@telili wrote:
Good morning all, for a digital card I setting up two groups, the trunk is on E1 mode, therefore 1-14 as group2 and 15-30 as group3, for the first fourteen channels I want to set a range of DID numbers for outbound calls and all the rest will use for inbound call. Any idea will appreciated.
Thanks.
Posts: 1
Participants: 1
@jasonmel wrote:
I disabled it and now have the following error when I hit 'Apply Config':
exit: 1
[FATAL] SQLSTATE[HY000] [2005] Unknown MySQL server host 'localhost:3306' (1) SQLSTATE[HY000] [2005] Unknown MySQL server host 'localhost:3306' (1)Trace Back:
/var/www/html/admin/libraries/BMO/Database.class.php:70 PDO->__construct()
[0]: mysql:host=localhost:3306;dbname=asteriskcdrdb
[1]: freepbxuser
[2]: 68e05d5f933e/var/www/html/admin/modules/cdr/Cdr.class.php:35 Database->__construct()
[0]: mysql:host=localhost:3306;dbname=asteriskcdrdb
[1]: freepbxuser
[2]: 68e05d5f933e/var/www/html/admin/libraries/BMO/Self_Helper.class.php:116 Cdr->__construct()
[0]:/var/www/html/admin/libraries/BMO/Self_Helper.class.php:36 Self_Helper->autoLoad()
[0]: Cdr/var/www/html/admin/libraries/BMO/Hooks.class.php:163 Self_Helper->__get()
[0]: Cdr/var/www/html/admin/libraries/BMO/Hooks.class.php:37 Hooks->preloadBMOModules()
/var/lib/asterisk/bin/retrieve_conf:26 Hooks->updateBMOHooks()
Posts: 18
Participants: 7
@iCapof85 wrote:
Hi Guys,
I tried to FreePBX 12 from an USB image that I downloaded from the official website. The installation goes fine but when It's done I don't get asterisk or FreePBX in general, just Centos 6.5.
I followed the steps from http://wiki.freepbx.org/display/FD/Installing+FreePBX+Official+Distro and I noticed that I don't get the screen in Step 4 (The welcome screen that gives the options between Asterisk 11 and 13).
I looked at this topic http://community.freepbx.org/t/freepbx-distro-doesnt-install-freepbx-or-asterisk/26962 but I don't get an option to choose from templates.
Posts: 1
Participants: 1
@jaymin wrote:
Hello All, I have one VoIP based application which will run on Olimex board. Currently pjsip version 1.8 use in application. Pjsip on board should talk with Asterisk on host machine. But on Olimex board A20 its not working properly. Pjsip restart some times it self and sometimes it does not find audio device while running on board.I write down below some of my configuration details below. For that different types of configuration which passed to pjsib final compiled binary as below: ["--ec-tail=0","--add-codec=G722","--snd-auto-close=-1","--no-tones","--use-timer=0","--timer-se=86400","--timer-min-se=86300","--no-vad","--auto-answer=200","--log-level=6","--ptime=10","--play-file=s1.wav","--reg-timeout=60","--rereg-delay=60"] Is something missing or need to be modify on above configuration? I am facing problem to compile the pjsib. The results and observations of compilations are quit unexpected sometime. It compiled without any errors and sometimes it not compiled while configuration arguments same both times. Tool-chains used for compilation are Linaro gnueabihf. The configuration setting is same in all condition. Observations are: ./configure --host=arm-linux-gnueabihf --disable-floating-point LIBS=-lm 1. Toolchain - arm-linux-gnueabihf-gcc-4.7.3 - pjproject 2.4 - Not compiled Error - .pjsua2-lib-arm-unknown-linux-gnueabihf.depend:1: *** missing separator. Stop. 2. Toolchain - arm-linux-gnueabihf-gcc-4.8.3 - pjproject 2.4 - Not compiled Error - .pjsua2-lib-arm-unknown-linux-gnueabihf.depend:1: *** missing separator. Stop. 3. Toolchain - arm-linux-gnueabihf-gcc-4.7.3 - pjproject 1.8 - Not compiled Error - .pjsua2-lib-arm-unknown-linux-gnueabihf.depend:1: *** missing separator. Stop. 4. Toolchain - arm-linux-gnueabihf-gcc-4.8.3 - pjproject 1.8 - compiled successfully(as its generate binary) binary name - pjsua-arm-unknown-linux-gnueabihf Any ideas related to this error. I saw all Makefile in pjproject and also checked about space that whether it is with 'TAB KEY SPACE' or without. But did not find any error, for that reason I did not make any conclusion that for which reason problems occurs. Provide me some informations and details that How and which configuration arguments need to passed to pjsip binary for running on A20 processor? Which codecs need to passed and how audio device added to pjsip? (Mean to ask that if audio coded is on A20 processor it self then How to enable it? and How to pass it to pjsip? and Is it necessary to provide audio coded G722 to pjsib?)Any help and suggestions will be helpful for me.
