Perhaps instead, put your groups into a queue? You can apply the same ring group type strategy, but then you can install some free reporting software (asternic / FOP2) have a pretty good free one, and report that way?
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CDR Reports/Useful info
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Guide to using custom config files?
This is straight 100% Asterisk, so if there is a guide it would be in the Asterisk wiki. Similarly bugs would be in the Asterisk issue tracker.
That said, your lines work for me, adding the following to pjsip.endpoint_custom_post.conf:
[2009](+)
fax_detect_timeout=15
fax_detect=trueAfter doing a core reload and param is set:
[root@lgaetzdev2 ~]# asterisk -x "pjsip show endpoint 2009" | grep fax_detect
fax_detect : true
fax_detect_timeout : 15↧
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New instalation FREEPBX on VM, 401 not authorization
You have named your trunk from-trunk which may be permitted, but could also conflict with an existing dialplan context. You might try changing the trunk name.
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Upgrade 6.12.65 to 10.13.66
User Manager module failed to install:
fwconsole ma install userman↧
RMS Alerts Page never renders
If you haven't done so, open a commercial support ticket:
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Upgrade 6.12.65 to 10.13.66
Hi,
Thank you very much for your attention time and your answer, but...
Yes this option - fwconsole ma install userman - could solve the problem but not in my case as when you write fwconsole you get nothing, I tryed to find where fwconsole could be in my computer but the answer is that I do not have the fwconsole in it, even the directory that it suppose to be I even do not have, so I made some mistake most likely in the upgrade process or this is something that allways happens when you try to upgrade from 6-12-65-to-10-13-6 using a terminal, in my case I was using Putty.
Another problem (just to - maybe - help others users as I'm a newbee in Freepbx) is that the computer that I was using -Vmware - had a too small configuration to do the upgrade - 1gb ram - 10gb hard disk.
Anyway I read here in forum that to do the upgrade it will be better to use the Gui console and can take about 3 hours but works.
Thats it
Thanks again
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Fwconsole file is not accessible
Installed FreePBX 14 with asterisk 13 logged as root and get an error
** CRITICAL SYSTEM ERROR **
Unable to generate MOTD.
The /usr/sbin/fwconsole file is not accessibleYou are likely to experience significant system issues.
[root@freepbx log]# locate fwconsole
locate: can not stat () `/var/lib/mlocate/mlocate.db': No such file or directory
How can I fix that?
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Upgraded container on VULTR, how to expand disk space on FreePBX?
6No , it I elite e the Archlinux is they offer has gpargsd and clonezilla on its live desktop
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Disable activation check
So how can i do please
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Upgraded container on VULTR, how to expand disk space on FreePBX?
So there is literally no way to resize a FreePBX partition once installed? 
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Help: phone in a subnet other than that of FreePBX
no, but voip-router is already used for voip calls and it works
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Fwconsole file is not accessible
That message occurs when FreePBX installation failed, due to a network problem or HW issue.
You need to perform installation again.
Thanks
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Build Your Own Business Communications Hosting Solution Using PBXact SaaS
Good talk, Thanks for taking the time to tell me I'm going the right direction.
Anyone need help to set this up, I can help if you want to use KVM and Proxmox.
About a year ago when I moved from a shared hosting solution located in the midwest and my hosted PBX's keep having issues. I thought because I purchased from this vendor I wouldn't have issues. I was wrong.
So I got into a new hosting facility with 3 dedicated servers running Proxmox VM. Proxmox VM allow you do manager the VM or virtual machines. (when I started this I thought VM stood for voicemail. Shows my age).
I tried for many months to make the HA or High Availablity to work that way I think it should, then I decided that HA should be installed only as directed in the manual, using dedicated machines and not VM's. You can make it work, but it's not worth the benefit.
Now I have VM's, one per customer that is backed up to a central location. This backup can be restored, turned on by any server in the cluster with just 2 clicks. Same IP address and all the FreePBX licenses will continue to work. Starting up with a backup takes less than 30 seconds after you press the button. A bash script could run the code, but you don't want to flip-flop and systems.
So this was working well, but now I have the customers that have dynamic IP addresses and I need to open the firewall. This is where SBC comes in to protect my FreePBX server and to hide my partners IP addresses.
This way I send my customers with on-premise equipment to this IP address that I own and control.
I'm in the process of using SBC to process all my traffic and to firewall off my FPBX's.
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Fwconsole file is not accessible
No Internet when you tried to install the distro. The distro never installed.
Is this on hardware or a VM?
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RMS Alerts Page never renders
I have had the same issues for a while now. All other parts of the website works.
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Upgraded container on VULTR, how to expand disk space on FreePBX?
Not quite true, but unless your bash fu is exceptional and you have nerves of steel I suggest you go the alternative live linux boot iso route.
Just try it, take a snapshot (in case) , boot the Archlinux iso, gparted your /dev/vda remove the iso , it will reboot into your resized machine.
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Help: phone in a subnet other than that of FreePBX
Not an Asterisk type of issue, this is an IP routing issue.
Can you ping? No, then you need to fix that before you continue.
Us a laptop connect to the same LAN to troubleshoot your LAN connection before you involve the PBX.
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Hide real caller Id and display different number to the callee using Asterisk freePBX
You can purchase a SIP trunk that allows you to control the outbound caller ID number.
Most carriers prohibit this type of caller ID manipulation, but some still allow it to happen.
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Inbound CID manipulation similar to Google Voice
Good morning
I am using FreePBX 14 on Asterisk 11
Is there a way to manipulate the inbound caller ID on specific routes to show the called DID as the caller ID? Google Voice allows you to do this so you can determine when someone is calling your Google Voice number that forwards to your cellphone.
The powers that be do not want to use a softphone to accomplish this when their phones are forwarded to cellphones so I was hoping that there was a way to manipulate this in FreePBX
Thank you!
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FreePBX incoiming calls not working
That's not enough data.
what do you get when you run this from the CLI
show sip peers
Do you see both of your sip trunks?
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