@andrexp wrote:

Hi,
Based on this thread (The thread is already closed) : https://community.freepbx.org/t/moh-per-extension/19845
I've tried the thread and if only one extension works fine,
But when I try more than one extension does not work,
Here's the contents of my sip_custom_post.conf file (i have 3 extension for example, real i have 20 extension)

100
mohsuggest=greenday
101
mohsuggest=gunsnroses
103
mohsuggest=lethergo

If more than one extension is like that?

Regards,
Andre

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@claloano wrote:

Sometimes I can make switchboards with 30 or 40 interiors and the band is a bit in some areas ...

Obviously I work on codecs and where I need more adsl together

But it's important to control traffic, today I use iptraf and I find it easy

But in other areas I have tried bwm-ng only that it seems that freepbx repositories do not expect it ...

Is there a way to solve it?

Or are there better or alternative assets to monitor freepbx traffic?

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@billsimon wrote:

I'm getting 500 Internal Server Error when I try to pull modules for FreePBX 14. Both mirrors give this error.

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@tkldr wrote:

Hello,

I've been working on getting IAX Trunks working for a while now and am having some issues. My goal is to bring up IAX and DUNDi trunks so I can inform each PBX about the others exensions. If there is a better way, please let me know.

The sites are connected via VPN.

site1 contains 2 Sangoma 300 Appliances in HA mode, so there is a floating IP. Sites 2 and 3 each have 1 Sangoma 100 Appliance.

Configuration is something like this:

site1 floating IP. 10.1.1.3
site1 FreePBX-ha01 IP 10.1.1.1
site1 FreePBX-ha02 IP 10.1.1.2

site2 FreePBX1 IP 10.2.1.1

site3 FreePBX1. IP 10.3.1.1

in the log in site two, I see this over and over:

2017-09-08 22:19:20] NOTICE[2271] dnsmgr.c: dnssrv: host '01.voice.com' changed from 10.1.1.3:4569 to 10.1.1.1:4569
[2017-09-08 22:19:20] NOTICE[2271] dnsmgr.c: dnssrv: host '01.voice.com' changed from 10.1.1.1:4569 to 10.1.1.3:4569

In DNS, "01.voice.com" is a single A Record pointing 10 the floating IP: 10.1.1.3.

In a Packet Capture I am seeing this:

-9:-5:-41.549905 IP 10.1.1.3.4569 > 10.2.1.1.4569: UDP, length 14
-9:-5:-41.550537 IP 10.2.1.1.4569 > 10.1.1.1.4569: UDP, length 12
-9:-4:-58.335394 IP 10.2.1.1.4569 > 10.1.1.3.4569: UDP, length 14

It makes me wonder if there is something special I am missing with the HA setup. Should I be pointing the trunk at the 'real' IP? If so, does that mean I need multiple trunks for each HA participant?

Thank you!!!

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@Pentium5 wrote:

Hi all,

My SIP provider requires using DNS SRV lookup as it balances the load between several SIP servers.
The option to enable or disable SRV lookup is available In an "old" chan_sip driver. However, I did not manage find it in a "new" chan_pjsip driver.
Is SRV lookup supported by chan_pjsip? If "yes", how can I enable or disable it?

Thanks in advance.

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@msheshi wrote:

Hello,
I am new to FreePBX and don't have experience.
I am trying to setup the outbound route for my FreePBX server.
My VOIP provider assigned to me a phone number ( 045351xxx )
I have configured my trunk but the problem is my dial pattern

This is what we have :
0 4 xxx xxxx (from within the Tirana administrative unit)
0 4 xxx xxxx (from within Albania, but outside Tirana)
+355 4 xxx xxxx (from outside Albania)
I want to know how to configure my dial pattern correctly in my outbound route

Thanks
Mikeli

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@Janthro wrote:

After performing an upgrade, I get errors.

PHP Fatal error:  Class 'PDO' not found in /var/www/html/admin/libraries/BMO/Database.class.php on line 17
Whoops\Exception\ErrorException: Class 'PDO' not found in file /var/www/html/admin/libraries/BMO/Database.class.php on line 17
Stack trace:
  1. () /var/www/html/admin/libraries/BMO/Database.class.php:17

This is the main error and the one causing the most trouble. It seems that the upgrade process says the server does not have internet connection, but I was able to ping out, ping via dns, perform wget, and everything else that I could think of.

Looking through the log from /var/log/post_sngupdate is where I saw the line saying unable to detect an internet connection.

I am a big time newbie when it comes to FreePBX and honestly quite the beginner with Linux as well, so I am not sure what to do.

When I do the route -n I see my ipaddress and my default gateway and everything looks good.

