@abcym15 wrote:

Hello,
I've been Googling and searching the forums here for the past hour or so and can't seem to find an answer. I was wondering if anyone has experience implementing Distinctive Ring via Alert Info on the Cisco 78xx series? I have a 7821 & 7841 and would like to have different rings for queue calls vs normal internal calls. I've tried various things in the alert-info box such as Chirp-2 and but these don't seem to do the trick. Ideally I would like to have Chirp 1 play for internal calls and Chirp 1 also play for queue calls, but with a double ring rather than a single.

Just wondering if anyone's managed to do this?

Kind Regards

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@neuronetv wrote:

I've installed freepbx-13 on centos 6 and I'm trying to set up fail2ban to block failed GUI logins but i can't find where freepbx logs these.

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@jrosetto wrote:

I am integrating asterisk and vTigerCRM.

I need this line in my trunk in extensions_additional.conf

exten => _X.,1,Agi(agi://0.0.0.0/incoming.agi)

If I add it manually and do 'core asterisk restart' everything works.

When I make a change in FreePBX the addition is removed and the CRM integration stops.

How do I add the extra dialplan to the trunk without FreePBX overriding it?

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@xaruqui wrote:

Hi there. I've been struggling with this but I'm in a dead end. I've read in this community about not ringing whilst external call reaches FreePBX, but in my case is just the opposite; there isn't any ringing while placing an external call from an extension. It worked with the former main trunk, but a few days ago I set up a new one, no ringing is happening since.

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@chacaman wrote:

hi, guys thanks for time and help.

Issue: Lost CDR Uniqueid
Freepbx: 10.13.66-13
Asterisk: 11.23.1

i have appr running a macro when the user hit *100 i call a macro when i call the macro i get the next:
-- Executing [s@macro-on:10] NoOp("SIP/1000-00000d72", "1476836678.4560") in new stack
but when the call finish i get in the cdr record a diferent uniqueid
CDR Uniqued: 1476836678.4559

how i can have the same uniqueid of the cdr record in the macro when this run?

Note: this happens only white the incoming call from PSTN or other PBX Trunks when i run this command from extension to extension i get the same cdr uniqueid.

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@zirophyz wrote:

Hello,

Been reading in the Polycom forums about this, and whoah what a mess it seems to be.

I found one post, that seems to have a solution including recompiling chan_cip.c which tbh, is a bit of a pain for ongoing support.
http://community.polycom.com/t5/VoIP/Server-based-DND-using-FAC-NOTIFY/td-p/538/page/2

Then, there are forum posts for Polycom mods that state something along the lines that this had been fixed up in UCS4.0.X, then other posts where it doesn't work. Then there are posts about server-side DND doco being under an NDA, as I feel it was something special they did for Broadsoft, and perhaps Broadsoft have a policy of NDA'ing things (i don't know!).

Kind of throwing my hands in the air on this one. I know these handsets are moderately popular, and wondering who has had luck with REAL server-side DND (Real = not a BLF key but using the built-in DND toggle).

Also willing to hear war stories about server-side CF as that config seems to be very similar.

The UCS 4.0.5 Admin Guide lists commands to enable server feature control, but nothing documented about where or how to set a feature code when server feature control is enabled. This part of the no documentation is where NDA's start being dropped and people go awfully silent. Support won't support anything that isn't officially documented, and then drop the NDA and refuse to talk.

What a mess ... so glad we're getting rid of these Polycoms shortly....

Edit: this is on a FPBX2.11 box.

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@lblokland wrote:

Hi,

after setting the LDAP settings in the User Manager, ths status says connected however import fails.
I want to start the search on the base dn.

