October 30, 2016, 7:52 am
@aryncavage322 wrote:
We just switched internet providers in our office. I am now just working on the FreePBX machine, but it wont let me connect via online or via console cable.
To start, the online port, set by the switch is in the scheme of 192.x.x.x (what it used to be) when now it is supposed to be in the scheme of 10.x.x.x.
Next, when I try to direct connect with the cable and Putty, it never works. It simply makes a sound signifying that it cannot connect. I have confirmed the COM port as well. Note, this is my first time doing a direct connect.
Do I need to factory reset the Sangoma box to get its new IP address? Also, what I am doing wrong with Putty?
Thank you!
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October 30, 2016, 1:06 pm
@dcitelecom wrote:
I have a somewhat older installation of the FreePBX distro and started the upgrade from 5.211.65-19 to 10.13.x.x but we got stuck after the FreePBX Distro 6.12.65-22 upgrade. Module admin now shows only 2 modules.
FreePBX Framework 2.11.0.43
System Admin 2.11.0.59
Check Online does not allow to update the modules
The "12.0 Upgrade Module" says "You appear to have partially upgraded to version 12.0, your current version number is 12.0.0alpha1"
Goto module admin and upgrade FreePBXFramework but module admin doesn't let me upgrade.
According to the upgrade instructions "FreePBX Disrto 5.211.65-100" will take your 5.211.65 version system to a 6.12.65-20 version but it does not look like it worked.
I am also getting the error below each time I run amportal start/restart/upgradeall, etc...
SETTING FILE PERMISSIONS
chattr: Operation not supported while reading flags on /var/www/html/cxpanel
Permissions OK
Can I salvage this system?
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October 31, 2016, 3:32 am
@digiteltlc wrote:
Perhaps not the proper way to do....
FreePBX 10.13.66-16
It executes daily (crontab) this:
!/bin/sh
sudo -u asterisk /var/lib/asterisk/bin/module_admin upgradeall
sudo -u asterisk /var/lib/asterisk/bin/module_admin reload
in order to keep all modules upgraded
Sometimes it no more connects to asterisk (retrieve.conf error) and I need to issue manually the commands
fwconsole ma upgrade framework
fwconsole reload
to get it working again
( sometimes fwconsole ma remove restapps )
Is this a "normal" issue due to a bad way to proceed with automatic upgrades ?
I noted that
26 0 19 * * /usr/sbin/sysadmin_update_system -a > /dev/null 2>&1
is also present in crontab , can this conflicts some way with daily script ?
Thank you
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October 31, 2016, 10:36 pm
@hipark wrote:
Dear friends
while checking my CDR reports, things like number of unanswered calls are not the same as I see in Asternic module. in this case which of them are more reliable?
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November 1, 2016, 2:14 am
@dirweb wrote:
Hi
i made some audio file for music on hold, but when i upload it on my freepbx i less quality.
How is possible to upload it without less quality?
Thank you
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November 1, 2016, 2:52 am
@bhaskart wrote:
i am using SugarCRM software in web server and FreePBX in local server. how to integrate both ? can any one suggests me that will helpful for me thanks in advance.
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November 1, 2016, 3:39 am
@ipvinner wrote:
Hello, could somebody help to. installed FreePBX 13.0.171 and Backup and restore13.0.25. Created new backup and set up Run Automatically: weekly. backup created, but crontab is
crontab -u asterisk -l
* * * * * [ -x /var/www/html/admin/modules/dashboard/scheduler.php ] && /var/www/html/admin/modules/dashboard/scheduler.php
58 * * * * /var/lib/asterisk/bin/freepbx-cron-scheduler.php
10 * * * * /usr/sbin/fwconsole util cleanplaybackcache -q
* * * * * [ -x /var/lib/asterisk/bin/schedtc.php ] && /var/lib/asterisk/bin/schedtc.php
so rule for weekly backup has not created.
I tried to fwconsole reload.
/var/log/asterisk/freepbx.log
WARNING - Depreciated Function module_getinfo detected in /var/www/html/admin/modules/backup/bin/backup.php on line 11
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November 1, 2016, 6:45 am
@chrischevy wrote:
In FreePBX13, I know that I can record my prompts in more than one language. This prevents me from duplicating my IVR structure (one for each language). It's a really great feature but I have a little problem:
Here's the scenario:
In my welcome message, I say "for service in french, please press 9"
When pressing 9, I switch the language to french and I use the next IVR as the destination.
