@depster wrote:

The following appers when attempting to load the /recordings login since upgrading to 13.0.1 RC1.4.

Posts: 2

Participants: 2

Read full topic

@shillamus wrote:

FreePBX 12.0.76 Aterisk Asterisk 11.19.0

I am unable to get voicemails (.wav files) to forward via email. The most instruction I have at this point is to set the email address for each extension and to enable email attachement

I am not clear on how FreePBX or Asterisk knows how to send an email. What email address would the message come from?

I see this matter has been discussed before http://community.freepbx.org/t/email-voicemail-messages/15800. It appears there is no easy "out of the box" solution.

Many thanks for any help, tips,hints or clues you can share.

Regards,
Scott

Posts: 4

Participants: 2

Read full topic

@hlusher wrote:

Unable to find asterisk results after upgrade to FreePBX 13.

Posts: 5

Participants: 5

Read full topic

@mohammed_kakuji wrote:

Hi,

I am currently using FOP2 which is a flahs operating panel. It is showing me that conferences are remaining active with no extension dialled in.

Are the users not doing something correctly?

Posts: 1

Participants: 1

Read full topic

@charlesc83 wrote:

I'm testing out FreePBX and I have setup a site-to-site with one of my remote locations, VPN traffic is good on both sides, I can ping the phone from HQ and remote site can ping the FreePBX. I'm using a Yealink T20P, and I have set the remote local subnet in the SIP settings and allowed the remote local subnet to that extension. However while the phone will not register, I did install X-Lite on a workstation and it registered fine. Ami I just missing something?

Posts: 1

Participants: 1

Read full topic

@xrobau wrote:

For those coming in late, read this thread:

(If you don't want to read it, basically, I'm writing a new FOSS firewall module for FreePBX, that does most of the thinking for you, and it's almost done!)

As it was getting a bit long, I thought I'd start a new thread, and update everyone on what's been happening!

It's been moving up and down my priority list - as you can see from the commits - but it's almost at a point where I'm looking for people to actually test it, for real!

The requirements are:
- You need to be running FreePBX 13
- You need to be running a CentOS 6 based Distro (I'll post more about C7 and firewalld shortly)
- You need to have the Sysadmin RPM package installed.

The Sysadmin RPM package is used to enable limited privilege escalation. I'll probably redo it so it uses something slightly more portable (such as sudo) as I get it working on Debian/Ubuntu and other distros (or someone else can, this is all open source, go wild! Pull requests welcome!)

But, for the moment, this means that if you don't want to put a bunch of extra work in, a recent FreePBX Distro machine will be the easiest thing for you to test on.

If you're feeling enthusiastic about testing, please either ping me on IRC (join the #freepbx channel on freenode, and type 'X-Rob' or 'xrobau', and that'll attract my attention), or, send me a PM here, or just reply to this thread.

I'm also interested in anyone who has any scripts or things like that to actually ATTACK a SIP server, as I want to do some active testing too. I'll also be exposing a couple of unfiltered machines to the internet (and publishing their details here) for random attacks, if anyone wants to have a go.

The downside is, I'm off at a Vintage Moto Trials event this weekend, so whilst I will have internet access, I won't have much of a development environment if everything breaks. You may end up with nothing to play with this weekend!

So. Who's interested?!

Posts: 3

Participants: 1

Read full topic

@Starlifter wrote:

Hello!

I was testing the chan_pjsip driver this weekend, because I like the idea of sharing one extension on multiple endpoints.

When I was using chan_sip, I had created some BLFs for other extensions on my Yealink T38G. They worked fine. After I had changed to chan_pjsip, the BLFs stayed dark. After an hour of anger, I figured out, that BLFs start working again, when you set a presence state in the UCP. Then all ist fine: The BLFs light green, when the extension is not in use, blink, when the extension is ringing and red, when the extension is in use.

So, I see the benefits of coupling the "device"-presence to a "user"-presence. A user can log in to his UCP and set, let's say, DND and this is reflected to all BLFs that point to his account.

