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Asterisk-version-switch - Asterisk 15 failed

October 18, 2017, 5:20 am
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@jersonjunior wrote:

Asterisk 15 does not initialize and generates dmesg error:
[ 118.846272] asterisk[6939]: segfault at 0 ip 00007f101993a921 sp 00007ffeb39d67f8 error 4 in libc-2.17.so[7f10197d5000+1b8000]
[ 124.082048] asterisk[8243]: segfault at 0 ip 00007fd9836ea921 sp 00007fff11ae0e38 error 4 in libc-2.17.so[7fd983585000+1b8000]
[ 129.430361] asterisk[9574]: segfault at 0 ip 00007f1ae65f1921 sp 00007ffe1abe6778 error 4 in libc-2.17.so[7f1ae648c000+1b8000]
[ 136.449842] asterisk[10934]: segfault at 0 ip 00007f6bf1a1c921 sp 00007ffd73f19768 error 4 in libc-2.17.so[7f6bf18b7000+1b8000]

CLI:
== res_stasis_snoop.so => (Stasis application snoop support)
Loading pbx_lua.so.
[2017-10-18 12:49:41] ERROR[2593]: pbx_lua.c:1642 load_or_reload_lua_stuff: Error loading extensions.lua: Unable to find 'extensions' table in extensions.lua

[2017-10-18 12:49:41] ERROR[2593]: loader.c:1246 load_resource_list: *** Failed to load module pbx_lua.so
[2017-10-18 12:49:41] ERROR[2593]: asterisk.c:4451 check_init: Module initialization failed. ASTERISK EXITING!

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How to configure the new Property Manager app

October 18, 2017, 6:52 am
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@posi211 wrote:

Checking out the new Property Manager app and need some help.

I have been wanting this app for a very long time. So I'm determined to make it work.

When I log in I get Warning! Your property is not configured. Please configure your property to continue.
So I check the config.

General, Hotel mode not slave.
Features, turn them all on.
Company, looks good to me
Email setting, look good here
Dialplan, auto-configured correctly.
Taxes, I don't think this part is completed. I can create a tax formula as I should with local, state and other components
Discounts, not going to give any as this time
Customer ticket, don't know what this does yet.

Add a room type, Double, $195.00 or something
On to add a room. and room type
in user manager need to create a group with HPMS as the type

I try to check a room in, but I have an error
There are no configured group privileges, please finish configuration before proceeding.

STUCK. can't log in a guest.

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PHP Fatal Error with IVR

October 18, 2017, 8:13 am
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@Das wrote:

/var/lib/asterisk/bin/module_admin reload
Error(s) have occured, the following is the retrieve_conf output:
exit: 255
PHP Fatal error: Call to undefined function recordings_get_file() in /var/www/html/admin/modules/ivr/functions.inc.php on line 96
Whoops\Exception\ErrorException: Call to undefined function recordings_get_file() in file /var/www/html/admin/modules/ivr/functions.inc.php on line 96
Stack trace:
1. () /var/www/html/admin/modules/ivr/functions.inc.php:96

version
+----------+-------------+
| variable | value |
+----------+-------------+
| version | 13.0.192.19 |
+----------+-------------+
1 row in set (0.00 sec)

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HowTo: attended internal call forwarding, configured but does not work

October 18, 2017, 8:34 am
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@Roger_Weihrauch wrote:

FreePBX 13.0.192.19
Current Asterisk Version: 11.25.1
Phones: Grandstream GXP1625 and GXP2140
Problem: Attended call forwarding does not work
Description:
When an external call comes in, we want to forward it attended.
We get the external call, pickup the phone, can hear the external caller, press the call-transfer key on the phone(s), enter the destination extension, BUT; a this point, the destination extension does not ring nor show the forwarded call in the display.
So we get no information on this forwarded call on the destinations extension.
Question(s):
HowTo configure FreePBXs extensionsnumbers to have this feature activated for all phones/extensions?
HowTo check/prove this feature is (already correctly) activated in FreePBX?
HowTo check in the phones admin area/webinterface which feature how to be configured?
Does someone know about grandstreams phones and how to activate/configure thisfeature`/function?
Thank you all for your highly appreciated support.
I am at the end with my Latin, as we say in german language. :wink:

Reagrds,
Roger

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FreePBX 14 - updating mysql libraries

October 19, 2017, 12:01 am
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@johedou wrote:

Hi everybody,

I wonder how to upgrade mysql libraries et related stuff as now mariaDB is installed. I use the freePBX pc to develop custom apps based on a remote mySQL database (v5.7.19).

When trying to install mysql clients and libs there's a conflict with mariaDB.

