I’ve just found the config extensions_override_freepbx.conf. and i dont know why it is using… not me configured the server.
i’ve added the line
exten => recordcheck,n,MixMonitor(${MIXMON_DIR}${YEAR}/${MONTH}/${DAY}/${CALLFILENAME}.${MON_FMT},a${MONITOR_REC_OPTION}i(LOCAL_MIXMON_ID)${MIXMON_BEEP},${MIXMON_POST})
and calls started to record again ))
Bonjour Nick.
This has been broken for a while, and is the same issues as is tracked in this ticket:
https://issues.freepbx.org/browse/FREEPBX-19194
I attempted to fix it some time ago, but couldn’t locate the issue. When time permits I will take another look. Issue appears to be with $trunk_info array in Superfecta.class.php.
Thank you Tom,
I’m using asterisk 15.7.2 and 16.3. I’ll test your suggestion and back to you.
Regards,
This topic was automatically closed 7 days after the last reply. New replies are no longer allowed.
Bonjour Lorne! 
It makes one wonder why this older ticket was recently (2 months ago) committed:
https://issues.freepbx.org/browse/FREEPBX-15699
The ticket does have a status of QA NEW so I guess it has not yet gone through QA…
I guess a “depends on” should be put on FREEPBX-15699 indicating that it “depends on” FREEPBX-19194…
I might try to take a look if time permits but I doubt I will find it if you didn’t as I nowhere know CID-Superfecta and FreePBX as much as you do and I rarely do any PHP (and only know the basics)…
Thank you very much and have a nice day Lorne!
Nick
Hmm, so my memory failed me once again. Ticket you linked was work I did about a year ago (1 yr exactly today), and it was fixed at the time, but now it’s broken again. Let me see if I can figure out why.
There isn’t a problem with your setup, when they press “transfer” the caller gets placed on hold, and the system starts playing MOH, as soon as the call actually gets transferred, the call is no longer on hold and the caller hears a ringback tone.
This topic was automatically closed 365 days after the last reply. New replies are no longer allowed.
Locate the file:
/var/www/html/admin/modules/superfecta/sources/source-Send_to_email.module
and change this line from:
$from_did = $this->trunk_info['did'];
to:
if(isset($this->trunk_info['dnid'])){
$from_did = $this->trunk_info['dnid'];
}
Assuming that works, update the ticket you identified with details.
There is a REST API in FreePBX 14, but it is NOT the GitHub module. It is there to support (among other things) the UCP and Sangoma’s phone module (whose name escapes at this moment). The GitHub module may have been the genesis for the embedded REST API, but it is not the same thing.
Hi,
We have a FreePBX system running with Sangoma phones. Everything works normally, however we are now trying to use click to dial.
I have setup Jolt chrome extension and got it to work with a few tweaks in the php file, however the Call information and history on my phone screen only shows the initial leg of the call (the received call). It doesnt show the actual call that was dialled when I picked up my receiver.
I thought i had messed something up in my Jolt php file so I installed FOP2. It dials out ok but still only shows the received leg of the call on screen and in the phone history.
Has anyone else found this and fixed it?
FreePBX 14.0.5.25
Thanks
Dave
The REST API (restapi) module has been around for a while. It once was a dependency for the Phone Apps (restapps) module, but not since ver. 12, and as a result has been hardly touched since.
Right because the phone didn’t dial it. The call doesn’t come from the phone, it comes from the PBX. So you’re phone is only going to have the call it knows about, the one it received from the PBX. Once it is answered, the PBX is dialing the other number, not the phone.
If you want this to be logged at the phone level, you would have to use the phone’s API (if they have one) to send the request to the phone so the phone makes the outbound call and then it will log the destination number as an Outbound Call in the history.
The trick is to set the Caller ID of the inbound leg of the call the same as the outbound destination. It will still show as an incoming call in call history, but if you attempt to dial the call from the history it will go out to the destination.
You are in the testing repo. yum remove sangoma-devel
This topic was automatically closed 365 days after the last reply. New replies are no longer allowed.
Thanks for the super speedy replies, I understand now why its only showing that leg of the journey.
I just need to work out the behind the scenes coding for my php file in Jolt. (I’m unsure how I could play with FOP2 code just yet)
In the jolt php I have some lines like these
$callerid
and
$strCallerId
The original code was
$strCallerId = $callerid." <$number>";
but this did not connect to my pbx, I had to change it to this next line to get it to connect.
$strCallerId = $ext;
I shall keep looking to see how/where I need to change my code. Thanks for the help so far,
Dave
Update - Not sure why, but while playing with the php code again to find the error messages, the original piece of code started to work.
$strCallerId = $callerid." <$number>";
This did exactly what I wanted and what you said should happen. It still shows as an incoming call but has the number dialled available in history.
Thanks for your help, now I will see if I can find where I can tweak this in FOP2
Dave
This topic was automatically closed 7 days after the last reply. New replies are no longer allowed.
