@ThetaCoder wrote:

After a long (6 year) hiatus, I have once again been tasked with setting up a FreePBX based phone system at work. As I have been away for so long, I wanted to ask the community to look over my initial idea and let me know if it’s do-able, or if I’m drinking the funny kool-aid.

The SIP provider we are talking to installs a dedicated box with a network port.
The FreePBX box will have 2 NICs, one for the general internal LAN (with internet access).
ALL Phones will be connected to a dedicated switch, this switch will be connected to the SIP box and the 2nd NIC on the Free PBX.

Is this a decent setup, do I need to have a DHCP server somewhere on the isolated network, or do I need to re-think my setup.

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@able wrote:

the following kills the server every 15 minutes or so, killing all reloading gets it running again. this is the only thing i can find in the logs:

May 23 15:34:02 freepbx pnp_server: Starting Sangoma pnp_server
May 23 15:34:02 freepbx pnp_server: Traceback (most recent call last):
May 23 15:34:02 freepbx pnp_server: File “/usr/local/bin/pnp_server”, line 242, in
May 23 15:34:02 freepbx pnp_server: sysadmin = get_sysadmin_settings()
May 23 15:34:02 freepbx pnp_server: File “/usr/local/bin/pnp_server”, line 98, in get_sysadmin_settings
May 23 15:34:02 freepbx pnp_server: dbh = get_db_handle()
May 23 15:34:02 freepbx pnp_server: File “/usr/local/bin/pnp_server”, line 57, in get_db_handle
May 23 15:34:02 freepbx pnp_server: db = MySQLdb.connect(host=fpbx[‘AMPDBHOST’], user=fpbx[‘AMPDBUSER’], passwd=fpbx[‘AMPDBPASS’], db=fpbx[‘AMPDBNAME’])
May 23 15:34:02 freepbx pnp_server: KeyError: ‘AMPDBHOST’
May 23 15:34:02 freepbx systemd: sangoma-pnpd.service: main process exited, code=exited, status=1/FAILURE
May 23 15:34:02 freepbx systemd: Unit sangoma-pnpd.service entered failed state.
May 23 15:34:02 freepbx systemd: sangoma-pnpd.service failed.
May 23 15:34:02 freepbx systemd: sangoma-pnpd.service holdoff time over, scheduling restart.
May 23 15:34:02 freepbx systemd: Stopped Sangoma PnP Service.
May 23 15:34:02 freepbx systemd: Started Sangoma PnP Service.

I’ve looked all over but the only thing i can find is someone who altered that manually, that is great until the first reload

/etc/asterisk/extensions_additional.conf:AMPDBENGINE = mysql
/etc/asterisk/extensions_additional.conf:AMPDBHOST = localhost
/etc/asterisk/extensions_additional.conf:AMPDBNAME = asterisk
/etc/asterisk/extensions_additional.conf:AMPDBUSER = freepbxuser
/etc/asterisk/extensions_additional.conf:AMPDBPASS = (notshown)

Any ideas that may be helpful ?

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@mvogel4949 wrote:

I’m running on Asterisk 13.22 with FreePBXV14 and my memory usage is through the roof. So much so I think the system actually froze up this morning. The system only has 10 physical extensions and 25 Zulu users. Running htop from the command line I see asterisk taking up almost all of the memory

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@mvogel4949 wrote:

What would the command look like to upgrade to a specific version of framework? Say I want to go to 14.0.5.2 because that is where my backup version is and I’m at 14.0.3.1

I’m thinking something like

fwconsole ma downloadinstall framework --14.0.5.2 but that’s not quite right

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@cdsJerryw wrote:

I have critical errors showing on my Dashboard saying there are 14 bad destinations then listing them with the top one being Dest Status: Orphan. Under that are several that say "Follow-Me:xxx (extension number). Further down it shows those same extensions listed as "Exten: name of ext (xxx)

I’ve looked at the follow me settings for each of these users and I don’t see anything out of place. I’ve tried turning off the enable follow me to see if that had any effect, but it does not. I’ve searched for Queues and Ringgroups that have all these extensions, but found none.

How do I find where these destinations are being routed from?

Not sure if it’s important, but I had an inbound route that suddenly didn’t work too. The call would ring to the right place and the other party could hear us, but we could not hear them. Tried all sorts of things but could not resolve it. Deleted the route and entered it again with the same settings and it worked. Both the route issue and these bad destinations started at the same time which is why I wonder if they are related.

It seems like I should resolve this before I do the upgrade to the new version of FreeBPX.

I’m on PBX 14.0.11 with Asterisk 13.22.0

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@herbw wrote:

I have configured a few ring groups. Is it possible to configure a different ring tone for each ring group?