In the console fwconsole userman sync --verbose I get:

[Whoops\Exception\ErrorException]
ldap_search(): Search: Operations error

When adding CN=Users in front of the base DN, I get only the users in that folder, looks like its not searching deeper in the tree.
please help

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@shuffman wrote:

Hello all and thanks for any help I can get. I'm new to Linux so please help with any commands needed to be ran. I'm running 10.13.66-11 distro and all updates are done and have even ran the yum update and still get the error below when trying to apply the updates. I've tried fwconsole chown as I have seen on some forums has fixed this and still have the problem. I've ran fwconsole debug and get the error at the bottom of this page. Thanks again for any help

Error:
exit: 1
Exception: Unable to find the Asterisk binary in file /var/lib/asterisk/bin/retrieve_conf on line 39
Stack trace:
1. Exception->() /var/lib/asterisk/bin/retrieve_conf:39

Version:
PBX Firmware: 10.13.66-11
PBX Service Pack: 1.0.0.0

fwconsole debug Log:
==> /var/log/asterisk/freepbx.log <==
[2016-Oct-19 09:13:21] [WARNING] (libraries/modulefunctions.legacy.php:7) - Depr eciated Function module_getinfo detected in /var/www/html/admin/modules/cxpanel/ functions.inc.php on line 58
[2016-Oct-19 09:13:21] [WARNING] (libraries/modulefunctions.legacy.php:7) - Depr eciated Function module_getinfo detected in /var/www/html/admin/modules/cxpanel/ functions.inc.php on line 67
[2016-Oct-19 09:13:21] [CRITICAL] (BMO/Notifications.class.php:507) - [NOTIFICAT ION]-[freepbx]-[RCONFFAIL] - retrieve_conf failed, config not applied (Reload fa iled because retrieve_conf encountered an error: 1)
[2016-Oct-19 09:13:46] [WARNING] (libraries/modulefunctions.legacy.php:7) - Depr eciated Function module_getinfo detected in /var/www/html/admin/modules/cxpanel/ functions.inc.php on line 58
[2016-Oct-19 09:13:46] [WARNING] (libraries/modulefunctions.legacy.php:7) - Depr eciated Function module_getinfo detected in /var/www/html/admin/modules/cxpanel/ functions.inc.php on line 67
[2016-Oct-19 09:13:47] [WARNING] (libraries/modulefunctions.legacy.php:7) - Depr eciated Function module_getinfo detected in /var/www/html/admin/libraries/featur ecodes.functions.php on line 42
[2016-Oct-19 09:17:13] [WARNING] (libraries/modulefunctions.legacy.php:7) - Depr eciated Function module_getinfo detected in /var/www/html/admin/modules/cxpanel/ functions.inc.php on line 58
[2016-Oct-19 09:17:13] [WARNING] (libraries/modulefunctions.legacy.php:7) - Depr eciated Function module_getinfo detected in /var/www/html/admin/modules/cxpanel/ functions.inc.php on line 67
[2016-Oct-19 09:17:43] [WARNING] (libraries/modulefunctions.legacy.php:7) - Depr eciated Function module_getinfo detected in /var/www/html/admin/modules/cxpanel/ functions.inc.php on line 58
[2016-Oct-19 09:17:43] [WARNING] (libraries/modulefunctions.legacy.php:7) - Depr eciated Function module_getinfo detected in /var/www/html/admin/modules/cxpanel/ functions.inc.php on line 67
OUT > [2016-Oct-19 09:28:02] [WARNING] (libraries/modulefunctions.legacy.php:7) - Depreciated Function module_getinfo detected in /var/www/html/admin/modules/cxpanel/functions.inc.php on line 58
OUT > [2016-Oct-19 09:28:02] [WARNING] (libraries/modulefunctions.legacy.php:7) - Depreciated Function module_getinfo detected in /var/www/html/admin/modules/cxpanel/functions.inc.php on line 67
WARNING - Depr eciated Function module_getinfo detected in /var/www/html/admin/modules/cxpanel/ functions.inc.php on line 58

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@assos40 wrote:

I have a strange issue.
I am using a vega 200 gateway(192.168.0.75) for my pri line ,and the FreePbx is on 192.168.0.200.
Until today everything was working fine.
Suddenly i can not ping from FreePbx the Vega and via versa.
I had to change the vega ip to 192.168.0.74 to receive again calls.
It seems that the only ip address the Freepbx cant ping is 192.168.0.75.
Everything else is fine.Just this specific ip address is not accessible.
Any ideas,please
Thank you
EDIT:
I made a firmware update on the vega 200.
In the documentation it states that i should do a factory reset after firmware update.
Will i loose my current configuration if i perform a factory reset ?
Thank you

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@mvogel4949 wrote:

So I setup a page group (3333) to match up to a multicast paging group and then programmed each Grandstream to listen on that same port. When I press the blf for 3333 there is no multicast paging audio but if I setup a multicast paging button on the phone with the port I programmed for them to listen on then it works great. Am I missing something in the setup?