Any help would be appreciated.

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@MacsOffice wrote:

FreePBX 14.0.14
Asterisk 13.16.0
Sangoma S405
Can the Ringer be turned off completely and permanently in the template, using only the flashing LED to indicate incoming calls?

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@alexandre17220 wrote:

Hi,
I have installed FreePBX 13.0.192.16 on a new server.
I have configured extension and ring group.
ringgroup 200 (ring time 5 sec) -> extension 100 and 101 on destination if no answer ring groups 200.
No voicemail on extension.
When I call this ring group, the loop is ok, but after 30seconds I have in my log :

[2017-09-12 16:04:18] VERBOSE[10948][C-000000a2] app_dial.c: Nobody picked up in 5000 ms
[2017-09-12 16:04:18] VERBOSE[10948][C-000000a2] pbx.c: Executing [s@macro-dial:23] Set("PJSIP/156-0000033c", "DIALSTATUS=NOANSWER") in new stack
[2017-09-12 16:04:18] VERBOSE[10948][C-000000a2] pbx.c: Executing [s@macro-dial:24] GosubIf("PJSIP/156-0000033c", "0?NOANSWER,1()") in new stack
[2017-09-12 16:04:18] VERBOSE[10948][C-000000a2] pbx.c: Executing [s@macro-dial:25] NoOp("PJSIP/156-0000033c", "Returning since nobody answered") in new stack
[2017-09-12 16:04:18] VERBOSE[10948][C-000000a2] pbx.c: Executing [s@macro-dial:26] MacroExit("PJSIP/156-0000033c", "") in new stack
[2017-09-12 16:04:18] VERBOSE[10948][C-000000a2] pbx.c: Executing [220@ext-group:15] Gosub("PJSIP/156-0000033c", "sub-record-cancel,s,1()") in new stack
[2017-09-12 16:04:18] VERBOSE[10948][C-000000a2] pbx.c: Executing [s@sub-record-cancel:1] Return("PJSIP/156-0000033c", "") in new stack
[2017-09-12 16:04:18] VERBOSE[10948][C-000000a2] pbx.c: Executing [220@ext-group:16] Set("PJSIP/156-0000033c", "RingGroupMethod=") in new stack
[2017-09-12 16:04:18] VERBOSE[10948][C-000000a2] pbx.c: Executing [220@ext-group:17] GotoIf("PJSIP/156-0000033c", "0?nodest") in new stack
[2017-09-12 16:04:18] VERBOSE[10948][C-000000a2] pbx.c: Executing [220@ext-group:18] Set("PJSIP/156-0000033c", "__NODEST=") in new stack
[2017-09-12 16:04:18] VERBOSE[10948][C-000000a2] pbx.c: Executing [220@ext-group:19] Macro("PJSIP/156-0000033c", "blkvm-clr,") in new stack
[2017-09-12 16:04:18] VERBOSE[10948][C-000000a2] pbx.c: Executing [s@macro-blkvm-clr:1] Set("PJSIP/156-0000033c", "SHARED(BLKVM,PJSIP/156-0000033c)=") in new stack
[2017-09-12 16:04:18] VERBOSE[10948][C-000000a2] pbx.c: Executing [s@macro-blkvm-clr:2] Set("PJSIP/156-0000033c", "GOSUB_RETVAL=") in new stack
[2017-09-12 16:04:18] VERBOSE[10948][C-000000a2] pbx.c: Executing [s@macro-blkvm-clr:3] MacroExit("PJSIP/156-0000033c", "") in new stack
[2017-09-12 16:04:18] VERBOSE[10948][C-000000a2] pbx.c: Executing [220@ext-group:20] Goto("PJSIP/156-0000033c", "ext-group,220,1") in new stack
[2017-09-12 16:04:18] VERBOSE[10948][C-000000a2] pbx_builtins.c: Goto (ext-group,220,1)
[2017-09-12 16:04:18] VERBOSE[10948][C-000000a2] pbx.c: Executing [220@ext-group:1] GotoIf("PJSIP/156-0000033c", "0?cid") in new stack
[2017-09-12 16:04:18] VERBOSE[10948][C-000000a2] pbx.c: Executing [220@ext-group:2] PlayTones("PJSIP/156-0000033c", "ring") in new stack
[2017-09-12 16:04:18] VERBOSE[10948][C-000000a2] pbx.c: Executing [220@ext-group:3] Progress("PJSIP/156-0000033c", "") in new stack
[2017-09-12 16:04:18] VERBOSE[10948][C-000000a2] pbx.c: Executing [220@ext-group:4] Macro("PJSIP/156-0000033c", "user-callerid,") in new stack
[2017-09-12 16:04:18] VERBOSE[10948][C-000000a2] pbx.c: Executing [s@macro-user-callerid:1] Set("PJSIP/156-0000033c", "TOUCH_MONITOR=1505225027.1048") in new stack