Regards Jaymin D
Posts: 2
Participants: 1
@BrettKuntz wrote:
Another employee here was making a few changes regarding extensions and this error came up when he tried to apply the changes. We have tried rebooting the server as well as trying to execute "amportal a r" which gives the below output:
-bash-4.1# amportal a r Fetching FreePBX settings with gen_amp_conf.php.. Error(s) have occured, the following is the retrieve_conf output: exit: 255 found language dir en_UK for broadcast, not installed on system, skipping found language dir fr for directory, not installed on system, skipping found language dir en_UK for pagingpro, not installed on system, skipping Added to globals: VM_OPTS = s Added to globals: VM_DDTYPE = u Added to globals: VM_GAIN = 15 Added to globals: OPERATOR_XTN = Added to globals: VMX_TIMEOUT = 5 Added to globals: VMX_REPEAT = 1 Added to globals: VMX_LOOPS = 1 Added to globals: VMX_OPTS_LOOP = Added to globals: VMX_OPTS_DOVM = Added to globals: ASTETCDIR = /etc/asterisk Added to globals: ASTMODDIR = /usr/lib/asterisk/modules Added to globals: ASTVARLIBDIR = /var/lib/asterisk Added to globals: ASTAGIDIR = /var/lib/asterisk/agi-bin Added to globals: ASTSPOOLDIR = /var/spool/asterisk Added to globals: ASTRUNDIR = /var/run/asterisk Added to globals: ASTLOGDIR = /var/log/asterisk Added to globals: CWINUSEBUSY = true Added to globals: AMPMGRUSER = admin Added to globals: AMPMGRPASS = >>edited out Added to globals: AMPDBENGINE = mysql Added to globals: AMPDBHOST = localhost Added to globals: AMPDBNAME = asterisk Added to globals: AMPDBUSER = freepbxuser Added to globals: AMPDBPASS = >>edited out Added to globals: VMX_CONTEXT = from-internal Added to globals: VMX_PRI = 1 Added to globals: VMX_TIMEDEST_CONTEXT = Added to globals: VMX_TIMEDEST_EXT = dovm Added to globals: VMX_TIMEDEST_PRI = 1 Added to globals: VMX_LOOPDEST_CONTEXT = Added to globals: VMX_LOOPDEST_EXT = dovm Added to globals: VMX_LOOPDEST_PRI = 1 Added to globals: MIXMON_DIR = Added to globals: MIXMON_POST = Added to globals: DIAL_OPTIONS = Ttr Added to globals: TRUNK_OPTIONS = Tt Added to globals: TRUNK_RING_TIMER = 300 Added to globals: MIXMON_FORMAT = gsm Added to globals: REC_POLICY = caller Added to globals: RINGTIMER_DEFAULT = 20 Added to globals: TRANSFER_CONTEXT = from-internal-xfer PHP Fatal error: Uncaught exception 'Exception' with message 'Endpoint already exists.' in /var/www/html/admin/modules/core/functions.inc/PJSip.class.php:544 Stack trace: #0 /var/www/html/admin/modules/core/functions.inc/PJSip.class.php(399): FreePBX\modules\Core\PJSip->generateEndpoint(Array, Array) #1 /var/www/html/admin/modules/core/functions.inc/PJSip.class.php(86): FreePBX\modules\Core\PJSip->generateEndpoints(Array) #2 /var/www/html/admin/libraries/BMO/FileHooks.class.php(92): FreePBX\modules\Core\PJSip->genConfig() #3 /var/www/html/admin/libraries/BMO/FileHooks.class.php(25): FileHooks->processNewHooks() #4 /var/lib/asterisk/bin/retrieve_conf(740): FileHooks->processFileHooks(Array) #5 {main} thrown in /var/www/html/admin/modules/core/functions.inc/PJSip.class.php on line 544Does anyone know how to solve this problem? Thanks!
