@ryancurtis wrote:
I have found multiple posts similar to this:
This is not working for me. I have ran the commands grouped and even separated them out. Everytime I run yum update or upgrade I get the same error"
Error: Trying to remove “sangoma-pbx”, which is protected
Posts: 1
Participants: 1
@jpark1205 wrote:
Hello everyone good morning
[2019-08-23 10:01:10] ERROR[31806] res_pjsip.c: Unable to retrieve PJSIP transport ‘udp,tcp,ws,wss’
So this is the error log that i am getting and I’m just wondering where i can find the “res_pjsip.c”?
It that the reason why i am getting an error on the log because i dont have that file? And what does it do??Thank You
Posts: 1
Participants: 1
@BGM wrote:
Is there a way in FreePBX 14 to enable intercom for all phones? On an individual phone, you dial *54 to enable intercom, but I want to do it globally.
Posts: 1
Participants: 1
@comfine wrote:
Hi All,
some of our Systems got shutdown every night cause we backup the VM and need to shutdown therefor.
While this is happening we got 4 warnmails with the following content per reboot:Exception: Asterisk is not connected in file /var/www/html/admin/libraries/php-asmanager.php on line 242 Stack trace: 1. Exception->() /var/www/html/admin/libraries/php-asmanager.php:242 2. AGI_AsteriskManager->send_request() /var/www/html/admin/libraries/php-asmanager.php:591 3. AGI_AsteriskManager->Command() /var/www/html/admin/libraries/php-asmanager.php:1653 4. AGI_AsteriskManager->parseAsteriskDatabase() /var/www/html/admin/libraries/php-asmanager.php:1622 5. AGI_AsteriskManager->database_show() /var/www/html/admin/libraries/php-asmanager.php:210 6. AGI_AsteriskManager->LoadAstDB() /var/www/html/admin/libraries/php-asmanager.php:1708 7. AGI_AsteriskManager->database_get() /var/www/html/admin/modules/timeconditions/bin/schedtc.php:28
This is annoying and of course this warning should just be send out if asterisk is not connected WITHOUT rebooting. I red about a bug in the past, but that was solved and the system is up to date, so there should no warnmail. Can anybody help?
Posts: 2
Participants: 1
@GarFin wrote:
Hi, total newbiw here , trying to get a SPA3120 working with FreePBX12/Asterisk11 (yes i know its an old build, but its the only one i could make run in a vm.)
I’ve got the Line1 side of the ATA sorted and communicating with the FreePBX as extension 776 , and other softphones can call it , and it can call them. I too have sorted out outbound dialing from all extensions to my SIP provider. (who is used for outbound only.)
My only trouble that i cant get my fat head around, is setting up the inbound route (PSTN on the ATA) through into FreePBX… How do i config the ATA (PSTN) side and have it route into the FreePBX?, is it an inbound trunk , and/or an inbound route or inbound extension i am meant to be configuring ?? (and what do all those settings look like?)
I can make calls between “Line 1” configured on the SPA-3120 (extension number=776) and the other extensions (and viceversa). If I go and activate the “PSTN Ring Thru Line 1” option, I can answer incoming calls from the PSTN on the Line 1 handset but thats routing straight through the ATA3120 from PSTN port to LINE Port where the handset is connected (which i dont want, as i want all incoming PSTN to end up being processed buy a FreePBX IVR before being handed off to the handset extension 776.
Help please!!
Posts: 2
Participants: 2
@jtomelevage wrote:
What, if anything, is being worked on to accommodate the American Stop Robocalls Act? Specifically, what can FreePBX/Asterisk users do to prevent their legitimate phone numbers from being marked as SPAM / Robocalls?
John
Posts: 1
Participants: 1
@charlietoner wrote:
I cannot save any config changes. I get the following error:
exit: 1
Unable to continue. Failed to acquire the “crontab-asterisk” lock. in /var/www/html/admin/libraries/Composer/vendor/symfony/lock/Lock.php on line 101
#0 /var/www/html/admin/libraries/BMO/Cron.class.php(115): Symfony\Component\Lock\Lock->acquire(true)
#1 /var/www/html/admin/libraries/BMO/Cron.class.php(228): FreePBX\Cron->addLine(’*/1 * * * * /va…’)
#2 /var/www/html/admin/modules/pagingpro/Pagingpro.class.php(557): FreePBX\Cron->add(Array)
#3 /var/www/html/admin/modules/pagingpro/Pagingpro.class.php(542): Pagingpro->addMasterCron()
#4 /var/www/html/admin/libraries/BMO/DialplanHooks.class.php(107): Pagingpro->doDialplanHook(Object(extensions), ‘asterisk’, 900)
#5 /var/lib/asterisk/bin/retrieve_conf(860): FreePBX\DialplanHooks->processHooks(‘asterisk’, Array)
#6 {main}a little guidance would be greatly appreciated.