[2017-09-12 16:04:18] VERBOSE[10948][C-000000a2] pbx.c: Executing [s@macro-user-callerid:2] Set("PJSIP/156-0000033c", "AMPUSER=156") in new stack
[2017-09-12 16:04:18] VERBOSE[10948][C-000000a2] pbx.c: Executing [s@macro-user-callerid:3] GotoIf("PJSIP/156-0000033c", "26?report") in new stack
[2017-09-12 16:04:18] VERBOSE[10948][C-000000a2] pbx_builtins.c: Goto (macro-user-callerid,s,15)
[2017-09-12 16:04:18] VERBOSE[10948][C-000000a2] pbx.c: Executing [s@macro-user-callerid:15] GotoIf("PJSIP/156-0000033c", "0?continue") in new stack
[2017-09-12 16:04:18] VERBOSE[10948][C-000000a2] pbx.c: Executing [s@macro-user-callerid:16] ExecIf("PJSIP/156-0000033c", "1?Set(__CALLEE_ACCOUNCODE=)") in new stack
[2017-09-12 16:04:18] VERBOSE[10948][C-000000a2] pbx.c: Executing [s@macro-user-callerid:17] Set("PJSIP/156-0000033c", "__TTL=0") in new stack
[2017-09-12 16:04:18] VERBOSE[10948][C-000000a2] pbx.c: Executing [s@macro-user-callerid:18] GotoIf("PJSIP/156-0000033c", "0?continue") in new stack
[2017-09-12 16:04:18] VERBOSE[10948][C-000000a2] pbx.c: Executing [s@macro-user-callerid:19] Wait("PJSIP/156-0000033c", "") in new stack
[2017-09-12 16:04:18] VERBOSE[10948][C-000000a2] pbx.c: Executing [s@macro-user-callerid:20] Answer("PJSIP/156-0000033c", "") in new stack
[2017-09-12 16:04:18] VERBOSE[10948][C-000000a2] pbx.c: Executing [s@macro-user-callerid:21] Wait("PJSIP/156-0000033c", "1") in new stack
[2017-09-12 16:04:19] VERBOSE[10948][C-000000a2] pbx.c: Executing [s@macro-user-callerid:22] Gosub("PJSIP/156-0000033c", "macro-user-callerid,lang-playback,1(hook_0)") in new stack
[2017-09-12 16:04:19] VERBOSE[10948][C-000000a2] pbx.c: Executing [lang-playback@macro-user-callerid:1] GosubIf("PJSIP/156-0000033c", "0?macro-user-callerid,fr,hook_0():macro-user-callerid,en,hook_0()") in new stack
[2017-09-12 16:04:19] VERBOSE[10948][C-000000a2] pbx.c: Executing [en@macro-user-callerid:1] Playback("PJSIP/156-0000033c", "**im-sorry&an-error-has-occurred&with&call-forwarding**") in new stack
[2017-09-12 16:04:19] VERBOSE[10948][C-000000a2] file.c: <PJSIP/156-0000033c> Playing 'im-sorry.ulaw' (language 'fr')
[2017-09-12 16:04:20] VERBOSE[10948][C-000000a2] file.c: <PJSIP/156-0000033c> Playing 'an-error-has-occurred.ulaw' (language 'fr')
[2017-09-12 16:04:22] VERBOSE[10948][C-000000a2] file.c: <PJSIP/156-0000033c> Playing 'with.ulaw' (language 'fr')
[2017-09-12 16:04:23] VERBOSE[10948][C-000000a2] file.c: <PJSIP/156-0000033c> Playing 'call-forwarding.ulaw' (language 'fr')
[2017-09-12 16:04:25] VERBOSE[10948][C-000000a2] pbx.c: Executing [en@macro-user-callerid:2] Return("PJSIP/156-0000033c", "") in new stack
[2017-09-12 16:04:25] VERBOSE[10948][C-000000a2] pbx.c: Executing [lang-playback@macro-user-callerid:2] Return("PJSIP/156-0000033c", "") in new stack
[2017-09-12 16:04:25] VERBOSE[10948][C-000000a2] pbx.c: Executing [s@macro-user-callerid:23] Macro("PJSIP/156-0000033c", "hangupcall,") in new stack
[2017-09-12 16:04:25] VERBOSE[10948][C-000000a2] pbx.c: Executing [s@macro-hangupcall:1] GotoIf("PJSIP/156-0000033c", "1?theend") in new stack
[2017-09-12 16:04:25] VERBOSE[10948][C-000000a2] pbx_builtins.c: Goto (macro-hangupcall,s,3)

Thanks a lot,

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@PitzKey wrote:

Hello all,

We just completed a fresh installation, we ran module updates, it did failed to update specific modules due to depending on other modules, now when we ran it again, we have the attached error.

cant update.png967x349 28.9 KB

We tried rebooting, we did system updates, but still getting same error when checking for online module updates.

