December 1, 2019, 5:50 pm
@moussa854 wrote:
I upgraded my FreePBX to 15 (Asterisk Version: 13.29.2) over the weekend and I noticed my extensions were going offline. I noticed (so far) this happens after an Apply Configuration OR /var/lib/asterisk/bin/module_admin reload
[2019-12-01 20:41:08] ERROR[2439] res_pjsip.c: Error 171060 'Unsupported transport (PJSIP_EUNSUPTRANSPORT)' sending OPTIONS request to endpoint 201
[2019-12-01 20:41:08] VERBOSE[2439] res_pjsip/pjsip_configuration.c: Endpoint 201 is now Unreachable
In the past, I had Allow Transports Reload set to Yes but Googling PJSIP_EUNSUPTRANSPORT suggests it need to be set to No?
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November 13, 2019, 2:28 pm
@Red wrote:
Where are the VPN logs to tell me if my vpn server is running or not. Also a way to tell if my tunnels are active. I can see by my vpn server, under sys admin pro, that none of my clients are connecting, and they were Monday.
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December 3, 2019, 5:53 pm
@tletourneau wrote:
When I try to update the modules the status for all modules is “Enabled; Not available online”. I have Standard, Extended and Commercial repositories enabled… I’ve tried several times and rebooted the system with no change. The system is running FreePBX 14.0.11 and Asterisk 13.29.2. I’ve done some searching on the forum but haven’t found a resolution yet.
What can I try to resolve this?
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December 4, 2019, 3:10 am
@AmidouFlorian92 wrote:
Hi
I’m searching a way to create and test an IVR using asterisk CLI.
Is someone can explain or can give me a way to create and test an IVR using CLI ?
Thanks
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December 4, 2019, 8:08 am
@acjay wrote:
Hello -
I’m having some issues with my S700 after adding the EXP100 and I’m hoping someone can help.
For reference, I am using Endpoint Manager to configure my S700 and EXP100 module, so all settings are made through EM.
Since adding my EXP100, my phone is slow to respond, often fails to register properly and frequently has issues with buttons working. Restarting the phone takes significantly longer now that it is set up to use the EXP100. There is also a noticeable draw delay on the screen for incoming calls.
The firmware on the EXP100 is the most recent (2.0.5.0), however I cannot seem to get the phone to update to 2.0.4.74 (from 2.0.4.72) via EM. I have the correct firmwares in the two slots (1.60 in 1 and 1.59 in 2). I have a 1 week delay in EM for the phones to automatically check for updates (config only, I believe; not firmware), yet my phone now seems to be checking multiple times an hour, each time completely rendering the phone unusable. I cannot call it (busy signal), and picking up the handset removes the “Checking firmware” box but will not allow me to place a call.
None of this happened before I added the EXP100. I am going to try manually updating the firmware, but I suspect two small minor releases don’t address this issue.
Has anyone else had this issue before?
Thanks in advance.
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November 14, 2019, 2:20 am
@gettrashed wrote:
So i have FreePBX and i have Global Ip address. i want that someone who goes on global ip on web redirect on UCP site ( i want that Admin Page to be accessable from only local addresses ) any help?
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December 5, 2019, 2:23 am
@AmidouFlorian92 wrote:
Hi all
We are trying to kill all processes that running on tcp port 4573 (Fastagi listening port) but when we kill a process another application uses the port.
So :
-Can we move Fastagi to another port and what can we do that ?
-Or is there a possibility to definitvly kill all process listening on port 4573?
thanks
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December 6, 2019, 8:24 am
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December 6, 2019, 8:51 am
@faisalkhan wrote:
Hi Guys,
I have installed fop2 on Freepbx 14 and it worked fine in the beginning. I have configured it successfully and it was showing every thing fine.
But now i am not able to view any buttons on the fop2 user protal.
I need help.
How can I uninstall this FOP2 package as a whole or if we can fix it.
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November 14, 2019, 1:06 pm
@chrischevy wrote:
I just installed my first FreePBX 15 system (I manage a lot of FreePBX 14)
When I’m running a backup, there is no email being sent. Email type option is set to “Both” , but I also tried “Success”. The backup works and gets transfered on my FTP server.
I receive emails regarding fail2ban from the same system so I guess that my email configuration is ok.
I checked the email server logs and I don’t see any traces of the message.
