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Back ups with Multiple Virtual Machines running Freepbx12

August 15, 2019, 3:18 pm
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@FVIChris wrote:

Hey there everyone, I currently have two servers both running Virt-Manager with a couple PBX’s running. One being the primary and the other serving as a backup. Currently I manually replicate the pbx each time I create a new one or make any changes to one on each server. Im trying to ease the process. I tried using the backup module with a SSH server. But im running into problem. Errors saying cant resolve host name and no such directory or file for the SSH key location. I believe the issue is the backup is running inside the VM and cant access SSH key location on the main server outside the VM. Any ideas or other back up methods when running PBX’s on Virtual Machines other then snapshot.

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All calls work... except from verify Google voice?

August 16, 2019, 7:55 am
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@cdsJerryw wrote:

I have Google voice numbers forwarded to my FreePBX. They work fine. However every now and then Google requires me to verify them. When I have Google voice call the number to give me the code, there’s nothing. No sound. If I “Follow me” the call to my cell, same thing. Dead air. Again, only when doing the verify? Anyone else run into this or have a solution?

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Text to speech module

August 16, 2019, 8:32 am
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@AmidouFlorian92 wrote:

Hi
I have upgraded my freepbx to version 15.04 but I didn’t have the Text To speech module on my IVR.Is there a way to download It manually?
Thanks

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Perform GUI actions from the CLI

August 16, 2019, 9:12 am
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@dpittman wrote:

Is there a way to perform GUI actions from the CLI, specifically for use in a script? I simply want to enable custom firewall settings in Connectivity --> Firewall --> Advanced --> Advanced Settings --> Custom Firewall Rules. Or…I see a GET request (“GET /admin/ajax.php?command=updateadvanced&module=firewall&option=rejectpac kets&val=enabled HTTP/1.1” 200 91 “http://1.2.3.4/admin/config.php?display=firewall&page=advanced&tab=settings”) happens when I enable/disable that button - would anyone know how I could recreate that request for use with curl or wget? Any other ideas?

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LTE gateways also capable of being an LTE modem?

August 14, 2019, 8:40 pm
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@stephen wrote:

Does anyone know of any LTE gateways which can do the following with a SIM card?

  1. LTE modem
  2. SMS via API or SMTP/IMAP
  3. SIP trunk via mobile voice service or VoLTE

The devices I’ve looked at seem to do either 1 or 2 & 3, but not all in one device. The Yeastar TG200 seems to be capable of 2 & 3 for instance.

I’m in Australia, so Optus/Telstra/Vodafone compatibility is what I’d be interested in, that will come down to the relevant frequencies of those carriers.

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Constant crashes / server locks up complety

August 16, 2019, 5:59 pm
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@fischb22 wrote:

Hi,

randomly(random hour/random minute) my freepbx server completly locks up(cannot even ping it). i have this on both an old machine, that ran great on an earlier version for at least a year.

thinking i have a hardware failure, i also have a ESXI server, that i was going to migrate to, but on a brand new, latest ISO download last night, it also randomly locks up to where the console doesnt respond, nor can i ping it. (different IP address)

there is nothing telling i can see in the logs

help?

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Freepbx dependence on external services

August 17, 2019, 4:46 am
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@ugintl wrote:

I installed freepbx distro on hyper-v windows 10 successfully. When I visit the dashboard, i see some ads and I have noticed from the lower left of the browser, that it is connecting to cdn.schmoozecom.com.

Is it possible to remove ads and load everything locally so that I do not have to depend on external services? For example if somehow cdn.schmoozecom.com goes down or shutsdown completely, my freepbx project goes completely useless.

I am sure I have conveyed my point

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Asterisk CDR Database

August 4, 2019, 1:28 am
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@like2143 wrote:

By default, asterisk will generate a CDR record in database for every call when hangup,
Is it possible to configure asterisk which generate CDR record in database when the call is answered by extension.
Aterisk version is 16.4(VitalPBX for GUI)

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BLF to extension that was deleted

August 18, 2019, 9:38 am
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@rsarceno wrote:

When an extension is deleted but the phones are still configured for BLF for the deleted extension, I start getting a notice on the log “does not exist in context ‘from-internal’ or has no associated hint”

