SETTINGS -> Advanced Settings -> Call Record.
This isn’t controlled in a module, it’s core and controlled by core system systems.
SETTINGS -> Advanced Settings -> Call Record.
This isn’t controlled in a module, it’s core and controlled by core system systems.
Hmmm… well, in looking over everything, dnsmasq has been running, and 127.0.0.1 is the first in my etc/resolv.conf
So, the question still remains… why did my phone system lose its mind when my internet was interrupted today?
16.3.0
After yum upgrade make sure to restart asterisk otherwise you’re using old code!
And it was answered. Chan_SIP didn’t like not being able to issue a DNS resolution. But without seeing any logs or activity from this time frame it is going to be hard to say what was causing the issue.
Correct Andrew.
Just out of curiosity…could this be accomplished via a custom application and then assign that application to a feature code via FreePBX?
As was mentioned, even if you were to designate a divert destination and tie it button, you’d still have to deal with the race condition of what happens when you divert a ring-group call to VM and someone tries to answer the call at the same time.
Chan-SCCP gets around this by having the code running on the server and not at the phone. It will be harder to get right in SIP.
Is there a way to force the system to tag a voicemail that’s been sent from one user to another?
IE; I send a recorded voicemail from extension 123 to extension 345 in a way that 345 knows that 123 sent the voicemail.
Thanks,
Dennis
The email system is Comedian Mail, which I dont has a feature like that.
FreePBX 14.0.10.3
Asterisk 13.22.0
Getting error when trying to update with Admin:Updates module:
RPM command errored, Delete /dev/shm/yumwrapper/* and try again. Exit code 1 - see FreePBX log for more info.
I deleted the contents of /dev/shm/yumwrapper and tried again but still get the error. Can anyone suggest a way to fix?
Also displayed on screen:
So I switched to asterisk 13.26.0 and still get the same page.
I fixed my error by removing “/volume1” from my server address. Here was my initial error message that was confusing. FreePBX was trying to create a new Volume which of course it could not. After removing the /volume1 it worked perfectly. So my new path was /PBXact_Backup
Creating backup…
Storing backup…
Creating directory ‘/volume1/PBXact_Backup/POMPs_Full_Backup’
Directory ‘/volume1/PBXact_Backup/POMPs_Full_Backup’ did not exist and we could not create it
ftp_mkdir(): /volume1: Permission denied.
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Go to Admin --> Module Admin and disable the “Sangoma Property Management” application. That should clear the “You have to be kidding” warnings. Go to Settings --> Voicemail Admin --> Settings --> Limits and be sure that “Max Message Silence” is less than “Min Message Length” noting that one is in milliseconds and the other is in seconds. I set Max Message Silence to 4000 and Min Message Length to 5. That should clear the maxsilence/minsecs warning.
Thanks Tom. I’m going to have to look deeper. In my SETTINGS…Advanced Settings there is no Call Record anywhere. Must be missing a module or something. This is all I have in that section. I didn’t paste any of my entries in this thread, just the options available.
FreePBX Advanced Settings
IMPORTANT: Use extreme caution when making changes!
Some of these settings can render your system inoperable. You are urged to backup before making any changes. Readonly settings are usually more volatile, they can be changed by changing ‘Override Readonly Settings’ to true. Once changed you must save the setting by checking the green check box that appears. You can restore the default setting by clicking on the icon to the right of the values if not set at default.
Advanced Settings Details
Display Friendly Name? True False
Display Hidden Settings? True False
Display Readonly Settings? True False
Override Readonly Settings? True False
Asterisk Manager
Asterisk Manager Password?
Asterisk Manager User?
Backup Module
Email “From:” Address?
Call Flow Control Module
Hook Time Conditions Module? True False
Camp-On Module
Maximum Active Camp-On Requests?
Non Extensions Callee Policy?
Only Use Default Camp-On Settings? True False
Caller Policy Default?
Callee Policy Default?
Caller Timeout to Request Default?
Max Camp-On Life Busy Default?
Max Camp-On Life No Answer Default?
Default Time to Ring Back Caller?
Default Caller Callback Mode?
Default Max Camped-On Extensions?
Default Callback Alert-Info?
Default Callback CID Prepend?
Announce the Callee Extension? True False
Default Callee Alert-Info?
Default Callee CID Prepend?
Default Max Queued Callers?
BLF Camp-On Available State?
BLF Camp-On Pending State?
BLF Camp-On Busy Caller State?
BLF Camp-On Recalling State?
Developer and Customization
Always Download Web Assets? True False
Debug File?