I updated the Backup and Restore module to Edge version 15.0.8.68 but still no luck.
Anyone has any clue on what to do next ?
EDIT: I just checked /var/log/maillog and there is nothing getting logged after a backup
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December 6, 2019, 1:54 pm
@chrischevy wrote:
I have an issue with speed dials on Yealink phones.
Endpoint Manager adds ‘**’ in the “Extension” field (the rightmost column if you check Yealink’s web interface) of to every DSS Keys. It’s fine for BLFs, but it creates a problem for speed dials.
What it does is that once the call is dialed using a speeddial, the phone dials ** after a few seconds. For most call, it doesn’t create any problem, as the “" is dialed during ring time, but for calls answered rapidly by an IVR, the "” in caught by the IVR.
To fix the problem, the “**” should only be added to DSS Keys configured as BLFs, but Endpoint manager adds it to every key, regardless of the type.
As anyone ever encountered this situation ?
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November 14, 2019, 1:06 pm
@chrischevy wrote:
I just installed my first FreePBX 15 system (I manage a lot of FreePBX 14)
When I’m running a backup, there is no email being sent. Email type option is set to “Both” , but I also tried “Success”. The backup works and gets transfered on my FTP server.
I receive emails regarding fail2ban from the same system so I guess that my email configuration is ok.
I checked the email server logs and I don’t see any traces of the message.
I updated the Backup and Restore module to Edge version 15.0.8.68 but still no luck.
Anyone has any clue on what to do next ?
EDIT: I just checked /var/log/maillog and there is nothing getting logged after a backup
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December 6, 2019, 6:51 pm
@FPBX135 wrote:
Hello there! I try to apply changes and I keep getting this message no matter what I do. What can I do to fix this?
Reload failed because retrieve_conf encountered an error: 1
exit: 1
Unable to continue. Unable to locate the FreePBX BMO Class 'Userman’A required module might be disabled or uninstalled. Recommended steps (run from the CLI): 1) fwconsole ma install userman 2) fwconsole ma enable userman in /var/www/html/admin/libraries/BMO/Self_Helper.class.php on line 213 #0 /var/www/html/admin/libraries/BMO/Self_Helper.class.php(106): FreePBX\Self_Helper->loadObject(‘Userman’) #1 /var/www/html/admin/libraries/BMO/Self_Helper.class.php(37): FreePBX\Self_Helper->autoLoad(‘Userman’) #2 /var/www/html/admin/modules/contactmanager/Contactmanager.class.php(21): FreePBX\Self_Helper->__get(‘Userman’) #3 /var/www/html/admin/libraries/BMO/Self_Helper.class.php(124): FreePBX\modules\Contactmanager->__construct(Object(FreePBX)) #4 /var/www/html/admin/libraries/BMO/Self_Helper.class.php(37): FreePBX\Self_Helper->autoLoad(‘Contactmanager’) #5 /var/www/html/admin/libraries/BMO/Hooks.class.php(298): FreePBX\Self_Helper->__get(‘Contactmanager’) #6 /var/www/html/admin/libraries/BMO/Hooks.class.php(39): FreePBX\Hooks->preloadBMOModules() #7 /var/lib/asterisk/bin/retrieve_conf(81): FreePBX\Hooks->updateBMOHooks() #8 {main}
1 error(s) occurred, you should view the notification log on the dashboard or main screen to check for more details.
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December 8, 2019, 5:02 pm
@willskipp wrote:
I’ve got two FreePBX 15 installations. Both are on Edge mode and fully updated. Following this guide: https://wiki.freepbx.org/pages/viewpage.action?pageId=165806910 I have configured a primary and secondary server.
Running a backup job returns this error:
In CurlFactory.php line 186:
cURL error 60: Issuer certificate is invalid. (see http://curl.haxx.se/libc
url/c/libcurl-errors.html)
At a guess cURL is rejecting the self signed FreePBX certificate?