[2019-08-18 09:06:34] NOTICE[29065] res_pjsip_exten_state.c: Endpoint ‘363’ state subscription failed: Extension ‘369’ does not exist in context ‘from-internal’ or has no associated hint
[2019-08-18 09:06:52] NOTICE[29065] res_pjsip_exten_state.c: Endpoint ‘421’ state subscription failed: Extension ‘423’ does not exist in context ‘from-internal’ or has no associated hint

I wish I was using Endpoint Manager but instead I have to go to each site to remove BLF for each phone. For the meantime, my log is getting full and using disk space, Is there a way to suppress this message.
.
Thanks

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Asterisk CDR Database

August 4, 2019, 1:28 am
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@like2143 wrote:

By default, asterisk will generate a CDR record in database for every call when hangup,
Is it possible to configure asterisk which generate CDR record in database when the call is answered by extension.
Aterisk version is 16.4(VitalPBX for GUI)

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VM notifications not sending. Email testing works fine

August 19, 2019, 11:50 am
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@jhcomputergroup wrote:

We have 5-6 systems running and not had this issue before.

If i go to the sysadmin module and do email setup. In the debug mode i can send test messages to myself all day long no problems.

in the console same thing using the send mail through console has no issues.

But we get zero email notification for voicemail. In the extension we have email address added. In the user we login the UCP and see the voicemails listed. But we never get an email stating we have a voicemail. i have tried with email attachment on and off to no avail.

mailq in console is always empty.

Email log in the debug section of the email setup never mentions sending an email for voicemail but will show when doing any tests.

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Fwconsole unlock seems to break my existing admin account

August 19, 2019, 1:49 pm
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@bksales wrote:

I can get into the GUI fine when its unlocked and I can create a new admin account but no matter what I set the password to for the existing one I cant log in with that account anymore. Is there another command that needs to be run after unlock?

12.7.6-1904-1.sng7

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Extension is conflicting , incomming calls are dropped out

August 19, 2019, 2:47 pm
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@lcn wrote:

Hello,

I’m using Free PBX in version 15 with Raspberry.
I recently get an error reported from the Dashboard :

There are 1 conflicting extensions
Extension: 77:
INUSE: ParkPlus: PickupSlot Default Lot
INUSE: Ring Group: Ring Group

So now when i receive an incomming call, Freepbx ends the calls automatically.

Here is a log when an incomming call happens

|Mon, Aug 19, 2019 4:05 PM |HANGUP |1566223553.19 |1566223530.18 |77 |77 |from-internal |Local/00xxxxxxxxx@from-internal-00000005;1|
|Mon, Aug 19, 2019 4:05 PM |CHAN_END |1566223553.19 |1566223530.18 |77 |77 |from-internal |Local/00xxxxxxxxx@from-internal-00000005;1|
|Mon, Aug 19, 2019 4:05 PM |HANGUP |1566223553.20 |1566223530.18 |098829xxxx |h |from-internal |Local/00xxxxxxxxx@from-internal-00000005;2|
|Mon, Aug 19, 2019 4:05 PM |CHAN_END |1566223553.20 |1566223530.18 |098829xxxx |h |from-internal |Local/00xxxxxxxxx@from-internal-00000005;2|
|Mon, Aug 19, 2019 4:05 PM |HANGUP |1566223530.18 |1566223530.18 |098829xxxx |hangup |app-blackhole |PJSIP/OVH_pjsip-00000008|
|Mon, Aug 19, 2019 4:05 PM |CHAN_END |1566223530.18 |1566223530.18 |098829xxxx |hangup |app-blackhole |PJSIP/OVH_pjsip-00000008|
|Mon, Aug 19, 2019 4:05 PM |LINKEDID_END |1566223530.18 |1566223530.18 |098829xxxx |hangup |app-blackhole |PJSIP/OVH_pjsip-00000008|
|Mon, Aug 19, 2019 4:05 PM |CHAN_START |1566223553.19 |1566223530.18 ||00xxxxxxxxx |from-internal |Local/00xxxxxxxxx@from-internal-00000005;1|
|Mon, Aug 19, 2019 4:05 PM |CHAN_START |1566223553.20 |1566223530.18 ||00xxxxxxxxx |from-internal |Local/00xxxxxxxxx@from-internal-00000005;2|
|Mon, Aug 19, 2019 4:05 PM |CHAN_START |1566223530.18 |1566223530.18 |098829xxxx |s |from-trunk |PJSIP/OVH_pjsip-00000008|
|Mon, Aug 19, 2019 4:05 PM |ANSWER |1566223530.18 |1566223530.18 |098829xxxx |77 |ext-group |PJSIP/OVH_pjsip-00000008|

Now the call is no longer forwarded to GSM usb modem , the calls hangs up.