".......
Resolving Dependencies
--> Running transaction check
---> Package mariadb-libs.x86_64 1:5.5.56-2.el7 will be obsoleted
......
--> Finished Dependency Resolution
Error: Package: freepbx-14.1-1.sng7.noarch (@anaconda/1707)
Requires: mariadb
Removing: 1:mariadb-5.5.56-2.el7.x86_64 (@sng-base)
mariadb = 1:5.5.56-2.el7
Obsoleted By: mysql-community-client-5.7.20-1.el7.x86_64 (mysql57-community)
Not found
Error: Package: freepbx-14.1-1.sng7.noarch (@anaconda/1707)
Requires: mariadb-server
Removing: 1:mariadb-server-5.5.56-2.el7.x86_64 (@sng-base)
mariadb-server = 1:5.5.56-2.el7
Obsoleted By: mysql-community-server-5.7.20-1.el7.x86_64 (mysql57-community)
Not found
You could try using --skip-broken to work around the problem
You could try running: rpm -Va --nofiles --nodigest
....."

Any advices ?

Cheers

Jean-Olivier

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How to hide inbound phone number i softphone

October 19, 2017, 5:16 am
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@ed123 wrote:

How to hide an inbound number in softphone/hardphone? I tried to change it in Set caller ID and was able to hide it but unfortunately the crm link to our pbx was also affected. Our crm need to view the inbound number for it to process automatically.

additonal:
How many digits min/max in ${UNIQUEID} and ${EPOCH} ?

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Issues after first login

October 19, 2017, 10:17 am
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@stevenfrazier02 wrote:

I just installed from the ISO:

FreePBXSupport Dashboard ISymphonyV3 Panel UCP.

I can't get back into the configuration. I am getting in as "http://hostname/admin/config.php#.

Is there another URL that I should be using. I can login with my user name/password but no configuration tab.

TIA.

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Assign DID numbers

October 19, 2017, 11:19 am
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@papelini wrote:

Hello,

Hello i'm new to the FreePBX world. I have setup two SIP servers and I made a trunk between the two servers and I can dial to extension numbers. Now I would like to setup DID numbers like +32473781627 so users can be reached thru the DIDs numbers. Any idea how I can make this happen

Pape

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Low incoming audio on SIP trunks

October 19, 2017, 1:44 pm
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@edlentz wrote:

Asterisk version 13.15.0
FPBX version 10.13.66.20
SIP trunks

This is a new one on me. Intermittantly, about 15% of the calls the system handles has an inconsistant incoming audio level. What I mean is that the callers voice will start out just fine and as the call progresses the caller volume level will decrease to a point where the call will drop. Sometimes the volume level will return to normal. The SIP trunks are being sent out over a Fiber connection. Chan-sip info shows the connect time <30ms. This audio problem is also exhibited when a caller leaves a voicemail message as well, so we do not suspect the phones. We have TCPdump running but have not captured a bad call yet. SIP trunk provider issue? Something going on with the Fiber? As I said this is new to me. Not one complaint from the outside that they cannot hear the user.

Ideas??

Thanks

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Sangoma FreePBX Phone System 60: Multiple FreePBX errors out of the box

October 19, 2017, 2:19 pm
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@rchase wrote:

A client of mine just got a System 60 from VoIPSupply and it came with lots of errors

1.

New Interface Detected
A new, unconfigured, network interface has been detected. Please assign interface 'eth0' to a zone.


2.

Trusted Interface Detected
A network interface that is assigned to the 'Trusted' zone has been detected. This is a misconfiguration. To ensure your system is protected from attacks, please change the default zone of interface 'eth1'.

3.

retrieve_conf failed, config not applied
Reload failed because retrieve_conf encountered an error: 255

exit: 255
Unable to continue. Error with the templating engine: array_merge(): Argument #2 is not an array:: in /var/www/html/admin/libraries/utility.functions.php on line 207#0 /var/www/html/admin/libraries/BMO/Less.class.php(209): die_freepbx('Error with the ...')
#1 /var/www/html/admin/libraries/BMO/Less.class.php(94): FreePBX\Less->getCachedFile('/var/www/html/a...', '/admin/assets/d...', Array)
#2 /var/lib/asterisk/bin/retrieve_conf(162): FreePBX\Less->generateModuleStyles('directory')
#3 /var/lib/asterisk/bin/retrieve_conf(543): connectdirs->generate_less('directory')
#4 {main}

4.

OpenCNAM Requires Authentication
Unauthenticated calls to the OpenCNAM API will soon fail. You will need an OpenCNAM account to continue using their API

5.

No Trusted Network or Host defined
No Trusted Network or Host has been defined. Every server should have a 'Trusted' host or network to ensure that in case of configuration error, the machine is still accessible.

3 is the most problematic because we are unable to Apply Config, it gives the error every time, so the system is totally unusable at the moment

I have tried:
fwconsole chown
Download and update all modules
yum update
fwconsole ma upgrade core
fwconsole ma refreshsignatures
fwconsole ma install -f manager

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FreePBX Shutdown Firewall/MYSQL Error

October 19, 2017, 3:33 pm
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How to fetch Calls logs using REST Api

October 20, 2017, 1:05 am
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@yeasindsi wrote:

I am running FreePBX 14.0.1.14. I have Installed the REST API module.