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@moussa854 wrote:

I have a script to check if an extension is up and it run on cron job every 30 minutes. Any thoughts on how to make the script send an email once and not every 30 minutes while the extension is down?

#!/bin/bash
email="myemail@yahoo.com"
A=`/usr/sbin/asterisk -x 'sip show peer 200' | grep -i status | cut -d' ' -f11`
if [ "$A" != "OK" ]; then
(
echo "Subject: Extension 200 is down"
printf 'Extension 200 is down"...'
) | /usr/sbin/sendmail ${email}
fi

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@jburk wrote:

Hi All, I have FreePBX running on a PBXact Appliance with a Vega 60G 4FXO Gateway

I configured the trunk between the 2 following this guide:
https://wiki.freepbx.org/pages/viewpage.action?pageId=60915768
other than CHAN_SIP is using 5160 on my FreePBX so I used that port for SIP Registrar on my Vega.

I have a one-way audio issues on outbound calls from FreePBX to the PSTN - the FreePBX side can hear the remote caller, but the remote caller cannot hear the PBX side.

Inbound Calls are not affected at all.

There is no firewall between PBXact and Vega, they are both on the same VLAN along with the phones and all can ping each other just fine. Phones are Sangoma S500

I verified that the subnet is whitelisted in the FreePBX firewall and whitelisted in the Intrusion Prevention as well.

I can’t find a network issue and I’m ready to pull my hair out - I only have uLaw allowed on the trunk and it looks like that is what is always used for all calls. My first thought was codec mismatch…

What else can I be missing?

SIP reinvite is not allowed on Chan_SIP also…

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@herbw wrote:

It appears as if it is possible to change the ringtone for Sangoma phones using the Endpoint Manger (https://wiki.freepbx.org/display/PHON/Changing+Ring+Tones).

Is there any way to perform the equivalent function for Snom 821 phones using either the Endpoint Manager or the Extensions Manager? After provisioning the phone, I can change the ringtone using the Ringer Preferences menu on the phone, but I am unable to find any way to do this as part of the provisioning process within freepbx.

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@diegovillalba wrote:

guys, how i can restart asterisk?

i cant create extension in the GUI.

i use asterisk -vvvc, but when i close the shell it does not work again.

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@herbw wrote:

All of my internal extensions are 2XX or 4XX.

No valid external numbers begin with either 2XX or 4XX, since callers must dial 1 for long distance calls, and no local area codes begin with 2XX or 4XX.

Is there any way to force 2XX and 4XX extension numbers to be dialed immediately (without pressing the call button)?

It appears as if the outbound dial rules only apply to trunks, which are not used when dialing local extensions.

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@Chrismur91 wrote:

Today we upgraded our FreePBX version from 14 to 15.0.16.

The upgrade went smoothly, except once it was done, the Backup & Restore module needed to be updated. Trying to update it to the latest version threw errors, so I tried uninstalling it and installing it fresh, but I continue to see the same behaviour when I was trying to update the module.

When I try to update it in the web interface, I receive this screen briefly:

which then turns to this screen after a few seconds:

The error that displays in that window reads: “Cron line added didn’t remain in crontab on final check. Check /tmp/cron.error for reason.” However when I check /tmp/cron.error, the file is empty, so it’s of no help.

I’ve tried checking crontabs for various users that might be causing an issue, but I can’t find anything that might cause an issue as far as I can tell. Any advice or help here would be appreciated so I can get the module installed

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@dsivoip wrote:

Solved this one! Yay!! (posted in Spiceworks, copied here)

Placed the following script into the file “extensions_custom.conf”

then click Save and Apply Config.

Note: Our queue extensions are 600, 601, and 602, adjust these extensions in this script accordingly for your system.

Copy and paste this into that file:

[ext-queues-custom]
; Begin Queues Hints
*exten => 47600,1,Gosub(app-queue-caller-count,s,1(600))
*exten => 47600,n,Hangup
*exten => 47600,hint,Queue:600
; End of Queue 600 Hint
*exten => 47601,1,Gosub(app-queue-caller-count,s,1(601))
*exten => 47601,n,Hangup
*exten => 47601,hint,Queue:601
; End of Queue 601 Hint
*exten => 47602,1,Gosub(app-queue-caller-count,s,1(602))
*exten => 47602,n,Hangup
*exten => 47602,hint,Queue:602
; End of Queue 602 Hint

The file can be edited from within PBXact/FreeBPX by going to Modules/Config Edit, then selecting “extensions_custom.conf” from the list on the left in the tree under the heading Asterisk Custom Configuration Files. Paste it into the field on the right once the file is selected.

The file is also found via CLI at /etc/asterisk/extensions_custom.conf and can be edited with Nano or VI.