Posts: 5
Participants: 2
@MAA wrote:
Hello,
FreePBX 6.12.65-28
I have 2 sip-trunk to external world.
I need:
Calls from extensions 101, 102, 103 to external go through Trunk1
Calls from other extensions to external go through Trunk2
How can I do that?
Posts: 4
Participants: 2
@swaterhouse wrote:
I have always just built my own servers from scratch but thinking about using the distro on some fresh hardware I am ordering soon. Are there plans on updating to the latest clone of Redhat 7?
My hardware refreshes are every 5-6 years, that puts me dangerously close to having support lapse on the 6x branch I am not a fan of in-place SO upgrades.
Posts: 2
Participants: 2
@jasonmel wrote:
Hi all,
PBX Firmware:6.12.65-28
PBX Service Pack:1.0.0.0
FreePBX 12.0.74Fresh install of Distro from iso FreePBX-32bit-6.12.65
Can't activate, keeps looping to the "This machine is not currently activated." page. Sometimes I get to the "Thanks for installing FreePBX" but when I click activate I go back to the first activation page. Tried with Firefox, Chrome and IE.
The DNS is fine and I do not appear to have any delay when pinging google.com. Module updates work fine and I upgraded with upgrade-6.12.65-28.sh to see if it would fix the problem. Upgrade worked fine, but the problem persists. Tried starting over with with fresh install, no change.
Running amportal a dbug and hitting activate gives,
[2015-Jul-25 13:54:29] [PHP-NOTICE] (/var/www/html/admin/modules/sysadmin/Oobe.class.php:90) - Undefined index: ampconf
[2015-Jul-25 13:54:29] [WARNING] (libraries/modulefunctions.legacy.php:7) - Depreciated Function module_getinfo detected in /var/www/html/admin/libraries/featurecodes.functions.php on line 42
[2015-Jul-25 13:54:29] [PHP-NOTICE] (/var/www/html/admin/modules/sysadmin/functions.inc/general.php:41) - Undefined index: deploymentid
[2015-Jul-25 13:54:38] [PHP-NOTICE] (/var/www/html/admin/modules/sysadmin/page.sysadmin.php:98) - Undefined index: deploy_type
[2015-Jul-25 13:54:38] [PHP-NOTICE] (/var/www/html/admin/modules/sysadmin/page.sysadmin.php:98) - Undefined index: deploy_type
[2015-Jul-25 13:54:38] [PHP-NOTICE] (/var/www/html/admin/modules/sysadmin/views/rnav.php:42) - Undefined index: deploy_type
[2015-Jul-25 13:54:38] [PHP-NOTICE] (/var/www/html/admin/modules/sysadmin/views/rnav.php:42) - Undefined index: deploy_type
[2015-Jul-25 13:54:38] [WARNING] (libraries/modulefunctions.legacy.php:7) - Depreciated Function module_getinfo detected in /var/www/html/admin/libraries/featurecodes.functions.php on line 42
[2015-Jul-25 13:54:38] [PHP-NOTICE] (/var/www/html/admin/modules/sysadmin/functions.inc/general.php:41) - Undefined index: deploymenThank you,
Posts: 2
Participants: 1
@atcom wrote:
I'm backing up the /tftpboot directory but the exclusions don't seem to work. Is something wrong with my syntax?
