FreePBX 2.10 plans - what we are thinking


I have two small suggestions;

  • One I spent a bunch of time recently walking someone through DAHDI who had used ZAP at one time. We got to the Chanel Banked Centrex lines and setup the “ZAP Channel DIDS” module so that inbound the routing worked the way they wanted. Since this is mostly tags and context names, could we get it updated to Dahdi Channel DIDS or maybe some non technology specific label (Inward Dids)?? Spending lots of time explaining why we don’t have ZAP any longer and then telling someone to use ZAP Channel DIDs is a bit confusing.

  • On another note, a number of years ago Phillipe was nice enough to help me during a Huntsville class with a mod to force the outbound call to a Centrex Channel (Channel Bank slot) with a combination of a data entry in Asterisk and a modified context. This has been a big help because now the outbound PSTN call almost always goes out the trunk assigned as the Inward route so it acts like a private line in a copper world. A SIP /IAX2 solution would be better, but the customer is always right…



I don’t know how many organizations use Conferences like we do (4xPRIs + SIP Trunking). We’ve issued one or more conference “rooms” to our employees who can bring up their conferences on an ad hoc basis. Much like the voicemail having a portal, could we have a conferences portal?

The WebMeetMe is something close to what I’m thinking, but without the scheduling and AD integration. The core features would include:

[code] - Ability to change Admin PIN

  • Abiiity to change User PIN
  • Ability to change the conference options:
  • Leader Wait:
  • Quiet Mode:
  • User Count:
  • User join/leave:
  • Music on Hold:
  • Music on Hold Class:
  • Allow Menu:
  • Record Conference:
  • Maximum Participants:
  • Mute on Join:
  • While a conference is in session, the admin can:
    • see all callers by number and caller-id
    • mute individual channels in an active meetme
    • mute all but admin in an active meetme
    • control the speaking level of each channel
    • control the loudness of each channel
    • dial a number to be added to the conference without PIN
    • kick a caller
  • Bonus points would be given for an active GUI that can display the current “talker”
  • Multiple Admin’s may be using GUI on the same meetme simultaneously
  • Ensure it can support >70 simultaneous members in the meetme
    This probably has merits of being its own app or an app in the fbpx store. (I only wish I had time and php/ajax/html5 skills.)


I’ve been using the system for a couple of years now to host small business VOIP Service and want to thank you guys and say it’s great. I’ve managed to get custom parking lot group assignments working (including a GUI change to assign it to extensions), Missed Call e-mail notifications (custom script), CDR and report modifications to show DNID (dialed number), and recording integration with our web-based platform. I’ve also been able to restrict outbound calls to specific routes using available tools and do some other interesting things with the platform. Thanks again for all your hard work - it’s really great.

I’m currently on on Asterisk 1.8.4.

I was originally a fan of FOP, but as our systems have grown, we’ve turned it off. A replacement desktop tool for this or seperate service for it - runnable on a seperate server would be nice, but we’ll do iSymphony if we REALLY need it - so far, we’re ok without it.

I’m far from an expert, but have some experience and here are some things I’d like to see - forgive my ignorance in advance if there are other solutions for these that I should be looking at - I also realize that these are my personal and business desires and that there are a lot of other people out there with other needs that may take priority:

  1. Customers and/or Groups creation ability along with the ability to assign any object (extension, ring-group, inbound route, ivr, etc…) to that group with permissions to see or change it and REPORT on it. I’d like to be able to assign inbound routes to a client and run some call reports, for example. Reporting in general could definitely use a lift. The CDR reports, much like FOP seem to be pretty stagnate.

  2. Recording flexibility and new reporting capabilities - it would be nice for me to get away from the seperate web and sql based platform that I run for this. There’s a big question in my mind about whether recording changes (and possibly a new interface) would be part of CDR data and reports or would have options to configure and control it completely seperate from CDR’s. Ideal for me would be recordings including in a call log report which was tied to the customers/groups concept as described above - as part of an overall reporting overhaul - where CDR reports would become call-detail reports with play controls for recordings, but could be grouped and/or restricted by customer and/or group. Summary reports would be great too - by group, geography, area code, DNID, etc… Customizable call dispositions, notes, and other fields would be a nice add. here too for the call detail interface. Customers like to listen to them, select if they made a sale or not, write notes about the call, etc…

  3. Redundancy/Replication - Some sort of built-in support to replicate the configs/db/recordings, etc… and/or have a back-up/secondary server with a shared IP would be awesome. I know there are solutions out there for this, but I think nearly everyone would be more confident in having 2 servers look like 1.

  4. End of call post-back URL support might be good - we could send our call data to google analytics which may help with the reporting end of things…not sure how far that could be taken, but it’s worth a look.

  5. E-mailing recordings at call completion would be nice along with the above - similar to voicemail, but for all recorded calls if you turn it on.

  6. Missed call notification via e-mail on the interface would be nice as opposed to the custom scripting currently required.

  7. Simple screen-pop tool and/or Com object to write one. I don’t necessarily want a soft-phone - although a tool that provided call control of existing hard-phones would be a nice addition…not sure if we could control answer and dialing on a hard-phone, but hangup and hold probably could be pretty easy. Maybe answer and dialing could also be done via info/options msgs.

  8. Number pool management built into the system seperate from inbound routes. It would be nice to keep track of numbers that you may have which are not assigned to anything - or numbers that you decommission. Also nice if assigned numbers showed the client/group requested in item 1.