Posts: 2
Participants: 2
@jeetzz wrote:
Hii…
I am starting a IVR, following is the code I have written up till now, the problem is it accepts but it doest send any thing.
This is the PHP page…
#!/usr/bin/php -q <?PHP require('/var/lib/asterisk/agi-bin/phpagi.php'); $agi = new AGI(); $agi->answer(); //required parameters and create variables $systoken = "3pvUYhSgf/x0Wc+CDAI="; $unique=$agi->request[agi_uniqueid]; $caller=$agi->request[agi_callerid]; $incoming=$agi->request[agi_dnid]; $texts = ""; $transferTos = "100"; //update variables $data = array( "systemToken" => "$systoken", "uniqueCallId" => "$unique", "callerNumber" => "$caller", "incomingNumber" => "$incoming", "text" => "$texts"); $data_string = json_encode($data); //CRM endpoint $ch = curl_init('http://liveasia.com/api/GetIvrResponse'); //create a new cURL resource curl_setopt($ch, CURLOPT_CUSTOMREQUEST, "POST"); curl_setopt($ch, CURLOPT_POSTFIELDS, $data_string); curl_setopt($ch, CURLOPT_RETURNTRANSFER, true); curl_setopt($ch, CURLOPT_HTTPHEADER, array( 'Content-Type: application/json', 'Content-Length: ' . strlen($data_string)) ); curl_setopt($ch, CURLOPT_TIMEOUT, 5); curl_setopt($ch, CURLOPT_CONNECTTIMEOUT, 5); //execute post $result = curl_exec($ch); curl_close($ch); $audio = json_decode($result, true); //get variables $audioPrompt = $audio["audioPrompt"]["text"]; $nextAction = $audio['nextAction']; //play response recording $agi->exec("AGI","live-tts.agi,\"$audioPrompt\",en"); ?>And this is the fiddler of the CRM…
When i given this command to it…
{“errorCode”:0,“nextAction”:“WaitForKeyPress”,“audioPrompt”:{“text”:“Thank you for calling HollywoodCars. Press 1 for immediate pickup at tesco kirkdale 151 kirkdale london. Press 2 for sainsburys supermarkets ltd22-24 market street huddersfield. Press 3 for 15 cross church street huddersfield. And for pick up at 8 chapel hill huddersfield please Press 4. or dial 0 for an operator”,“interruptible”:true},“maxDigits”:“1”,“wait”:“10000”}
Any heads up would be appreciated…
Jeet
Posts: 1
Participants: 1
@ugintl wrote:
Someone told me that using freepbx distro locally on a vm is not recommended. It is ok if you are using up to 10 phones, but above that it is not good. Is it true? Obviously, it is not easy to manage everything on-premise. Now My question is:
Can I upload the freepbx distro ISO image to a hosting like azure, aws etc etc? Which ones and how to upload?
Posts: 2
Participants: 2
@Shopgoblins wrote:
Heya,
New user here! I apologize if this has already been answered, I couldn’t seem to find the right terms in the search bar to find an answer.I work at an Escape Room and I set up a phone using a PAP2T and a Raspberry Pi. My goal was to have multiple phone numbers players could dial and a recording would be played when they do. Everything works great, dial tone, the phone picks up and plays a recorded .wav file. The issue is that there is a considerable delay between dialing a number and when the phone starts ringing. I’m not sure if this is normal behavior or if I have some setting wrong that is delaying the connection.
Any help you can offer is appreciated! Thanks!
Posts: 2
Participants: 2
@JessicaRabbit wrote:
My personal login is in the Admin group. When I click on anything it will only open a single entry, my login name and extension entry.
FreePBX 14.0.13.4
User Management 14.0.3.49
Posts: 1
Participants: 1
@SebastienPBX wrote:
Hi,
I’m looking to see if theirs an already existing way to achieve this with freePBX or if their’s someone who have done the same, or maybe some hint on how to do this.
Basically, we have a callcenter which is doing outbound and inbound call. With our current system (mitel) we are not able to properly categorize our outbound call. We have different sort of campaign for outbound call and we would like to get statistic data about these just like what we can get with inbound Queue.
What I was thinking was to do something like describe right there : www queuemetrics com/blog/2017/11/09/automatic-outbound-calls-tracking/
So the agent would log in the Queue and is call would be counted as made in the queue.What I’m not really sure is if I can use the dialing plan with this. For example, if the agent is calling is manager to ask a questions, I dont want this queue to be include in the “Outbound Sales Queue”.