![]()
I was also hoping to exclude some of the module files to reduce size but that's not working either:
Posts: 5
Participants: 3
@avraham wrote:
we need to upgrade our freepbx/asterisk server in a lab/educational environment.
i'd be happy to hear opinions of whether we should start w/ the
AsteriskNOW-612-current-64
or
FreePBX-64bit-6.12.65images.
i understand that they're extremely similar, but not identical. it will be used in a lab for a VOIP
course in an MS program in Networking/Communications.
tnx,
ams
Posts: 1
Participants: 1
@Sc00by wrote:
HI all
I am currently running FreePBX 2.11 Asterisk 11 and have been having issues with OpenVPN for a few months now.
I have OpenVPN running on FreePBX which was installed through the standard yum repos
I believed I have traced the issue I am having back to a bug in that the iPhone iOS client is causing the OpenVPN server to restart
OpenVPN 2.0.9 is the only version on the Schmooze repo which was from 2010. Would it be possible for the latest rpm to be made available please?
Posts: 10
Participants: 3
@el_es wrote:
Hi,
sorry for being a bit thickmy current distro/fw version is 5.211.65-12. My FreePBX version is 12.0.74. My Asterisk is 11.14.2.
I regularly run module updates, also sometimes I do just 'yum update' from command line. I have never ran ANY of the update scripts before.
Does the (quoted below) text on http://wiki.freepbx.org/display/FD/FreePBX-Distro-5.211.65 this page mean, I should START the upgrades FROM the script FOR the version, i.e. http://upgrades.freepbxdistro.org/stable/5.211.65/upgrade-5.211.65-12.sh, then run EVERY next script till I get to http://upgrades.freepbxdistro.org/stable/6.12.65/upgrade-5.211.65-to-6.12.65-20.sh, then every script in the stable track (http://upgrades.freepbxdistro.org/stable/6.12.65/upgrade-6.12.65-20.sh to http://upgrades.freepbxdistro.org/stable/6.12.65/upgrade-6.12.65-28.sh) too ?
Basically: do I start FROM the script with my current EXACT number then follow up from that?
TIA.
All upgrades need to be installed in numeric ascending order. Do not
skip any upgrade step. Upgrade scripts are not cumulative. Each upgrade
script should be run in ascending order to get to the desired final
version.
Step 1: Check the current FreePBX Distro versionDisplay the current version file (as above) to confirm the current installed version of FreePBX Distro.Step 2: Download and run the applicable upgrade script
To install an update script via the Linux command line, use the following
commands, substituting the proper scripts from above. Download the
upgrade script matching the current version of FreePBX Distro installed,
mark it as executable and run it.
Posts: 2
Participants: 1
@chivasregal_ wrote:
Im curious if anyone is using FreePBX as a collaboration solution, specifically the ability to provide:
- Presence in Outlook/Email Client - or somewhere else?
- Video Conferencing
- Chat solution
I understand the Cisco BE6000 uses Jabber to achieve some of the above, but what is a solution for FreePBX users?
Be great to hear someone achieving some of the above.
Posts: 1
Participants: 1
@alexeynikolaev wrote:
Hi, community!
I have two FreePBX Distro installations. Then i trying to login to new instance via https i have got this error sec_error_reused_issuer_and_serial. Mozilla Firefox wan't get me access via https.Certificate contains the same serial number as another certificate
How can i avoid this problem using System Admin module? @xrobau
Posts: 4
Participants: 3
@alexeynikolaev wrote:
Hi, all!