Developer Mode? True False
Disable FreePBX dbug Logging? True False
Disable Mainstyle CSS Compression? True False
Leave Reload Bar Up? True False
POST_RELOAD Debug Mode? True False
Provide Verbose Tracebacks? True False
Use Packaged Javascript Library ? True False
Device Settings
Show all Device Setting on Add? True False
Require Strong Secrets? True False
Remove mailbox Setting when no Voicemail? True False
SIP canrenivite (directmedia)?
SIP trustrpid?
SIP sendrpid?
SIP nat?
SIP encryption?
SIP qualifyfreq?
SIP and IAX qualify?
SIP and IAX allow?
SIP and IAX disallow?
SIP and DAHDi callgroup?
SIP and DAHDi pickupgroup?
Dialplan and Operational
Block CNAM on External Trunks? True False
Call Forward Ringtimer Default?
CW Enabled by Default? True False
Disable -custom Context Includes? True False
Ditech VQA Inbound Setting?
Ditech VQA Outbound Setting?
Enable Custom Device States? True False
Extension Concurrency Limit?
Feature Codes Beep Only? True False
Force All Internal Auto Answer? True False
Generate Diversion Headers? True False
Internal Auto Answer Default?
NoOp Traces in Dialplan?
Occupied Lines CW Busy? True False
Only Use Last CID Prepend? True False
Polling Interval for Stopping Asterisk?
Use bad-number Context? True False
Use Google DNS for Enum? True False
Waiting Period to Stop Asterisk?
Display CallerID on Calling Phone? True False
Display Dialed Number on Calling Phone? True False
Use Automixmon for One-Touch Recording? True False
Conference Room App?
Follow Me Module
Create Follow Me at Extension Creation Time? True False
Disable Follow Me Upon Creation? True False
Default Follow Me Ring Time?
Default Follow Me Initial Ring Time?
Default Follow Me Ring Strategy?
GUI Behavior
Abort Config Gen on Bad Dest? True False
Abort Config Gen on Exten Conflict? True False
Check Server Referrer? True False
Include Server Name in Browser? True False
Report Unknown Dest as Error? True False
Require Confirm with Apply Changes? True False
Show Categories in Nav Menu? True False
Use freepbx_menu.conf Configuration? True False
Use wget For Module Admin? True False
Dashboard Info Update Frequency?
Dashboard Max Calls Initial Scale?
Dashboard Stats Update Frequency?
Queues Module
Set Agent Name in CDR dstchannel? True False
Use MixMonitor for Recordings? True False
Hide Queue No Answer Option? True False
Asterisk Queues Patch 15168 Installed? True False
Generate queuenum*/** Login/off Codes? True False
Ring Group Module
Display Extension Ring Group Members? True False
System Setup
FreePBX Log Routing?
Disable FreePBX Log? True False
Log Verbose Messages? True False
Send Dashboard Notifications to Log? True False
FreePBX Log File?
PHP Error Log Output?
User & Devices Mode?
Allow Login With DB Credentials? True False
User Portal Admin Username?
User Portal Admin Password?
Asterisk VMU Mask?
Browser Stats? True False
FreePBX Web Address?
Use Google Distribution Network for js Downloads? True False
Convert Music Files to WAV? True False
Dashboard Non-Std SSH Port?
Recordings Crypt Key?
Use Old Parking Patch? True False
jQuery UI Version?
jQuery Version?
Time Condition Module
Enable Maintenance Polling? True False
Maintenance Polling Interval?
Voicemail Module
Provide IMAP Voicemail Fields? True False
Real problem here is, as it was pointed out earlier, you’re on a version of FreePBX that is 5+ years old. So it may not exist at that point.
It was recommended you update. I concur with this because this means you’re also on an outdated and unsupported version of Asterisk.
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Very true. Most other platforms like Broadsoft, etc have firmware versions for their platform with various vendors because of exactly that. They use custom SIP headers and other features to do things like this with. Not only does the platform need to do it, the phone needs to understand and know what to do.
So yeah, generally wanting specific features like this require a specific platform and the phones that can support that platform to go with it. Or a really good understanding of SIP and a proxy because with a proxy, this could be done and eliminate the race condition Asterisk would have.
You are exactly right Tom. We had loaded the new copy on our backup PBX server to do testing and such and still have not converted. You don’t know if there’s an “easy” way to convert the files on this old version to the new version do you? When we first started studying the issue it looked like we would have to manually rebuild all the extensions/trunks, etc. If someone knows a good way to convert that would help.
Again, I apologize for not catching that point on the old version. It’s not something I work on every day unless something breaks.
Mark