But: At first, I don't need that. I just want to see, if someone is on the phone or of the extension is ringing. Like it did with chan_sip.
Then, there is a problem: A user with multiple extensions would have to set his presence state in every single UCP, as it is not allowed to link one user to multiple phones.
Also, as administrator, you have to login to every UCP and set "available" as presence state. When registering multiple extensions, this might be quite a work.

Do you encounter the same behavior? Is it a bug, or is it a feature?

Oh, nearly forgotten: I am using a FreePBX Distro, 10.13.66-5

Kind regards!
Andy

Posts: 2

Participants: 2

Read full topic

@w1zz wrote:

Installed ffmpeg following the pointers from Andrew.

Having done so playback fails with the following:

Details of the installed version are:

ffmpeg version 0.10.15
built on Aug 30 2014 15:49:19 with gcc 4.4.7 20120313 (Red Hat 4.4.7-3)
configuration: --prefix=/usr --bindir=/usr/bin --datadir=/usr/share/ffmpeg --incdir=/usr/include/ffmpeg --libdir=/usr/lib64 --mandir=/usr/share/man --arch=x86_64 --extra-cflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m64 -mtune=generic' --enable-bzlib --disable-crystalhd --enable-gnutls --enable-libass --enable-libcdio --enable-libcelt --enable-libdc1394 --disable-indev=jack --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-openal --enable-libopenjpeg --enable-libpulse --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libv4l2 --enable-libx264 --enable-libxvid --enable-x11grab --enable-avfilter --enable-postproc --enable-pthreads --disable-static --enable-shared --enable-gpl --disable-debug --disable-stripping --shlibdir=/usr/lib64 --enable-runtime-cpudetect
libavutil 51. 35.100 / 51. 35.100
libavcodec 53. 61.100 / 53. 61.100
libavformat 53. 32.100 / 53. 32.100
libavdevice 53. 4.100 / 53. 4.100
libavfilter 2. 61.100 / 2. 61.100
libswscale 2. 1.100 / 2. 1.100
libswresample 0. 6.100 / 0. 6.100
libpostproc 52. 0.100 / 52. 0.100

Alan

Posts: 2

Participants: 2

Read full topic

@mlsoftware wrote:

Hi to all and sorry for my english,

in sample, our environment:

5 "public" number: 123 456 301, 123 456 302, 123 456 303, etc
5 extension: 301, 302, etc (last 3 digit of public number)
5 inbound route: first, route incoming call from 123 456 301 to extension 301, etc

301 is operator, all device are SNOM and freepbx is v12

Our goals:
1. If someone call (for instances) 123 456 305 and there is no answer by extension 305, call must be forwarded to 301 after 30 seconds.
2. If someone call 123 456 301 and operator 301 blind transfer call to 305, after 30 sec without answer, call must go back to it
3. All internal call to 305 must not be forwarded to 301

We try to find solution with queue.

Inbound to 123 456 305 are destinated to queue 405 with only 305 as member and 301 as destination for no answer.
"No answer" option for extension 301 is set to terminate call and ring until caller hangup.

Everything works fine with one only exception in point 2. When 301 blind trasfer external call to 305, obviously with this configuration, call not return after 30 second

Where is our mistake? Is there another way to accomplish our 3 goals?

I hope to be clear in my explanation.

Thanks in advanced for any help.

Kind regards.

Massimo

Posts: 1

Participants: 1

Read full topic

@EriclKlein wrote:

When trying to listen to recording via the UCP I now get an error that Chrome does not support QuickTime player with a link (https://support.google.com/chrome/answer/6213033) which basically says:

To make browsing with Chrome safer, faster, and more stable, we stopped allowing NPAPI plugins on September 1, 2015.
Plugins that use NPAPI, including Silverlight, Java, and Unity, won’t work. If you want to use a website that uses an NPAPI plugin, you’ll need to use a different web browser.

Any suggestions beyond abandoning Chrome?