For now, everything is fine. However, the customer wants to be able to switch back and forth from English to French in any IVR. Because I'm using a unique IVR structure with mulilingual system recordings, I cannot use the same "9" key to switch automatically from French to English or vice-versa.
I'm having a hard time explaining my problem, but what I want to know is:
Is there a way to program a function like: When pressing 9, if current language is french, switch to english. If current language is english, switch to french.
Having a duplicate IVR structure, while behing cumberstone, permits this because I have a french and an english version of every IVR
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November 1, 2016, 6:47 am
@kaumell wrote:
More or less as the topic states.
I have recorded greetings for busy and unavailable, if I login to the UAC it is there, but when I call the extension and voicemail picks up, I am getting the canned default system greeting
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November 1, 2016, 7:57 am
@gedeon999 wrote:
Hello,
I made a new configuration some week ago,
The basis was Asterisk 11.21.0 / FreePBX 13.0.74.
Then I upgrade everything until two week ago.
Today I rebooted the system for the first time and it blocks
PHP Warning: The use statement with non-compound name 'Symfo' has no effect in /var/www/html/admin/libraries/Composer/vendor/symfony/polyfill-php55/bootstrap.php on line 12
PHP Parse error: syntax error, unexpected '~', expecting ',' or ';' in /var/www/html/admin/libraries/Composer/vendor/symfony/polyfill-php55/bootstrap.php on line 12
**** WARNING: ERROR IN CONFIGURATION ****
astrundir in '/etc/asterisk' is set to but the directory
does not exist. Attempting to create it with:
'mkdir -p '
mkdir: missing operand
Try 'mkdir --help' for more information.
**** ERROR: COULD NOT CREATE ****
Attempt to execute 'mkdir -p ' failed with an exit code of 1
You must create this directory and the try again.
Does someone have an idea ?
Thanks !
Patrick
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November 1, 2016, 8:16 am
@gene778 wrote:
Asterisk 13 Freepbx 12 on Debian 8
My custom destinations still seem to be working, but "none" of the dozen
or so that I have created show up on the Custom Destinations page!
Several days ago I truncated the Kvstore database because it was so
large I was getting errors. Is is possible I deleted some of the data
that was provided to the Custom Destinations module?
How can I restore this information? I am afraid to add a new destination,
fearing it will delete the currently working, but now NOT shown, destinations.
Thank you,
Gene in Virginia
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November 1, 2016, 11:12 am
@dbaddorf wrote:
Good afternoon! I am replacing an old legacy phone system which had the capability of a user picking up a call while the caller was leaving a voicemail. Can this be done in FreePBX? I've been trying to Google for answers but I haven't been able to find anyone asking about this specific feature in FreePBX. If anyone has any input, I would certainly appreciate it!
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November 1, 2016, 5:22 pm
@MC1 wrote:
Hello
I have two different trunks on my system: an analog trunk and a sip trunk.
The analog trunk I use for 911 and 511 dial outs.
All other calls (local, long distance + toll free) go out on the sip trunk.
Question: how can I verify what trunk a dialed number is going out on? For example if I dial 511 I'd like to verify that it went out on the analog trunk as it match the dial patterns for that trunk in the outbound routes. Similarly if I dialed 1-866-XXX-XXXX I'd like to be able to verify that it went out on the sip trunk (because the dialed number match the dial pattern for that outbound route).
I thought CDR would provide me with that information. Perhaps I haven't configured the report incorrectly? I've checked the wiki and the forums before submitting my question.
Thanks in advance for any help that you can provide.
FreePBX 13.0.190.1
Firmware 10.13.66-16
Michael
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November 1, 2016, 5:32 pm
@milazoila wrote:
Is there a setting on the GUI where to set 30days voicemail retention?
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November 1, 2016, 11:20 pm
@sahklh wrote:
I am running the current distro of 10.13.66, when we setup a ring group and assign an outside number to confirm calls, when the person being called presses #1 it will not connect the call, but will say the call is no longer available every time.
