramia wrote:
Hi All,
I’m using the Bulk Handler to import extensions and i have imported the first worked good, I had made a mistake which i had do delete all the extension from the extension module, now when i use the Bulk Handler I get this message “there is nothing to import” is the any cache i have to clear if so how do i do it thank you.
Posts: 1
Participants: 1
@ScottD wrote:
I have a site with a SIP trunk from Nextiva that occasionally fails to register. It will be working fine for days and then suddenly drops off. It just gives unreachable error in the logs. If I reboot the PBX it will register again sometimes for a few hours or a few days. I even have to reboot the router sometimes to get it to stay registered for more than 30 minutes. Sangoma PBXact 40 Version 14.0.5.2 Sonicwall Router with firewall configured according to Nextiva’s requirements
I have several other sites with Nextiva trunks with no issues using the same trunk parameters. The other sites use a different router.
Any thoughts?
Posts: 1
Participants: 1
@PeterJ wrote:
Hi,
We have a number of production systems (of varying versions 13 to 15) setup with warm spares doing nightly backups and restores. It seems that the default behavior (for FreePBX Restores) is for the CDR database to be restored so that it clears the current tables during the import of the SQL backup file.
This is desirable in most situations, but I wonder if it might be possible to alter the restore script for particular situations so that the SQL tables of CDR data are appended to during the import rather than overwritten ?
Posts: 1
Participants: 1
@leem wrote:
Hi, a strange thing has happenend to the IVR on one of my boxes (FreePBX 13.0.195.19). The IVR was fine up until 2 weeks ago and has been fine for a year previous.
- When the call comes in there is nothing and eventually cuts off
- The call registers as “answered” in the CDR
- The IVR rings through correctly from a test extension that forwards to the IVR
- The number rings through to another extension correctly
Thank you in advance for any thoughts.
Posts: 5
Participants: 2
@faisalkhan wrote:
Hi Guys,
I want to setup ** feature code to get out of the call instead of listening the voicemail.
Posts: 1
Participants: 1
@mrozema wrote:
So far, I can see how many temporary greetings are currently recorded in the voicemail system. What I can’t figure out is which extensions have them.
Any ideas?
Posts: 2
Participants: 2
@kanky wrote:
Hi,
We are having an intermittent issue (every other call) whereby incoming calls are being rejected and the caller hears the number you have dialed cannot be reached.
When we examined the logs we say that fail2ban was rejecting the IP of the sip provider.
We have since whitelisted the IP list of the SIP provider (194.213.29.0/24) via System Intrusion and restarted by the problem remains.
Now, when I look at the logs the failed calls do not appear as log entries.
Any ideas on what I can check or do to rectify the issue?
Posts: 1
Participants: 1
@wzkds wrote:
Has anyone here had success integrating Redtail CRM? They have an API, but I haven’t seen anything with a simple callerid variable that I can at least use to call pop.
Posts: 1
Participants: 1
@XcamaroX wrote:
I setup a Call Flow Control setup and it works wonders. But the only problem I have encountered is that when the call is transferred for the day (night time) to the intended answering service, the quality of the call is subpar. You get a choppy call, even tho you can understand is not 100% clear as regular calls during the daytime.
Any ideas why is this happening?
All the other aspects of the PBX setup work as intended during day time calls, it only happens when the system is in night time and the call gets sent outside our system and to the answering service.
I have tried different services, phone numbers, to make sure is not the main answering service, and all numbers that I dial experience the same issue.
Any help will be appreciated.
Thanks in advance!
Posts: 1
Participants: 1
