What are we thinking about for 2.11


How about retaining the account code for a CDR record of the extension that has the forward set?

Ring groups to have account codes too… so external calls out of them can be tracked to appropiate dept.

As said, Webmin is a great tool and well developed. Jamie Cameron works for Google too. Jamie and Joe got together and created Virtualmin and Cloudmin too.

Their products are great to manage web/email sever(virtualmin) and virtual servers(cloudmin. I use virtualmin a lot.


This contributed module may do what you are asking for, preserving account codes on “Call Forwarded” calls, which normally translates to any call that came into the system from the outside and is being sent back out, CF, Ring Groups, Follow-Me, etc.


Geeky I know but I would love a module or something that enabled watchdog features.

IP address to ping: xxx.xxx.xxx.xxx
ping frequency in seconds: xx
number of failed pings before action:xx
number to call upon fail:
recording to play:
destination after recording:

the ability to setup several ip’s to monitor.

setup alerting when certain trunks or extensions fail to register
high cpu alert
low disk space
option to email and or call number etc.


I love the new ideas planned for 2.11. I sure appreciate the changes in the 2.10 version that I’m using. It was such a great job!

Since we are talking about suggestions, one thing I’ve always wished for was the CNAM transmitted along with the number on forwarded calls. Right now, only the number shows up. I have a ring group setup where 2 extensions ring plus an outside phone number. The outside phone number never sees the name. IE:

Extension 33 rings with “DOE JOHN” <9994445454>
Extension 34 rings with “DOE JOHN” <9994445454>
9998885555# rings with <9994445454>

There may be a technical dead end as to why CNAM is not transmitted with the number. Hopefully there’s a way around it as it would be an excellent enhancement.

The other thing that would be nice is email notifications for extensions that become unregistered. Right now I find myself constantly checking the SIP INFO to ensure everything is registered ok.

Thanks for your time.

Guys, You are doing an excellent job.
One area I see that needs improvement is the Voicemail Admin module.
It would be awesome if there was a way to export to a csv the details of all voicemail boxes. I sometime find that wen folks have voicemail going to email and to the phone don’t check the phone and it fills up and stops taking messages.
A report that I can print out or if the system could send an email stating that the voicemail box is full would be perfect.

Mike Zenft

Hey guys, I really love FreePBX, helps a lot with asterisk. The only thing that is annoying me is that my asterisk somehow shuts the dahdi channels down when ever I apply changes in FreePBX. I am on asterisk 1.8 and FreePBX 2.8 at the moment as upgrading to 2.10 step by step put the whole system to an unrecoverable state, so I decided to stay. Maybe there is an easy fix to that.

This would be on my ARI wishlist…

  1. Do away with the folders, and the convoluted process of placing a checkmark next to the recordings you want to move, selecting a folder, and then clicking the move_to button. Instead, implement one or both of these two concepts:

A) Give the ability to drag and drop recordings into a folder.
B) People sometimes look at their voicemail the way they look at their to-do list. Therefore, Give people the ability to drag and drop voicemails up and down the list.

  1. Enable the voicemails to be delivered as secure podcasts. I have a custom script that I use today that iterates through the .WAV file and creates MP3’s and delivers those as a secure RSS feed. I wouldn’t want to lose that ability.

  2. Retain the ability to store the voicemails as separate files with the MSG0001.WAV, MSG0002.WAV naming convention. I have processes that move and archive voicemails based on that.

  3. Less important, but a consideration is that for multi-tenant applications, ensure the “forward to” functionality can be hidden in the UI as to not expose everyone’s name.

  4. Lose the “Pager notification” verbiage. Give the ability to forward the voicemail to more than one email address, and to text the notification.

  5. Allow the ability to email a voicemail from a smartphone to an extension. In other words, using the Smartphone’s voice recorder, someone can dictate a message, and email it to someones Asterisk account.

Probably sounds like I’m coming out of left field, but what about a giant leap to just FreePBX 3? It’s always been a lucky number for me. The drastic change to the look and feel may have warranted a 3 to begin with. It really is kind of a new animal. Plus, it would be nice to have a numbering pattern more unique and separate from the Asterisk versions to lesson the confusion when talking about the different species. ;]

FreePBX 3 turned into Blue.Box. I think it’d confuse a lot of people if we jumped to v3. Which is why (for now) we are staying on the 2.x line

It would be nice to choose a destination for a particular trunk instead of just inbound caller ID.

For instance, in a system that answers for multiple tenants or systems we may have 3 or 4 POTS lines that we want to direct to individual IVR’s or ring groups.

The correction of the blindtransfer/uniqueID issue mentioned in this post:


@mdoran, would it work to use the inbound DID number setting on inbound routes to direct calls coming in from particular POTS lines (each which would have a separate DID) to your preferred internal destination? If the lines were dialed separately by outside callers (not part of a hunt group), that might work.

Similarly to mdoran’s feature request, the reverse would also be helpful. For outbound routes, be able to select a particular outbound trunk based on the source extension, not just pattern-matching the outbound destination number.

This is helpful if a particular device should always use a particular trunk to dial out, e.g. an analog fax machine that always should use a T.38-enabled SIP trunk, or a business group that should be assigned just a subset of trunks based on their source extensions.

We found this workaround to accomplish this with the 2.10 release:

  1. create a custom context in extensions_custom.conf that adds a special numeric prefix to dialed numbers


[from-fax-fxs-1] exten => _1NXXNXXXXXX,1,goto(from-internal,999${EXTEN},1) exten => _NXXNXXXXXX,1,goto(from-internal,999${EXTEN},1) exten => _NXXXXXX,1,goto(from-internal,999${EXTEN},1) include => from-internal
  1. assign the special extension in “context” field of the extension settings to use the new custom context “from-fax-fxs-1”

  2. add an outbound route with a dial pattern what will only match the strings dialed by this extension and remove the sentinel pattern before dialing out:



and only assign the desired trunk to this outbound route.

But having build-in support for this would be much nicer!

Tim Miller Dyck

Remote extension monitoring etc… would be fantastic.

Where asterisk + freepbx is most useful is for companies that have multiple branches (so a pbx at each branch) and quite a few inter office calls.

Unfortunately the hacks you have to work with to get remote extensions working nicely in queues etc… makes it usually easier to have a vpn between branches, and setup a second extension for any phone, that needs to be in a queue that way your not disabling phone voicemail, and blf works etc…etc…

Since I support many sites across the US I find it difficult to know which extensions is part of which Pickup Group.
If there was a way to print out a report I would be a happy man.
I can’t tell you how many times I assigned an extension to a Pickup Group that is already used.

Mike Z