Posts: 3

Participants: 3

Read full topic

@bspann6 wrote:

Hello!

I am as new as they come to TrixBox/FreePBX/Asteriks and to say this is somewhat confusing is an understatement LOL. From what I am able to gather the company I am working for is using a somewhat older version of these items TrixBox version 2.6 and FreePBX/Asteriks 2.9.

What I have been tasked to do I am not even sure can be done at this point. They currently have everything up and running and all is well. However they have a small group of folks that are "on-call" and they have an IVR setup called Pager, within this, there is a drop-down called announcements that every morning depending on whom is on-call they select one of these options and update the system.

Now, what they are wanting is to automatically update this dropdown depending on a calendar...apparently this has been tried before(im not even sure how) however it did not work.

Now my question, is where is this dropdown information stored? is it within the a conf file somewhere, MySQL, or other? a lot of the code I have been reviewing talks about updating the dialplan, but I am not sure I need to do that..err maybe I do LOL,

I would like to thank everyone that has taken the time to read this, and perhaps might be able to help shed a little light on the subject.

Thank-You!

Posts: 1

Participants: 1

Read full topic

@nsimpson wrote:

Has any thought been put into expanding the search to allow for child containers? For example, some Active Directory models use OU to organize users so using a singular base dn fails. There might be a generic "Users" container with two child OUs of "Power Users" and "Normal Users". Specifying the Users container as the base DSN will leave out anyone in Power or Normal users.

A means to allow this would be to add a checkbox to the settings page with something along the lines of "Search child containers" which would cause the internal query to include child objects. For PHP, its a matter of adding ldap_set_option($connect, LDAP_OPT_REFERRALS, 0); after the connect and before the bind.

Posts: 2

Participants: 2

Read full topic

@nsimpson wrote:

I'm loving the AD authentication. It is great! I'm having an issue where I remove a user from AD, but then have no way to remove the user from freepbx/asterisk. Is there a way to remove old users without having to edit the mysql db directly?

Posts: 2

Participants: 2

Read full topic

@gene778 wrote:

Asterisk 13 fresh install using git. Error Message: "Could not determine Asterisk version (got: Asterisk GIT-master-857923d)" Please advise.

Posts: 3

Participants: 3

Read full topic

@transmatrix wrote:

I installed FreePBX 13 and Asterisk on Ubuntu Server 14.04.2 LTS following the instructions on the freepbx wiki (I'd post a link, but this forum won't let me)

Everything seemed to be working okay until I rebooted my server. I noticed that Asterisk wasn't running. I first started it manually via /etc/intit.d/asterisk start, but then the problem was that FreePBX couldn't access /var/run/asterisk/asterisk.ctl So, I tried running fwconsole start and I got the error [Exception] Sysadmin RPM not up to date

I tried to find out why I was getting this error or how to update Sysadmin, but apparently this shouldn't be happening on Ubuntu?

Posts: 6

Participants: 2

Read full topic

@el_es wrote:

Hi,
this is 'funny' and possibly very low on anyones priority list... but.

I have disabled RSS feeds that used to show on dashboard, cleared them off in the Advanced Config (I don't mind looking at the feeds as admin, but I'd rather have my users of UCP have different feeds on, like from our ticket system maybe; whereas as admin I don't need /that/ in the dashboard, I'd rather keep it with 'official' freepbx feeds; I /do/ understand it's way too far in the cycle to introduce this kind of change; I'm fine with that - but then I have to clear the RSS feed settings and that's it.).

Anyway, the above is not the point I'm after....

The point is now, I have now 3 widgets on dashboard remaining.
No matter what I do I can NOT move them about to be like:

[FreePBX Statistics] [Uptime]
[System Overview]
(so, system overview UNDER the statistics and justified to the left, and uptime on the right)

System overview ALWAYS moves itself from under Statistics to land under Uptime.
With RSS feeds widget on, the grouping I'm after was totally possible.