I was able to get it to work with queues but not with ring groups. Any ideas? Thanks
Scott
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November 2, 2016, 12:26 am
@asifahmed009 wrote:
Hi guys ,
Facing this problem with many of extensions.
Extension Seems Registered perfectly but can dial only outgoing calls no incoming neither from extension nor trunk.
This happening with several extensions
- Name : 1928
Description :
Secret :
MD5Secret :
Remote Secret:
Context : from-internal
Record On feature : automon
Record Off feature : automon
Subscr.Cont. :
Language :
Tonezone :
AMA flags : Unknown
Transfer mode: open
CallingPres : Presentation Allowed, Not Screened
Callgroup :
Pickupgroup :
Named Callgr :
Nam. Pickupgr:
MOH Suggest :
Mailbox : 1928@device
VM Extension : asterisk
LastMsgsSent : 0/0
Call limit : 2147483647
Max forwards : 0
Dynamic : Yes
Callerid : "1928" <1928>
MaxCallBR : 384 kbps
Expire : 3410
Insecure : no
Force rport : No
Symmetric RTP: No
ACL : Yes
DirectMedACL : No
T.38 support : No
T.38 EC mode : Unknown
T.38 MaxDtgrm: 4294967295
DirectMedia : No
PromiscRedir : No
User=Phone : No
Video Support: No
Text Support : No
Ign SDP ver : No
Trust RPID : Yes
Send RPID : No
TrustIDOutbnd: Legacy
Subscriptions: Yes
Overlap dial : Yes
DTMFmode : rfc2833
Timer T1 : 500
Timer B : 32000
ToHost :
Addr->IP : 172.16.5.185:62035
Defaddr->IP : (null)
Prim.Transp. : UDP
Allowed.Trsp : UDP
Def. Username: 1928
SIP Options : (none)
Codecs : (gsm|ulaw|alaw)
Codec Order : (alaw:20,gsm:20,ulaw:20)
Auto-Framing : No
Status : OK (3 ms)
Useragent : X-Lite release 4.8.0 stamp 75944
Reg. Contact : sip:1928@172.16.5.185:62035
Qualify Freq : 60000 ms
Keepalive : 0 ms
Sess-Timers : Accept
Sess-Refresh : uas
Sess-Expires : 1800 secs
Min-Sess : 90 secs
RTP Engine : asterisk
Parkinglot :
Use Reason : No
Encryption : No
FreePBX 2.11.0
asterisk 11.20.0
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November 2, 2016, 5:13 am
@dbaddorf wrote:
Does anyone know how to make the Unavailable Message play when someone is leaving a voicemail and the Asterisk DND (*76) is enabled for the extension? It appears that the FreePBX system is playing the Busy message instead. I appreciate anyone's input!
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November 2, 2016, 8:47 am
@dkwiebe wrote:
I administer 3 FreePBX machines all of which run current versions. This morning all of them were experiencing delays between the dial command and the sip phone rining. In each case removing stun.counterpath.com from Settings -> Asterisk SIP Settings was enough to take care of the issue. I first tried adjusting DNS servers but that didn't make a difference. Is this just my machines or are others running into this?
I came across an old thread someplace online that indicated this would be worth trying while troubleshooting delays caused by DNS. I can't find it to give them credit. But thanks to whoever it was!
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November 2, 2016, 11:22 am
@mvogel4949 wrote:
My FreePBX system upgraded to .17 last night and this morning any call that comes into a ring group doesn't ring. If I take the DID and point it at a VM, extension or IVR then things are perfect. Anyone else having RG troubles? I'm also at Asterisk V13.12.1
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November 2, 2016, 12:10 pm
@jsilva wrote:
Hello, good afternoon everyone.
I am by a problem with the freepb 12+ Asterisk 11. The same is not doing the recordings of calls to the database in MySQL, to observer the loaded modules observe the "cdr_mysql.so" is not enabled. What date is the module "cdr_sdlite3_custom.so".
I'm using CentOS 6.8 64Bit, I observe that the systems are using the modules in the following path "/ usr / lib / asterisk / modules" and not on the path "/ usr / lib64 / asterisk / modules". I can fix this or I need to compile the asterisk again?
Attached sending some images.
I thank the attention of everyone.![]()
![]()
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