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December 9, 2019, 3:20 am
@mjk666_mjk wrote:
Hi Fellas,
I have a situation
We have an issabel server which has 2 NICs
NIC1 for the connection to providers Intranet (10.105.0.0 network)
NIC2 for our local network and IP-Phones (172.16.0.0 network)
We have register sip accounts as a trunk with this configuration:
username=2222222
type=peer
trustrpid=yes
sendrpid=yes
secret=222222
qualify=yes
insecure=port,invite
host=10.105.X.X
disallow=all
allow=ulaw&alaw&gsm
and in my advanced sip i have:
NAT=Route
IPCONF=Static
ExternalIP: 10.105.Y.Y (MY NIC1 IP)
Local Net: 172.16.0.0 255.255.255.0
Reinvite Behavior: Yes
Everything works fine, but when i blind transfer a call to another phone or when i set always forward to another phone, the call will drop afters 30 second
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December 10, 2019, 7:08 am
@sorvani wrote:
I have a working FreePBX 14 Distro deployment. I spun up a new VPS instance and installed FreePBX 15 clean. I poked a round a few settings and setup a LE cert. Everything looked fine.
So I decided to restore the FreePBX 14 backup file. That seemed to go with no errors except for a single sound file not getting copied. Great.
I then look in Certificate management and I do not see my certificate. Ok. I expected that since I was using certbot and manually importing. No big deal.
But the new cert I had requested prior to the restore was also gone. Ok fine, it removed everything. I can accept that.
I look in /etc/asterisk/keys and see that the new cert was in the folder.
![image]()
So I clicked the Import Locally button and get this error.
Issue Created: https://issues.freepbx.org/browse/FREEPBX-20882
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December 10, 2019, 7:46 am
@muetin wrote:
Hola a todos, lo primero presentarme debido a que soy nuevo en el foro. Resulta que he realizado en la central la configuración para Movistar y cuando hago una llamada saliente a un número de teléfono un poco especial no funciona correctamente, es decir, llamo al 1004 y parece que de primeras comunica pero no se escucha las locuciones y si llamo a un número de una empresa que tiene algún tipo de centralita da un par de tonos y se corta y no logro entablar conversación, mientras que si llamo a un teléfono por ejemplo de una casa particular o a un móvil todo perfecto. Las llamadas entrantes funcionan perfectamente. Como es mi primera configuración lo estoy probando con Zoiper versión gratuita.
Por favor, alguien me podía indicar a que se debe, este hecho.
Gracias, un saludo.
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November 18, 2019, 9:19 am
@dan_ce wrote:
Similar to this question, I’m trying to access ring groups, page groups and queues from within the dialplan. I’ve tried all sorts of wording, but everything either returns no route to destination or subscriber absent.
This is what my context looks like at the moment
[check-whitelist]
exten => s,1,Noop(Entering user defined context [check-whitelist] in extensions_custom.conf)
exten => s,n,AGI(lgaetz-cmcheck.php,${CALLERID(number)})
exten => s,n,gotoif($["${whitelist}"="false"]?,1000)
exten => s,n,Return
exten => s,1000,Dial(PJSIP/22)
This works, if the number is found, it routes to extension 22. But I’d really like to route to ring group 61. Not sure how I’d do that?!
Thank you!!
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December 10, 2019, 9:05 am
@Joeli0 wrote:
Hello all. Been using hosted FreePBX platforms for a few years now and loving them so far. My question is this. I am looking to have an OpenVPN config where the clients can access each other, mainly Web GUI on phones. This is where I’m currently at…
OpenVPN running on hosted FreePBX @ 10.8.0.1
Ext1 is @ 10.8.0.2
Ext2 is @ 10.8.0.3
PC with OpenVPN client is @ 10.8.0.5
I have access to FreePBX from PC via VPN. All is good there
I have registered and fully functioning Grandstream phones Via OpenVPN
I want to be able to access the web GUI of the phones from the PC via the VPN.
So OpenVPN from 10.8.0.5 log into the web GUI of Ext1 @ 10.8.0.2.
I have seen the client to client feature on the OpenVPN forums but wasnt sure if and how it could be used here, Any help appreciated. Thanks!
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December 10, 2019, 11:39 am
@munozj wrote:
I use one of our DIDs to test line quality for remote employees and I use a stream from our broadcast encoder as the feed.
/usr/bin/mpg123 -q -r 8000 -f 8192 --mono -s http://10.255.96.50:8000
Today during some testing I was hearing the stream then after dialing back a few times the stream quit. I can listen to the stream directly from the url on my computer. Is there a service or something that may need to be restarted to get the stream flowing again?
I seam to remember this happening before and they only fix I could find to was to reboot the entire pbx.
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