Any help appreciated

Thanks

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@nsunseri0324 wrote:

Hello Everyone,

First time posting here, and I could not find an old post about this, maybe didn't search right. I have a DID that goes to a custom extension that rings the users cell phone. When call come in they go to the cell phone fine but always as "unknown caller". I have tried populating all of the CID fields with the DID number but Caller ID on the cell always shows unknown caller. Does anybody have any suggestions or a previous thread that I can research? Any help is greatly appreciated, thank you.

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@cyberdocwi wrote:

Hello,

A few weeks ago, I upgraded a customer from FreePBX 12 -> FreePBX 14, and I had a couple minor things to repair. Nice job Sangoma on the scripts!

I received a call today from the customer that their parking lot is broken. I tried it myself, and was unable to transfer a call to Parking Lot 70, the default. Tried both the BLF button and a manual transfer. Nope.

Looking at other posts on the web in various places, people were able to resolve by downgrading from Asterisk 13 to Asterisk 11. That worked for me too. I was running Asterisk 13.17.0 and using the standard parking lot FreePBX module 13.0.19.6

I will fire up my test box and see if Asterisk 14 works or not.

Christian

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@madav wrote:

Hi,

I am running into this error after an upgrade:

file /usr/sbin/amportal missing

So I ran:

cp /usr/local/sbin/amportal /var/lib/asterisk/bin/amportal

And then when I reload the amportal (amportal a reload)
I get the error:

Fetching FreePBX settings with gen_amp_conf.php..

Error(s) have occured, the following is the retrieve_conf output:
exit: -1

And /var/lib/asterisk/bin/amportal becomes an empty file. I cannot seem to properly copy it and it stays even after a restart.

Any advice is appreciated.

Thanks

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@parny wrote:

I recently migrated from Elastix 2.5, which was running on top of FreePBX 2.11.10 r17, to FreePBX-SNG7-FPBX-64bit-1707-1. Updated everything to current. Running Asterisk 13.17.1.

I can't get music on hold to work. When I got to settings/music on hold and create a new category, it doesn't show any available music files.

In musiconhold_additional.conf, there's the line:
directory=/var/lib/asterisk/moh/.nomusic_reserved
what does the .nomusic_reserved mean here? That there's no music here?

If I look in /var/lib/asterisk/moh, there are a number of files, including
macroform-cold_day.wav, manolo_camp-morning_coffee.wav, macroform-robot_dity.wav, reno_project-system.wav, macroform-the_simplicity.wav, and the same files with .alaw and .sln16 extensions. All are owned asterisk:asterisk.

So I think the music is there, and FreePBX isn't seeing it.

Any ideas on what I can do to fix this?

Thanks.

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@nevpro wrote:

Hi Team,

I have installed freePBX which comes in iso format within CentOS. I have static IP ---> Firewall/Router--->PBX server architecture in my office. All sip extension are working fine with PSTN as well as extension to extension in local LAN.But when I am registering extension (remote extension) out of my network with static IP (Forwarded port UDP 5060, 10000-20000 in my router), it gets register successfully, but when i am checking the seep peers this remote extension shows status as "UNREACHABLE". and not getting audio from onside and sometimes on both sides.

Please suggest any extra configuration which suppose to make to troubleshoot the issues.

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@perrcla wrote:

Hi,
the upgrade tool advises me to check if all my modules are eligibles for 13to14 upgrade (in system admin - activation)

But I don't understand.. please see the image below, no mentions about compatibility with v14.
So, can I try the upgrade or not?

image.png1090x449 53.8 KB

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@jerryriggin wrote:

I'm having an odd problem with AGI variable in FreePBX 13 with Asterisk 13.

It seems like the same effect as if the channel variables are in an array and the index is somehow offset by 1. I've come across this before some time ago but can't remember what fixed it.