Then, moving on to the Grandstream GXP2160 and 2170 phones, go to Settings/Programmable Keys/Virtual MultiPurpose Keys, then press Edit VPK,

• Mode: BLF

• Account: Account 1 (Probably doesn’t matter for this)

• Description: Something that makes the most sense for your users as it will be displayed on a screen

• Value: *47xxx (put your queue’s extension number in for the xxx)

Now, when there is a call in the queue this button will flash red. If an Agent is Paused or has Logged Out the flashing will remind them there is someone waiting. Also, it will flash when they are on a call when another is waiting in the queue.

This should work with any other phone that offers programmable BLF as an option.

Haopy Trails!

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@msobik wrote:

I wanted to follow up to a previous post I made requesting help setting up an Edgemarc (EM) 2900e PoE:

https://community.freepbx.org/t/edgemarc-and-freepbx/56276

Link to the device:
https://ribboncommunications.com/products/service-provider-products/session-border-controllers/edgemarc-2900-series

I finally have FreePBX, Twilio, and the EM working together. In order to make this happen I did have to call Ribbon for support. The EM was having some issues working with Twilio. Apparently, Twilio does not support SRV lookups and that was causing some issues with the EM. Arthur at Ribbon was very helpful in diagnosing the issue and getting everything up and running.

I have a 50/50 fiber connection going into a ISP owned router. The EM sits behind the ISP router and I have a Cisco SF300-24PP-K9-NA switch behind that along with a Ubiquiti wireless AP. I’m running Sangoma phones. FreePBX is hosted on a Virtual Box VM on an Ubuntu physical server on the LAN.

Couple of notes with the EM. You can’t really turn off the ALG. You can, but you also have to turn off the firewall in order for it to function properly. I didn’t realize this when I purchased it. As it happens, I kind of like the ALG functionality. I don’t have to put FreePBX in a DMZ and I also don’t have to port forward. The ALG also provides Qos and traffic shaping. The ALG seems to be working fine with FreePBX. Also, the EM has a lot of features and options, but the documentation is a bit lacking. The user guide documents all the options, but doesn’t do a good job explaining what those options do necessarily. Even though I am an end user I don’t think Ribbon intends for this particular device to be sold to directly to end users, rather through a partner or service provider. That being said, I did manage to get everything setup properly.

I used the following guides to get everything up and running:

Twilio FreePBX configuration:
https://www.twilio.com/docs/documents/53/TwilioElasticSIPTrunking-FreePBX-Configuration-Guide-Version1-0-FINAL-06122018.pdf

EdgeMarc VOS User Guide (search the support portal for the latest version):
ftp://ftp.edgewaternetworks.com/pub/docs/KB/VOSEdgeMarc_User_Guide_14.8.0_GA.pdf

EdgeMarc IP-PBX Configuration Doc (on the support portal):
“How to configure an EdgeMarc for SIP trunking with an IP-PBX.”

Follow the Twilio config doc as is, except in FreePBX→Trunks→pjsip Settings→Sip Server. This should be set to the LAN IP of the EM instead of the Twilio SIP Server FQDN. This is because the EM proxies traffic through the ALG. If don’t, you’ll see messages like this from Asterisk SIP debug since the invites are now coming from the EM (192.168.10.1):

[2019-05-21 13:20:24] NOTICE[9403]: res_pjsip/pjsip_distributor.c:649 log_failed_request: Request ‘INVITE’ from ‘“XXXXXXXX” sip:+XXXXXXXXXX@192.168.10.1’ failed for ‘192.168.10.1:5060’ (callid: d83ba2a66036182f3958c8cf171ed463@0.0.0.0) - No matching endpoint found

Follow the EdgeMarc IP-PBX config doc for “Scenario A” to setup the ALG. “Allowed SIP Servers” are where you whitelist the SIP signaling servers. As it was explained to me by Ribbon, you don’t need to whitelist SIP media servers because the EM dynamically puts holes in the firewall during SIP setup. Twilio has an EM guide that uses B2BUA on the EM, but using the pass through ALG is simpler and cleaner as you don’t need to manipulate the SIP headers.

Finally, on the EM under VoIP→Survivability set “Time (s) between DNS lookups:” to 3600 and make sure “Enable SIP server redundancy:” and “Enable SRV Lookup:” are both unchecked. That solves the issue with Twilio not supporting SRV lookups and lengthens the time between DNS A Name lookups.

The following command were useful:

On FreePBX:

asterisk -vvvvr

sip set debug on/off

On the EM CLI:

To turn on SIP tracing

mandctl log 0x19

To turn off SIP tracing

mandctl log 0

“mand” is the ALG process

System messages are written out to /var/log/messages as the EM is linux based.

I can no longer find the KB article that details how to take a tcpdump from the EM, but the KB article# was 96589. I think support is working to migrate these to their new support portal.