FreePBX 12.0.76.2,
MBoFirefox 41, on 1280x1024 19" monitor under Debian Wheezy+Backports (up-to-date)

The behavior persists and gets even 'funnier' when window is maximized (either system overview or the statistics widget is moved, they just don't want to stay in the same column of the screen, they always rearrange each other. Hell man, who's the admin here, I want them my way )

Posts: 3

Participants: 2

Read full topic

@doozer wrote:

Hi All,

This one has me confused.
We have 3 separate systems running Freepbx v12.0.76.2.
2 of these systems have "Video Support" enabled in SIP settings with h264 set as the only codec.
One has video disabled.

On both of the video enabled systems, placing a call via the outbound trunk to a regular POTS number, connects and voice works (no video obviously as its a POTS number), but after 3 minutes the call drops.
Every time, the logs indicate:

[2015-09-25 19:50:05] NOTICE[2798] chan_sip.c: Disconnecting call 'SIP/Engin-00000001' for lack of RTP activity in 31 seconds

This happens without fail every single time anywhere from 3 minutes, to 3:30.

On the system with video disabled, the call doesn't drop.

Disabling video on both other systems fixes the problem.

Any thoughts on how to fix this?
We need video enabled as both systems have an intercom at the front door with video.
We don't need video on outbound calls (ever).

Perhaps there's some way to disable video on all outbound calls?

Thanks,
Matt.

Posts: 1

Participants: 1

Read full topic

@kwriley87 wrote:

Is there a way to create a login for a user to log in to the PBX and listen to call recordings, but limiting what extensions they are allowed to listen to recordings from? I'm using FreePBX Distro 12.0.76.2

Posts: 5

Participants: 2

Read full topic

@normanghenderson wrote:

After minor module upgrades I got a tampered file warning for amportal. I tried:
amportal chown; amportal a ma refreshsignatures (which redownloaded framework saying sig was invalid); amportal a reload. The warning persisted no matter what I tried.This was later confirmed by Andrew Nagy as a module publishing issue that has been resolved. However...

In the meantime I downloaded and tried to install 13.0-latest from command line according to: http://wiki.freepbx.org/display/FOP/Installing+FreePBX+13+on+Ubuntu+Server+14.04.2+LTS
This seemingly went along fine until: ./install -n
at which point I got:
Checking if Asterisk is running and we can talk to it as the 'asterisk' user...Done!
Preliminary checks done. Starting FreePBX Installation
Checking if this is a new install...No (/etc/amportal.conf file detected)
PHP Fatal error: Call to a member function getAll() on a non-object in /var/www/admin/libraries/BMO/Freepbx_conf.class.php on line 216

Fortunately the GUI still runs. It shows version 12.0.76 and reports all modules up to date (stable stream). But now there are 7 additional "tampered files" all related to MoH - which has been disabled "pending upgrade to 13.0.6". Framework still shows as tampered.

I had hoped to get past the whole mess by installing 13.0 but as above that didn't work out so well. Now I'm stuck - unable to complete the command line install of 13.0, and the GUI doesn't want to upgrade anything either.

How do I get back to a a normal state either with 13.0.6 or 12.0.76? Naturally this is a live system - although not critical, it would be a huge paid to lose the current configuration. Any help appreciated!

Norm Henderson
Clinica CEML
(charity hospital in Angola)

Posts: 1

Participants: 1

Read full topic

@ntadmin wrote:

When the time is right, I have started to construct a proposal for how the design for modules could be tweaked to be both a bit easier to write and quicker to execute.

The heart of it is recognising that modules run in two, distinct, modes:

1. When executed during a call.
2. When accessed from the web interface for configuration.

The separation of these two functionalities allows the former to be executed with a very lightweight start-up (which I note that some of the inbuilt AGI scripts like dialparties already do), without sacrificing much. I have no desire to get in the way of the progress of 13.0, so I'm just flagging that, at the right time, I would love to have this discussion.

Posts: 2

Participants: 2

Read full topic