Here's AGI code:

    #!/usr/bin/php
    <?php
    require_once('fafs_include.php');
    require_once('lime_include.php');
    date_default_timezone_set('America/New_York');

    $myagi = new AGI();
    $myagi->set_music(true,"default");

    $APIUser = get_var('APIUser');
    $msg="APIUser: $APIUser";
    $myagi->verbose($msg);
    $APIPass = get_var('APIPass');
    $msg="APIPass: " . $APIPass;
    $myagi->verbose($msg);
    $SiteURL = get_var("SiteURL");
    $myagi->verbose("SiteURL: " . $SiteURL);
    $TransferPhone = get_var("TransferPhone");
    $myagi->verbose("TransferPhone: " .  $TransferPhone);
    $ProdDiscountPct = get_var("ProdDiscountPct");
    $myagi->verbose("ProdDiscountPct: " .  $ProdDiscountPct);

Here's the CLI ouput:

<SIP/png_in-0000002e>AGI Rx << GET VARIABLE APIUser
<SIP/png_in-0000002e>AGI Tx >> 200 result=1 (Test_IVR)
<SIP/png_in-0000002e>AGI Rx << VERBOSE "APIUser: " 1
 find-orders.php: APIUser:
<SIP/png_in-0000002e>AGI Tx >> 200 result=1
<SIP/png_in-0000002e>AGI Rx << GET VARIABLE APIPass
<SIP/png_in-0000002e>AGI Tx >> 200 result=1 (test_pwd!123)
<SIP/png_in-0000002e>AGI Rx << VERBOSE "APIPass: Test_IVR" 1
 find-orders.php: APIPass: Test_IVR
<SIP/png_in-0000002e>AGI Tx >> 200 result=1
<SIP/png_in-0000002e>AGI Rx << GET VARIABLE SiteURL
<SIP/png_in-0000002e>AGI Tx >> 200 result=1 (https://devapi.ourcrm.com/admin/)
<SIP/png_in-0000002e>AGI Rx << VERBOSE "SiteURL: test_pwd!123" 1
 find-orders.php: SiteURL: test_pwd!123
<SIP/png_in-0000002e>AGI Tx >> 200 result=1
<SIP/png_in-0000002e>AGI Rx << GET VARIABLE TransferPhone
<SIP/png_in-0000002e>AGI Tx >> 200 result=1 (17277248006)
<SIP/png_in-0000002e>AGI Rx << VERBOSE "TransferPhone: https://devapi.ourcrm.com/admin/" 1
 find-orders.php: TransferPhone: https://devapi.ourcrm.com/admin/

As you can see the call to get_var() produces expected output, but the variable assignment is offset by 1. The first call returns nothing, 2nd returns first, 3rd returns second, etc. Like somehow an array assumed to start at 0 actually starts with 1. I have googled this till my fingers have calluses and dug through the PHPAGI code until my eyes are blurry and still stuck.

Any help SINCERELY appreciated.

Note: the 2 include files contain functions that are not called before this code, with the exception of get_var() which is copied from distro recordings.agi function agi_get_var().

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@StephanK wrote:

We were seriously hit by Hurricane Irma. Solar Panels still hanging 50 food up in a tree. Generator and UPS destroyed. Power still down. Working on rented generator.

When I restarted the PBX, Asterisk came on fine, but FreePBX gave me an error message about a corrupted database.

After research I am trying to repair as root with

mysqlcheck --repair --all-databases

and get

mysqlcheck: Got error: 1045: Access denied for user 'root'@'localhost' (using password: Yes) when trying to connect

So I tried to do it with a password, but am not able to get it to work.

I have tried all kinds of user/password combinations.

root / admin etc.

I saw in one error message, that FreePBX seems to connect to Asterisk with admin and the password that I set when setting up the system. I had meanwhile changed the password.

mysqlcheck --repair --all-databases --user admin --password=xxxxxx

What password should I use? Why is root not able to handle this?

Thanks for any help!

Stephan

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@alexandre17220 wrote:

Hi,

I want to make a ring group ring 5 seconds and if no answser another ring group ring 5seconds and call the first ring group if no answer.. it''s a loop, and it's the wish of my customer.
But there is a mechanism to block more 6 loops , like Lorne Gaetz tell me.

[2017-09-12 16:04:18] VERBOSE[10948][C-000000a2] pbx.c: Executing [s@macro-user-callerid:17] Set("PJSIP/156-0000033c", "__TTL=0") in new stack

How to disable this ?

Thanks,
Regards,

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@alexandre17220 wrote:

Hi,
Try to put 7 extensions in a ring group, with ringall strategy.
call the ring group internaly (from another extension).
Not all the extension ring, but all extension are connected...
Why ?
idem with queue

FreePBX 13.0.192.16 offical distribution
Reports/Asterisk info/Current Asterisk Version: 13.14.0

Thanks a lot

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