Is there some documentation how to setup external authentication via Active Directory in the FreePBX 13 User Manager module?
Posts: 10
Participants: 4
@zhaolin wrote:
Hi,
I'm new to FreePBX/Asterisk need some help..
I am going to setup Freepbx system in our office. Everything is fine. We can make calls to outside world and do call transfer internally. But when we try to call extensions in the office from outside, calls keep dropping after only 1 second. I have contacted our voip provider and trunk settings has been checked. It should be OK.The below is sip debug output when we make calls from outside.
=======================================
Jack
Posts: 10
Participants: 3
@MAA wrote:
Hello,
Question about Reports - CDR Reports
How can i add to "Call Detail Record" report additional column "Trunk"? (trunk, through which the call has come, or gone)
Posts: 2
Participants: 2
@AdHominem wrote:
I just spent the weekend setting up the latest Distro.
I've found that sometimes after I make a change and hit the red apply config button at the top all incoming calls to the system will fail. When that happens, all ten of my trunks will show as unregistered. Qualify packets continue to be sent and received, even to the external trunks.Outbound calls continue to work, and internal calls from one extension to another continue to work.
When I check the logs, the outbound calls show that they are going out, but inbound calls are never logged. Systems sending calls to my system report that the calls are "declined." Here's a sample log from the other system that is sending calls to me:
[2015-08-23 17:40:08] VERBOSE[5786][C-0000006d] app_dial.c: – Called SIP/[REDACTED]/[REDACTED]
[2015-08-23 17:40:08] VERBOSE[1931][C-0000006d] chan_sip.c: – Got SIP response 603 "Declined" back from [REDACTED]:5060
[2015-08-23 17:40:08] VERBOSE[5786][C-0000006d] app_dial.c: – SIP/[REDACTED]-000000cf is busy
[2015-08-23 17:40:08] VERBOSE[5786][C-0000006d] app_dial.c: == Everyone is busy/congested at this time (1:1/0/0)Doing further apply changes will not remedy the problem.
However, an "amportal restart" or "service asterisk restart" or "reboot" does resolve the issue.
When I opened a bug ticket on this issue, the Devs closed it claiming that it was a configuration issue or an Asterisk bug, and that I needed paid support. I've read somewhere that Asterisk 11 has a bug related to including comments.
Does anyone have any suggestions on this issue?
Posts: 1
Participants: 1
@hanscanderson wrote:
/usr/bin/mpg123 -q -s --mono -r 22500 -f 8192 -b 1024 XXXX://10.1.3.250
The XXXX is to get the link statement to be accepted by the post. Replace by http for correct statement entered into FreePBX. This statement is accepted by the GUI Application update process.
The label for the category "ON HOLD IP" shows up in the file folder /var/lib/asterisk/moh.
I can not get the stream MOH to take the place of the "default" files and run the MOH audio from the IP stream.
How do you tell FreePBX to point to the Streaming Category?
Posts: 6
Participants: 4
@asnier wrote:
hi all
now my bot trunk works for incoming calls
trunk 1 - dongle 4177xxxxxxx nr
trunk 2 - sipcall 4132xxxxxxx nr
i have total 7 extension 3 of them now online
i have trouble with outgoing calls with both trunk
trunk dial plan as follow
prepend prefix match pattern
00. 00NXXNXXXXXXX 117 118 14400 NXXNXXXXXXX
032 NXXXXXX
my first question when i want to go out from dongle trunk i want use 7
and 9 for sipcall trunk.where i have to right this 7 and 9 at prefix (
outbound or trunk or outbound dial prefix option on trunk ) ?second and biggest problem
when i try to go out with dongle or sipcall i get only same message
NUMBER YOU HAVE DIALED IS NOT IN SERVICE - i hate this i try to fix problem last two days then i decide ask here bigboss.
wait for your kindly help
best regards
Posts: 1
Participants: 1