Lastly, we have to match the call information with our custom CRM so I also need to get that data directly from the SQL or better from an API. Is that something possible?
The goal was to get something like
Inbound
Q - Inbound Customer 1
Q - Inbound Customer 2
Q - Inbound Customer …Outbound
300 Q - Outbount Campaign type 1
301 Q - Outbount Campaign type 2
302 Q - Outbount Campaign type …With the dialer*, if I use the Prefix 7300 + “Number to reach”, the call is counted as made in the “300 Q - Outbount Campaign type 1”.
*Dialer : I was thinking about using the Asterisk API to initiate the call from our application.
For reporting, I want to be able to extract the data of these queue directly from the database. Since it’s an open system, I don’t think should be an issue, but since I still not tested it I’m asking.
Posts: 1
Participants: 1
@mvogel4949 wrote:
I’m running FreePBX V14 with Asterisk 13.22
Current System Version: 12.7.6-1904-1.sng7I am running queues which may contribute but within a week my memory is pretty well pegged at max and I start to see degraded call quality. Below is my htop page after 6d uptime. Is there a way besides a weekly reboot(which I’ve been doing) to keep my memory under control? Thank you
Posts: 9
Participants: 4
@GeekBoy wrote:
Following this thread, it does not make a lot of sense. I am in the same boat now, but with FreePBX 14 (if that makes any difference)
@lgaetz is recommending to use a custom directory.
The confusion part is that is really not directly dialing an extension, but only you have to enter the IVR code, then it takes you to the directory, then you have to take extra time to dial at least three letters of a name.
So anyway, I just tried one extension to test, and it functions, but that seems like a lot of extra dialing of digits. Also, what if they name is forgotten, but on ly extension is recalled?
Surely, there is a better way to do this?
Update:
Thinking of something else, I was able to come up with using a custom extension and the Extension Routing module.
Using a dial string, and Extension Routing, the custom extensions are only allowed to use the outbound route for that trunk connecting the two PBXs together.
Seems to be working out
Posts: 2
Participants: 1
@NorColorNorName wrote:
Hello !
since two days i’m digging in all extensions.conf to find where the outbound callernumber is set.
I have a WebRTC softphone and i’m calling a random smartphone => The number “+33972XXXX21” appear.
I wanted to know where this number is set in the dialplan, to modify it in te dialplan with a custom one.Thanks by advance !
Regards,
NCNN.
Posts: 1
Participants: 1
@mvogel4949 wrote:
I’m running FreePBX V14 with Asterisk 13.22
Current System Version: 12.7.6-1904-1.sng7I am running queues which may contribute but within a week my memory is pretty well pegged at max and I start to see degraded call quality. Below is my htop page after 6d uptime. Is there a way besides a weekly reboot(which I’ve been doing) to keep my memory under control? Thank you
Posts: 9
Participants: 4
@v70ff wrote:
I am clueless as to why the calls are being dropped after the 10s. Any sort of knowledge is welcomed.
Posts: 2
Participants: 1
@eggythetech wrote:
Hi all
I have a FreePBX Distro Server and Sangoma A400 with 6 FXO and 2 FXS.
it was working fine all calls where being received and sent fine until I did the latest Update.
PBX Firmware: 12.7.6-1904-1.sng7
PBX Service Pack:1.0.0.0
I can Call internal and external.
I can not call from an outside number to the Analog line it just rings. the system will not answer.
I have confirmed that the line works with my buttset.
How do I see if the wanpipe driver can see the incoming call?Thanks,
Greg
Posts: 1
Participants: 1
@GeekBoy wrote:
Following this thread, it does not make a lot of sense. I am in the same boat now, but with FreePBX 14 (if that makes any difference)
@lgaetz is recommending to use a custom directory.
The confusion part is that is really not directly dialing an extension, but only you have to enter the IVR code, then it takes you to the directory, then you have to take extra time to dial at least three letters of a name.
So anyway, I just tried one extension to test, and it functions, but that seems like a lot of extra dialing of digits. Also, what if they name is forgotten, but on ly extension is recalled?
Surely, there is a better way to do this?
Update:
Thinking of something else, I was able to come up with using a custom extension and the Extension Routing module.
Using a dial string, and Extension Routing, the custom extensions are only allowed to use the outbound route for that trunk connecting the two PBXs together.
Seems to be working out
Posts: 2
Participants: 1
@mniess wrote:
I noticed the mysql database size is growing quite quickly even on a freeobx instance that doesn’t do much. Is there a way to clear unneeded data or a setting in Freepbx to do that? I’m using the default distro.
Edit: Specifically my asteriskcdrdb is quite large with the cel table at around 6GB and the cdr table at 2GB. Can I just clear old records or are there any repercussions?
Posts: 1
Participants: 1
