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Error "Whoops\ Exception\ ErrorException" when rebuilding basefile edit config

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Hi,

Since last week, I seem to be getting this “Whoops\Exception\ErrorException Maximum execution time of 30 seconds exceeded” error when trying to save and rebuild a config for basefile edit… I have had multiple errors such as the below… if anyone can advise me it will be appreciated!

File:/var/www/html/admin/modules/endpoint/functions.inc/functions_views.php:371
File:/var/www/html/admin/modules/endpoint/Endpoint Common.class.php:2277

Have tried to look at similar issues however no luck…

Thanks…


No queue callback or announcements

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Updated cqplus module resolved issue.

[HOW TO] Relatively simple interface to TimeTrex for FreePBX/PIAF users

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You have to add an administrator that you want to use for the Time Accounting stuff in FreePBX. This (with whatever name you want to use) is your Time Accounting user in TimeTrex. Once you get logged in on that user in TimeTrex (the first time) you will be forced to reset the password on that user.

After you get that squared away, you need to set this user information in the AGI script so that your user can log into TimeTrex and get this going. For completeness I also add the user as an Admin user in FreePBX and give them the same credentials.

To - one last time - you need to set the Username for the TimeTrex script in TimeTrex (as an admin), in the script itself (TIMETREX_USER), and in the Administrators list in FreePBX.

Remember, you have to log into TimeTrex as the user in order to verify the user and make it so that the user can interact with TimeTrex.

Really wierd issue

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Nope - audio is always from the handset to the Asterisk process to the external caller. Asterisk is a back-to-back user agent, so there is no direct connection from the user to the external caller ever.

This is sounding more and more like either an Asterisk keep-alive problem or a router port session timeout problem. The fact that you interrupt the connection with an On-Hold action just amplifies that. Look through the archive for “30 seconds” and look at some of the threads that talk about one-way audio after a period of time.

Error "Whoops\ Exception\ ErrorException" when rebuilding basefile edit config

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There’s a PHP config file (/etc/php.ini)? that you might need to tweak (increase the wait time by another 30 seconds or a minute) and see if that solves the problem.

How-To Guide for Google Voice with Freepbx 14 & asterisk gvsip, Ubuntu 18.04

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Thanks alot this and a reboot fixed the problem!

Codec translation error on routing

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Hi Dave, I was in doubt about codec but why not to do another try, all codec/one at time codec where enabled without change in response, about other point we check one at time

Unimplemented is referred to registration, device execute dialing on PSTN, answer to PSTN line, this seems bad but not sure can be issue on stack. These messages are in response to registration string when no channel is in place.

And here there where two point :
1 as posted trunk is SIP not PJSIP
2 ok why is peer unknown?

I hope this can be close the real problem, I ask’d for help, I also have no idea what it mean and no record on help nor documentation.
At this point stack manipulation where pointing at SPA3000, after this connect channel to play message and say “dialed number is not in use, please check and retry”, “congestion tone” then hang up.

connection is active, so it answer the call, registration is wrong. From status chan sip registry:

And also this anonymous connection is OFF:

At end, too much time elapsed from first time I was using Asterisk, this is beautiful interface but hide to me too much details I was used from command line.
I don’t feel useful try to debug stack to pinpoint where SPA3000 offend.
After moving trunk to HT503 I post what happen. This is not readily available to test and we need wait next WE I can go to site.
Regards
Roberto

Error "Whoops\ Exception\ ErrorException" when rebuilding basefile edit config

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Hi, thanks for the reply…

I tried what you said and extended to 90 seconds, however after a reboot I now get the error " Whoops\Exception\ErrorException

Undefined index: type
File:/var/www/html/admin/modules/endpoint/functions.inc/functions_grandstream.php:301"


Queue hold times and announcements when past X time

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  1. What is the formula used for determing the wait time? It is it simply the average wait time for the last 24 hours or is it a more advanced calculation?

  2. Is there a way to play a specific announcement after a caller has been on hold for a while.
    Example we have an announcement to play every 1:15 min but if they are on hold for more than lets say 5 min. we want to play a different message apologizing for the long hold.

[HOW TO] Relatively simple interface to TimeTrex for FreePBX/PIAF users

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Thanks for the detailed reply.

I already tried all the steps that you mentioned and created the users as an asterisk manager and freepbx admin user. Both of those users with full freepbx and asterisk manager rights and permissions. I created the user in the ampusrs and usermanager. I tried both and individually with the same results.

I just tried it again after you posted your steps to confirm everything :frowning:

Can't dial trunk to trunk extensions

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Found the issue, somehow the NIC card was getting a static and DHCP IP, after hardwiring to manual, the problem went away.

Thanks so much for all your guys help!

Codec translation error on routing

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reflection about things…
Channel(recvip)
Why unknown? IP must be known otherwise it cannot route audio play to 192.168.1.210 (SPA3000)

On previous this appear after some stack manipulation when channel is active, so what hell is this?
Saturday I can try wireshark on PBX network. Definitive suspect is SPA3000 emit some offending code.
Not sure to remember too much about protocol details nor be able to still use wshark, again too much time elapsed, if it may be of some use I can try capture traffic and packet too.

A few new modules to play with

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This topic was automatically closed 7 days after the last reply. New replies are no longer allowed.

Disable Intercom Auto Answer

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Hi, are you sure this behave to FreePBX and not to DECT related settings?
From your writing I understand that:
DECT answer incoming call
Desk just ring
if so did you checked all DECT settings on both base and mobile?
About 2 seconds ring can you share extensions settings?
Regards

Call Parking and BLFs - Unexpected Behavior - S500

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If I create a few BLFs for parking spaces (71, 72, 73, etc.) all works well. I can transfer calls to them and pick them up as expected, however, if I’m going to pick someone back up out of the a parking space and my phone happens to ring at the same time, instead of picking up the parked caller I end up merging the calls of the person in the parking space and the person whom was ringing my phone.

Does anyone have any suggestions on how I could prevent that from happening?


Turn off MoH when caller listens to announcement

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Caller is in the queue listening to moh. The announcement comes on every 1 minute. The music keeps playing even while the announcement is playing.

How do we shut off the music while the caller is listening to their place in line and the announcement?

Transfer to Voicemail not a valid extension

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I am trying to setup transfer to voicemail. I type in ##* using DTMF then the extension and I get “I am sorry that is not a valid extension”. Then it says enter a valid extension which I do at 110. Then the logs show:

This extension does not exist or no password is set

I am dialing for example: ##*110#

The logs show “Invalid extension 0 entered”

Is this a bug? I don’t see any feature codes causing the issue. Fresh install as well

Callcentric + PJSIP

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This topic was automatically closed 7 days after the last reply. New replies are no longer allowed.

Park and announce not working after upgrade to Freepbx 14

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issue resolved. Had to pay for upgrade

Voice IVR - Any new updates or software to use?

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I said that before I actually watched this video from @tm1000 at Astricon.
He integrated FreePBX 15 with Amazon Lex. So it’s definitely doable in FreePBX 15.

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