Please don’t hesitate to contact me with any questions or comments.

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@dan_ce wrote:

Hi!

I have a trunk setup that I can dial out on, but I’m trying to setup an inbound route, and coming unstuck. I am trying to tell FreePBX that my DID is 012133322NDND (my phone number on this trunk is 012133322) but of course FreePBX is saying my DID has to be numerical only.

Can I somehow send all calls from this trunk to a certain flow without needing to use a DID?

Any ideas?!

Thanks!!

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@Alex23 wrote:

I have an analogic line that I connect to my gateway is a Grandstream GXW4104 I’m able to do the connection with PBX and able to dial out without problems, the problem is when I call to that phone number the calls never go thru. I configure my inbound route and looks okay to me.
I open the port 5060 on my modem but still not able to receive the calls how I can ensure that my configuration is the right one.
PD: I configure a Trunk to do the connection with the analogic line. This are my outgoing settings:
type=friend
secret=****
qualify=yes
port=5060
host=gatewayip
dtmfmode=rfc2833
disallow=all
context=from-trunk
canreinvite=no
allow=ulaw
Incoming are blank.(I tried to add a configuration there but nothing change it.)

Thanks in advance

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@kr0490 wrote:

See my previous thread here for history

Inbound calls work, but outbound calls fail with “all circuits are busy now” and in the log i see:

[2019-05-25 18:20:49] VERBOSE[25542][C-0003ccb4] app_dial.c: Called PJSIP/16144044940@FlowrouteTRUNK
[2019-05-25 18:20:49] VERBOSE[25542][C-0003ccb4] app_dial.c: Everyone is busy/congested at this time (1:0/0/1)
[2019-05-25 18:20:49] VERBOSE[25542][C-0003ccb4] pbx.c: Executing [s@macro-dialout-trunk:33] NoOp(“PJSIP/100-0003cce2”, “Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 21”) in new stack
[2019-05-25 18:20:49] VERBOSE[25542][C-0003ccb4] pbx.c: Executing [s@macro-dialout-trunk:34] GotoIf(“PJSIP/100-0003cce2”, “0?continue,1:s-CHANUNAVAIL,1”) in new stack
[2019-05-25 18:20:49] VERBOSE[25542][C-0003ccb4] pbx_builtins.c: Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)

Here is the full log: https://drive.google.com/open?id=1pMlceC3PbEdowJN56vJXjFW2N5rd9jKi

Any ideas, my trunk configuration is in the previously mentioned thread.

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@epp289 wrote:

Hello,
i am running the freepbx installed from the distro cd. It is the SHMZ release 6.5 (Final)
Yesterday, i updated the freepbx from version 12 to 13 following the link from the gui menu.

it seems that it didn’t do it successfully. I cannot access the recording’s page anymore (but the CDR reports is working fine though).
Please see the attached screenshots. One fault i see is that the process management module cannot be upgraded because the “node package manager” is not installed. I tried several tutorials on the web on how to install the latest version of nodejs in centos without success.
And my question is if i can easily fix the above nodejs problem in order to install the process management module
and another question is if i download the latest distro with freepbx if i will lose data because my centos machine is doing other tasks simultaneously (streaming, hosting, running cron tasks etc). Thank you

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@windswept321 wrote:

I’m trying to set up a dial plan for local numbers in the UK on Twilio.

The trunk is set to have an outbound dial prefix of + (requirement of Twilio).

A number would usually be something like area code (5 digits entered, leading 0 stripped) plus a 6 digit phone number.

Country code of 44 is set as prepend and 0 as prefix in the outbound routes settings (requirement for Twilio to work, again).

Following those, I want to be able to enter a number like 0[012378]XXXXXXXXX and have the rule match.

That doesn’t work - I get “…please check the number and dial again…”
Setting it as X[012378]XX. does work. However, 0[012378]XX.doesn’t.
Replacing the leading 0 with X in the first example also doesn’t work.

Can anyone help me with this issue?

Thanks for reading

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@slip_cougan wrote:

I’ve been running FreePBX on a Pi 2 for a few months now. I spent time figuring it out and have to say this is a great solution for getting to know FreePBX. I had some great help with one or two minor issues from this community and it is greatly appreciated.

Now I want to take things up a notch as although the Pi is great for projects and fiddling around, sooner or later the SD card will fail and I’ll have to start from scratch again. I’ve cloned the SD in the event that it all goes a bit Pete Tong, but want to put my FreePBX on some proper HW now.

More than likely I’ll go with an i3 NUC or something similar - I want to keep this as low power as possible.

So - how can I take the config from my Pi and place it on the NUC?
What files/folders can I copy to the NUC?

This is assuming of course that I have already downloaded and installed FreePBX on a newly formatted PC.

Many thanks
-s

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