According to Documentation I have Used the Curl command given in the documentation using my Token & server url. Following is what I get:

% Total % Received % Xferd Average Speed Time Time Time Current
Dload Upload Total Spent Left Speed
100 238 0 238 0 0 801 0 --:--:-- --:--:-- --:--:-- 1515{"error":{"type":"Exception","message":"Attempted to loa d Rest with a hint of \/var\/www\/html\/admin\/modules\/rest\/Rest.class.php and it didn't exist","file":"\/var\/www\/html\/admin\/lib raries\/BMO\/Self_Helper.class.php","line":161}}

My goal is to retrieve the unanswered calls of a queue as json.
Can someone help me how to solve this. Or What are my other options.

Thanks in advance

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Show "missed" calls in ring group even if they were picked up

October 20, 2017, 6:18 am
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@adtopkek wrote:

In older versions of Freepbx if a ring group has 5 members and a call comes in with phone #1 answering then phone #2-#5 would show a missed call.

We upgraded a client to FreePBX 13 and now the ring groups are not doing that anymore. It doesn't happen on Both Yealink and Grandstream phones. We have enabled missed calls and full call history on the Yealinks but with no luck.

Has anyone gotten this to work on FreePBX 13?

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EPM - File not found displayed on phone at end of restart, config not working

October 20, 2017, 6:55 pm
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@JessicaRabbit wrote:

I guess I broke EPM. Provisioning seems to now to be not working. Phones no longer respond to restarts from EPM. At end of boot, "FILE NOT FOUND" flashes, and pre-modification config is displayed.
Any changes in config do not appear unless factory reset and restart but "file not found" still appears.
I have no idea how to fix this. Any ideas would be appreciated.

10.13.66-21 32 bit Distro
Aastra

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MOH mpg123 TripleM Melbourne Australia

October 22, 2017, 8:40 am
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@mixhali wrote:

HI,

I have a client who wants to have the Melbourne radio station MMM as the music on hold. However I just can't get it to work. I've never configured MOH using a stream before however found this really useful guide:

http://www.voneto.com/blogger/2012/06/27/streaming-radio-music-on-hold-for-freepbx-and-asterisk/

I managed to get everything configured however it doesn't work. It sems to be the stream itself is maybe not suitable or I have not configured it 100%

In the MOH setting I have the application set to:

/usr/bin/mpg123 -q -r 8000 -f 8192 --mono -s http://sc02.scahw.com.au/3mmm_128

I get no audio when on hold.

If I try to run the same command ont he CLI I get:

[root@localhost bin]# mpg123 -q -r 8000 -f 8192 --mono -S http://sc02.scahw.com.au/3mmm_128
main: [src/mpg123.c:679] error: Unknown mpeg MIME type audio/aacp - is it perhaps a playlist (use -@)?
main: [src/mpg123.c:680] error: If you know the stream is mpeg1/2 audio, then please report this as mpg123 bug

Anyone done this before?

Rgeards

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FreePBX on VM

October 22, 2017, 9:34 am
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@Joeralph wrote:

Hi, badly need some help.

I am new in PBX world and trying to setup a Voip with my phone(w/ softphone) and Computer. I installed FreePBX-Asterisk to the virtual machine VMWare as my server for my computer and phone. i installed Softphone to my computer and registered to the FreePBX and i could dial and ring my Softphone. however, i could not connect external device to the FreePBX. in my case, my phone with Zoiper installed. with Chan_SIP
Host:IP of PBX:5160
Username:Extension
Password:secret.
it is always as Not Registered or timeouts.

FPBX IP: 192.168.127.128
CompIP: 192.168.8.100
PhoneIP: 192.168.8.101

im registering multiple times and somehow believe that my phone is being blocked by fail2ban.
do i have to configure that theyll be in the same network? but my softphones in PC still works fine with that IPs.

Thanks for help

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EPM - Base file edit error

October 23, 2017, 4:28 pm
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pfSense: cant register phone to internet based FreePBX

October 23, 2017, 5:05 pm
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@rchase wrote:

Anyone know what could be the cause of this?

-- Registered SIP '200' at 73.19.22.1:17820
[2017-10-23 20:05:03] NOTICE[1772]: chan_sip.c:23797 handle_response_peerpoke: Peer '200' is now Reachable. (80ms / 2000ms)
-- Unregistered SIP '200'

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Vega 50 vs Vega 60 Analog FXS

October 23, 2017, 6:31 pm
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@dmanolis79 wrote:

What is the difference between the 2 models, is the hardware better? Also in the EPM why can't i configure the Vega 50 like the sangoma phones?

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Search Domain Not Set?!?

October 23, 2017, 8:45 pm
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@vbman213 wrote:

Any particular reason why System Admin doesn't fill in /etc/resolv.conf based on the suffix of the FQDN from the Hostname settings page? Or at least add an entry to the DNS setings page to manually configure search domains.

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