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@tlarrea wrote:

FreePBX: 13.0.188.8
Asterisk: 13.11.2

Had an issue today where we could no longer make any calls to internal extensions. My handset appeared to be still registered but I tried to initiate a call to another extension and simply got silence, no ring or error. Tried to re-register handset and it failed.

When I looked through the logs, I found a gradual series of entries similar to the following, but for each extension.

[2016-10-20 14:40:54] VERBOSE[11138] res_pjsip/pjsip_configuration.c: Contact 238/sip:238@192.168.6.77:5060 has been deleted

Network functionality appeared OK, I was able to ping handset IP both from the PBX server console and my local workstation, PBX server was pingable from my workstation, as were all our other servers, and the FreePBX GUI was still functional.

Rebooting the server fixed the problem (not a graceful solution, but people were shouting at me to get it working).

Is there anything I can dig into to see what the cause of this problem was, or other things I can check in the event this problem happens again? I did look at the "/var/log/asterisk/full" log around the time that the problem started, there seems to be an 8 minute gap in the "/var/log/asterisk/full" file.

Not sure what else I should look at, love any points on how to run this issue down.

thanks,
Tim.

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@neuronetv wrote:

I have a new install of freepbx from the freepbx distro FreePBX-32bit-10.13.66.iso and fail2ban [pbx-gui] doesn't work
rpm -q fail2ban
fail2ban-0.8.14-1.shmz65.1.129.noarch
I have itpables and fail2ban running, the [sshd] jail works because I tested it and got banned but I can do failed gui logins until the cows come home and fail2ban just sits there doing nothing. Is this a bug that can be fixed?

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@JMickerts wrote:

Hi,

I need to be able to centrally enable and disable call forwardings for all extensions on my PBX. The reason for this is that the attendant can enable forwarding if employees are on holiday or sick. Is there a way to do this in FreePBX or is there a third-party module that accomplishes that?

Kind regards,

Jens

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@Wanderer wrote:

I tried installing RasPBX on a Raspberry Pi, it is running FreePBX 13.0.188.8 and Asterisk 13.11.2. If under Advanced Settings-GUI Behavior I set "Enable The Old Style FreePBX Dial Patterns Textarea" to Yes then what I find is that anything I enter in the textarea on an outbound route is not saved, and if there are already some patterns there because I've imported some from a CSV file, and I try to make any changes then all the changes and the existing patterns are lost, and the textarea is left empty after I click Submit. Has anyone else noticed this behavior?

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@sentinelace wrote:

We have noticed with all our PBX's that even the on hold default has terrible quality. We have tested this in the cloud and on premise. Even if you call Sangoma directly and are put on hold it has static. From what we have read, sln format has better quality and can clear this up. Is this supported? Is this a codec issue? This is only on the MOH side of things.

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@Oedi wrote:

I have just built a new PBX and I am having a problem dialing out from it. The phone number is getting lost along the way and i am not sure why. I have other systems that work no problem it is only with this new system the information is getting lost. I worked out the line where it is getting lost "dialout-trunk,2,6132185319,,off" from the working system and this is the non working system "dialout-trunk,2,,,off". I tried SIP and PJSIP and they have the same problem.

Anyone have any ideas on how to fix this? I have gone over the configuration and they are the same.