Can you please tell me what is wrong with extension 77 ?
How could i fix that please ?

All was working perfectly during several monthes.

Many thanks for your help,

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/var/lib/mysql/asteriskcdrdb/cel.ibd is huge [ solved ]

August 5, 2019, 1:59 pm
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@robfantini wrote:

Hello - after receiving a report about low free storage space we noticed that

/var/lib/mysql/asteriskcdrdb uses 6.3G .

cel.ibd uses 6.2G

Is there a way to limit the space used , or prune the file?

Is it normal for the file to be so big?

PS: In freepbx today one of our users tried to use Call Event Logging . that screen got stuck or was just taking forever to display. so perpahs just running that report populate cel.ibd .
I tried to disable the cel module, however we use it in timeconditions…

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FreePBX and an Audiocodes SBC

August 19, 2019, 4:57 pm
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@Digitalmocking wrote:

Has anyone set anything like this up? I found some old instructions based on a much older version of an SBC using Asterisk@Home, but so much has changed.

I’ve got an SBC running Direct Routing with Microsoft Teams. I’m looking for a phone system I can connect various SIP endpoints to (common area phones, SIP testing etc) and just have it all route through our SBC.

I haven’t touched Asterisk in a long long time, and so far my google searching hasn’t been awesome.

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Domain account to login softphone

August 20, 2019, 12:37 am
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@JackyCHEN wrote:

Hi everyone, I would like to know is it possible use a domain account to login softphone such as X-Lite, Jitsi…etc with FreePBX?
I want reduce account number of user to remember and I don’t need setup password on FreePBX for each sip account (Extension).
I have integrated MS AD in FreePBX already, and user can use domain account to login FreePBX GUI.
In the meantime, I setup the phone number on phone fields in domain account, Option “Create Missing Extensions” was select “PJSIP” in FreePBX, when I run sync, the user account and sip account will bind together.
But I cannot use domain account to login softphone, it just accept Extension number to login.
Have I missing in configuration or need some plugs/applications to do it?

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Check call logs

August 5, 2019, 4:40 pm
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@Alex23 wrote:

Question for you guys, I would like to check some call logs and be able to track if my calls are dropping or if the other side is hanging up is there any log where I can check that?
Thank you for your help.

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/var/lib/mysql/asteriskcdrdb/cel.ibd is huge [ solved ]

August 5, 2019, 1:59 pm
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@robfantini wrote:

Hello - after receiving a report about low free storage space we noticed that

/var/lib/mysql/asteriskcdrdb uses 6.3G .

cel.ibd uses 6.2G

Is there a way to limit the space used , or prune the file?

Is it normal for the file to be so big?

PS: In freepbx today one of our users tried to use Call Event Logging . that screen got stuck or was just taking forever to display. so perpahs just running that report populate cel.ibd .
I tried to disable the cel module, however we use it in timeconditions…

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All the CRON

August 20, 2019, 6:58 am
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@captain118 wrote:

I’m looking at our logs and man does freepbx fill up a log. Mostly with useless crap (By useless crap I mean cron jobs running). After looking at it there are +25 (in my system) cron jobs for the freepbx server. I was surprised that the system needed to use cron jobs to run tasks. While cron is a nice system I was surprised to find these functions were not internal to the application itself. Is there a plan to reduce the number of required cron jobs? Honestly I dont mind the ones that run daily or weekly but I have at least 5 that run every min. Those start to cause a log to cycle pretty quick.

Thanks,
Kirk

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Saying a date

August 20, 2019, 10:38 pm
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@comtech wrote:

FreePBX/Asterisk 14

In custom dial plan I am collecting three variables from the caller.

${MONTH}
${DAY}
${YEAR}

An example a caller could input is:
${MONTH}=01
${DAY}=05
${YEAR}=2019

What command could I use to string the variables together and speak back the date in a format like:
January 5th, 2019

Not sure if this is possible, but thought I would check.

Thanks in advance for any help or guidance here.

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