  9. Click-To-Call creation/management built into the interface would be cool. I’m doing click-to-call today with custom web-scripts and AMI, but it would be nice to be able to manage it in the interface.

I’m sure there’s more that I’m not thinking about, but hopefully the above spark a few ideas. Feel free to e-mail me or respond to the thread if you want more info.

(Tom Rymes) #24

While we don’t have much of a problem with managing WHEN or IF a call gets recorded, we do record EVERY incoming call to our main customer service queue.

Based on that fact, we generate a very large number of call recordings in /var/spool/asterisk/monitor, which take up surprisingly little space when recorded as wav49 format.

Based on this, I would like to see two things WRT recordings:

1.) A dedicated interface for finding and reviewing call recordings. Normally I know who took the call, and when, and I can find the call in the CDRs, but I cannot get to the recording without going into MySQL to look up the UniqueID, because ARI chokes on the large number of recordings.
2.) A tool that archives recordings by date. Without this, we end up with absurd numbers of recording files in the monitor directory, which makes finding what you want difficult.

I also like the idea of being able to track DID in the CDRs. Something as simple as knowing which numbers have been receiving the most traffic would be huge. One example is that many companies will use a new DID specifically for one marketing campaign, and track response by how many calls come in to that number. If there’s a way to do this now (besides the accountcode trick mentioned above), I am unaware of it.



Looking for CDR Report Module like trixbox to play the recorded file

(Tom Rymes) #26

I also have been looking for Trixbox-like call recording functionality for a long time now. Basically, it is just a link to the recorded file in a CDR report. IIRC, there were security and scalability/speed problems with the TB implementation, but the essence of the request is that the process for finding the call recordings should not require more work than just finding the call in the CDRs. Searching via incoming trunk, callerID, date/time, called party, etc would be ideal.


Can anyone tell me where the Run after record fields went in 2.9?



I’m in the middle of rebuilding my server at work for two reasons: 1) It needs an upgrade and a cleaning and some cleaner routing, and 2) Almost everyone in my 40-person office is complaining about FOP being broken. Like the above posters, my people use it to see who’s on a call and for how long.

I must at least ask to not get rid of it, though I understand if it must go for the greater good.

I would have no problem with FOP2 if it was just a little easier to manipulate during installation (especially if one is upgrading from the original). I have purchased a license for it and never did get it to work correctly- though that’s not the programmer’s fault! I’ve seen the product in operation; Asternic can be proud of themselves.


Look at the new CDR Reports module, it will show the recordings and you can play them.


Already reinstalled a few times.
And I’m not getting anywhere with installing mISDN.
The results are:

yum install asterisk18-misdn
(Since that is supposedly the correct incantation)
And I get:

—> Package asterisk18-misdn.i386 0: set to be updated
–> Processing Dependency: mISDN-kmod for package: asterisk18-misdn
–> Processing Dependency: mISDN for package: asterisk18-misdn
–> Processing Dependency: mISDNuser for package: asterisk18-misdn
–> Running transaction check
—> Package asterisk18-misdn.i386 0: set to be updated
–> Processing Dependency: mISDN-kmod for package: asterisk18-misdn
—> Package mISDN.i686 0: set to be updated
–> Processing Dependency: kmod-mISDN for package: mISDN
—> Package mISDNuser.i386 0: set to be updated
–> Finished Dependency Resolution
mISDN- from pbx has depsolving problems
–> Missing Dependency: kmod-mISDN is needed by package mISDN- (pbx)
asterisk18-misdn- from pbx has depsolving problems
–> Missing Dependency: mISDN-kmod is needed by package asterisk18-misdn- (pbx)
Error: Missing Dependency: mISDN-kmod is needed by package asterisk18-misdn- (pbx)
Error: Missing Dependency: kmod-mISDN is needed by package mISDN- (pbx)

Which I think stems from the fact that dependancies ask for:

  • mISDN-kmod
  • kmod-mISDN

But I’m not a yum guru, so I’m relucant to run anything like yum update, since that created an ever greater mess ending in me losing dadhi stuff.


Greetings …

I think this is the wrong place to be asking this question, but I will try and help away.

You need to install the kmod-mISDN for the kernel you running. I have had this problem before, so I have manually checked and installed the correct kmod that is available for my kernel. Some times, kernel come out before the kmod is updated, so you need to check if there is a kernel and kmod.

I think you should be able to do …

yum install mISDN-kmod-base asterisk-misdn

and that should pull in all your needed dependencies.

Mine looked like this …
[leet@pbx ~]$ rpm -qa |grep -i misdn | sort

And I have both mISDN and DAHDI running, so it should not be a problem.

You might also find more details at

Best of luck



I totally agree.
But I sort of forgot where I was, being happy to find anything about ISDN here…
It seems a sort of subject with very little concise attention.
I’d be happy to remove my submission and put it somewhere in the forum for history…

Perhaps because ISDN did not make it to what is was set to be, and it is not much used in the US?


Not a problem.

Mmm, I use mISDN because we have access to in-expensive USB ISDN TA devices. Which for up to 5 ISDN devices works well.

I hope you come right, if you need more help, let me know.

FreePBX mISDN module is not working so well at the moment and I don’t know enough php to fix the current problem, you will have to manually configure your mISDN install once the rpms are in place.


(system) closed #34