FreePBX 13 / Asterisk 13.13.1
(Trial Qxact)
Hi,
I have the same problem.
If I run this php script manually, it is OK (the data appear in Qxact reports).
But if I make new calls, they dont appear in our report again.
How can I solve this problem definitely?
Why does not the script run automatically for the data to appear in the reports?
I will wait your help.
Thanks,
Marcelo Henrique
Dhcp shows its getting the IP address. No difference in device behavior.
I use a cisco spa112 direct connect to a valcom 1020c . I also use the paid-for Paging and Intercom module for the bell tones .. registor the spa112 as an extension and select it in Paging and Intercom, In spa112 > voice>line X> set SAS to enable.You need the get the tip-ring output below 600 ohm of it to work (off hook). with Valcom Horns(V1030c) I use a TY-146p transformer.. To connect
Direct to a speaker I use a SnomPA1 it has a 4 watt output
Hola Gerardo, buen día
Si me puedes ayudar por Team Viewer con esto por favor?
Muchas gracias
The script runs every 5 minutes from a cron
Dear all
I have installed asterisk 13.17 and freepbx 13 according to
FOP-InstallingFreePBX13onUbuntuServer14.04.2LTS-120817-1342-4146.pdf on my zynq SoC.
after installation, connecting from browser to freepbx is right but freepbx can not connect to asterisk.
the error is FreePBX cannot connect to Asterisk via AMI.
could you help me please.
best regard.
The error said it required pear getopts not peardb. Nothing uses peardb anymore. We don't even install it in the distro.
We do require pear getopts and we have a warning that says it's not installed.
Might be that pear was an older version
I have purchased sysadmin, and other upgrades from the community. However, I would like to upgrade to the latest release, but I am not sure if it will have registration issues for all my commercial modules. Is there a simple way to do the upgrade to my system?
Please let me knwo and thank you in advance.
I think it's pretty clear what you want.
You want to dial extensions on a remote system (3cx) from your FreePBX IVR.
The threads describe some of the ways you can do that.
E.g. set your 3cx extensions as custom extensions on freepbx (in custom dial section put: sip/siptrunkname/remotepbxextension)
or as miscellaneous destinations.
OK. I'll try that. Thanks 
So after re-configuring all of the dahdi trunks and extensions i am still not further than before. To test the whole signalling/voltage possibility I ran the command: wanpipemon -i w1g1 -c astats -m 1 in the root CLI and watched for voltage changes when i took the phone off the hook and there was none which kind of matches my theory of there being a signalling issue.
I am not really sure what else I can do to fix this, i am guessing there is just some config somewhere that needs changing.
seems more like php-pear was not installed at all. getopts is installed (now).
That's a pretty specific request. I doubt very many people have played with that, but FreePBX (when using the Adaptive Firewall) implements something very close to that already, so setting the number reasonably high (say 20 or 25) should be OK. If you run into problems with people not being able to connect, try running the number up a little bit.
Problem solved! The audiocodes 310HD puts the DESTINATION address of Sip server in the wrong field. It did put in the SIP REGISTRAR field. The digitmaps and dialplan can be empty. Another mistaken is the EPM just has DIALPLAN field, and Audiocodes understand this field like digitmaps, and put one wrong NULL in dialplan rules.
I just fix the config file in the EPM editor, and now, my Audiocodes could be provisioned! 
Hi Guys is it possible to nat different port other than 5060 exposed to firewall . ie., I want external port to be something like 4532 and it should be routed through nat to my asterisk server in lan which will listen to 5060 .
I don't think this is a FreePBX configuration issue. This seems more of a hardware issue, either a faulty card or a DAHDI configuration issue. Since you doublechecked DAHDI conf, I would say definitely a faulty card or module. Can you try to swap the modules on the card? If after swaping the modules FXS ports still won't work, then it is definitely a module issue, provided the power connector is properly connected and working. If the FXS module works after swaping and the FXO stops working, then definitely a card issue.
It depends on the router, but I can't think of anyplace in the system that shouldn't support that.
Okay thanks. I'll give it a go when i can, may end up being a couple of days before i can do that
Hi
Thanks for your prompt reply ...okay now that we know the problem is not with the outbound dial pattern.
I went ahead and watched what actually happens in the background while I am placing a cool to the specified number in the fund the following error:
[2017-08-14 16:05:48] ERROR[26563]: phone_message.c:1645 build_dialplan_routing: Unable to build dialplan routing - invalid license
[2017-08-14 16:05:48] WARNING[26563]: res_digium_phone.c:1898 reload: No Valid DPMA License found. Module is loaded but disabled. Please reload module once valid license is installed.
It seems that the module is enabled on my free PBX so how would I go about renewing the license thereof?
Do you have any ideas how to correct this? If you do help would be greatly appreciated please.
Kind regards,
Mark Stoermer
I face once way audio issue or no register in pfsense .