Broken System - Asterisk 13.11.2

[2016-10-20 14:42:42] VERBOSE[21466][C-0000000f] pbx.c: Executing [6132185319@from-internal:3] ExecIf("PJSIP/4100-00000027", "0 ?Set(CDR(accountcode)=)") in new stack
[2016-10-20 14:42:42] VERBOSE[21466][C-0000000f] pbx.c: Executing [6132185319@from-internal:4] Set("PJSIP/4100-00000027", "MOHCLASS=default") in new stack
[2016-10-20 14:42:42] VERBOSE[21466][C-0000000f] pbx.c: Executing [6132185319@from-internal:5] Set("PJSIP/4100-00000027", "_NODEST=") in new stack
[2016-10-20 14:42:42] VERBOSE[21466][C-0000000f] pbx.c: Executing [6132185319@from-internal:6] Macro("PJSIP/4100-00000027", "dialout-trunk,2,,,off") in new stack

Working System 13.7.2 (also works on 13.9.1)

[2016-10-20 15:18:40] VERBOSE[24036][C-000012a6] pbx.c: Executing [6132185319@from-internal:3] ExecIf("PJSIP/201-00001024", "0 ?Set(CDR(accountcode)=)") in new stack
[2016-10-20 15:18:40] VERBOSE[24036][C-000012a6] pbx.c: Executing [6132185319@from-internal:4] Set("PJSIP/201-00001024", "MOHCLASS=default") in new stack
[2016-10-20 15:18:40] VERBOSE[24036][C-000012a6] pbx.c: Executing [6132185319@from-internal:5] Set("PJSIP/201-00001024", "_NODEST=") in new stack
[2016-10-20 15:18:40] VERBOSE[24036][C-000012a6] pbx.c: Executing [6132185319@from-internal:6] Macro("PJSIP/201-00001024", "dialout-trunk,2,6132185319,,off") in new stack

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@YellowPaper wrote:

Hi all!

I have a quick question, maybe someone can help me out. Since our business consists of multiple phone numbers, I wanted to know if there was anyway of knowing which DID is being called in?

I know that you can route them to specific extensions to ring at certain phones, but if all the phones are configured to receive inbound calls, is there a way for us to know which DID is being dialed?

Not sure if there is a feature on the grandstream phone that will allow us to know what business number is being called, as there in the future, two businesses will be hosted at the same location. So we dont really want to have separate phones or extensions, but more to know which way to greet the phone.

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@neuronetv wrote:

I have a centos 6 machine and I installed freepbx-13 using the tarball. I did a backup(A) from within the gui, then tried to restore a different backup(B), it didn't work. So then I tried to restore backup(A) again and the whole freepbx system appears to have crashed. In the browser it's just a white page with black text that reads:

0 System Admin 13.0.67 Copyright 2016 by Schmoozecom, Inc., All rights reserved By installing, copying, downloading, distributing, inspecting or using the materials provided herewith, you agree to all of the terms of use as outlined in our End User Agreement which can be found and reviewed at www.schmoozecom.com/cmeula_

in the console I tried fwconsole restart, fwconsole ma install framework but all commands give:
0

System Admin 13.0.67
Copyright 2016 by Schmoozecom, Inc., All rights reserved

By installing, copying, downloading, distributing, inspecting or using
the materials provided herewith, you agree to all of the terms of use as
outlined in our End User Agreement which can be found and reviewed at
www.schmoozecom.com/cmeula

is this a case of format re-install everything? or can this system be rescued? Thanks for any advice.

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@PegasusBS wrote:

I am looking for a solution that will allow me to send faxes by sending an email with pdf attachment and then having the fax go out. I have found the Fax Pro will send faxes but it seems to only do so from the UCP and I need to be able to do it from an email to integrate with some website setups. Currently we use a program called Faxmaker but we are trying to see if we can get away from that and use the freepbx system.

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@Shandley wrote:

Greetings,

I have an old system that I am upgrading. I am installing, in a new box, the latest distro.
I would like to know what I have to convert the audio from the Freepbx 10 (FreePBX 2.10.1.6) to the FreePBX 13 distro installation in order for the voicemail greetings (and IVR welcome messages, etc) to be copied onto the new installation and work. I have locations and I have tried it directly but it does not seem to recognize that there are any audio files in there. I am assuming there needs to be a bit of conversion in the audio files.

If I am way off my mark, please let me know!

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