@VoIP2Go wrote:

I have a project with hundreds of sites around the US each with an SPA112 VoIP device.

All devices are behind the same firewall and share the same public IP address.

How can I setup FreePBX 14.0.1.24 (running on a VPS) to allow calls from any device through a single extension.

As a side note, all devices only ever dial a single toll free phone number and so it’s not a problem to lower security by disabling the firewall.

There is only a single outbound trunk to a toll free VoIP provider.

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@radioring wrote:

Happy New Year all…!

This is most likely a simple thing to get sorted…

I have two FreePBX servers - one at my office and one at home. Both have SIP trunks to the outside world and can make and receive calls through their respective extensions.

Reciently, I have linked both systems through an IAX trunk. I am doing more work from how now and want the ability to have the office number available at home.

The systems work well together. When a call comes in via the office SIP trunk, phones ring at the office and in my home office via my home pbx. I can also call the office extensions from home and vice-versa. All good.

However, I have yet to sort out routing a call from my home PBX out through the SIP Trunk on the office PBX. Whatever I seem to do, all that happens is that the home PBX ends up calling the office PBX just as if a call has come in from the outside world . I cannot seem to get a call to come in via the IAX trunk and to route straight out the SIP trunk on the office PBX (I have a “welcome to company xxx” that is played to the caller prior to extensions ringing).

Can anyone give me pointers as to what I am not doing right?

Many thanks,

Dan

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@MacsOffice wrote:

Forgive my ignorance - but do I have to set up users and groups in a simple phone system? The phones have extension numbers - and I want VM to EMail to work, but the users of this 15 phone system will never need to enter the Free PBX GUI , Only I will. So I guess the question becomes - where does the users and groups information come into play? When will someone who is using this as a phone, need to logon to something?

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@me3ikan wrote:

Hello, I recently reinstalled freepbx 14 on my system. I had some trouble with disk partitioning so I ended up selecting “Fully automated installation” (or something similar, not sure). Freepbx was successfully installed, I didn’t see the installation wizard gui at all and I wasn’t asked for any password. Freepbx works without problems but I can’t login to the system as root. I googled everything and tried many passwords but nothing worked. Any ideas?

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@shawnsbrain66 wrote:

When a call comes in, ALL the lights ring on my phone. It’s like I’m winning the jackpot at the casino. This happens on multiple phones (all in the ring group).
How do I fix that?

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@comtech wrote:

FreePBX/Asterisk 13

I’m noticing Generate Device Hints is missing from the GUI. I am having an issue were *45 is not log me in and out of all queues that are assigned to me dynamically. A forum post said that Generate Device Hints needs to be enabled on all queues to fix this. Any ideas on how to resolve this?

Thanks!

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@PCS wrote:

I’m sure it’s something simple, but can someone provide information on provisioning multiple handsets from EPM?

I can get one handset working, but I can’t figure out how to add a second with a different extension.

Thanks!

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@Napol1 wrote:

Hello in my FreePBX when i try to install it showed the problems and i can not using to call also. Anyone can you me?
!

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@herotic wrote:

Hello!
I have installed FreePBX on Centos 7. But i cant see the Admin setup page.
For instalation i used that help Installing+FreePBX+14+on+CentOS+7 from this site.
But i see white screen with errors.

Warning: include_once(/etc/freepbx.conf): failed to open stream: Permission denied in /var/www/html/admin/config.php on line 100

Warning: include_once(): Failed opening ‘/etc/freepbx.conf’ for inclusion (include_path=’.:/usr/share/pear:/usr/share/php’) in /var/www/html/admin/config.php on line 100

Fatal error: Class ‘FreePBX’ not found in /var/www/html/admin/config.php on line 110

Warning: Unknown: open(/var/lib/php/session/sess_ce5490l2r3shu42p4it4qqtdl1, O_RDWR) failed: Permission denied (13) in Unknown on line 0

Warning: Unknown: Failed to write session data (files). Please verify that the current setting of session.save_path is correct (/var/lib/php/session) in Unknown on line 0

Set permission to folder /var/lib/php/session/ not help.
I have tried “fwconsole chown” - the same result.
Please, help!

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@thembimoyo wrote:

Good Day… I need help. I am trying to add a PJSIP Trunk but the submit button won’t go. In other words, after I fill in all the trunk details and I click on Submit the button does not respond.

I have tried 3 diffrenent browsers no joy…

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@comtech wrote:

FreePBX/Asterisk 14 SNG7

I have a server with 32GB of memory and every time I reset it I see most of the memory (~80%) free up. From the point of the reset over the next couple of days the memory usage will eventually max out and the swap will move to 1. Then it will just stay like that. No call complaints.

It doesn’t matter how many calls are processing, it never goes back down until I reboot. Is this expected behavior? Do we have a problem?

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@happyyasu09 wrote:

So I found out how to do that using code, but really would be awesome to understand how to do that via GUI.
So far, the question is how to place any audio file to caller (any external caller came over trunk), after he finished with IVR and on call-answer.

Basically a little scheme:

Caller (Mr.White) ----Calling—>(Company Phone Number)----->Greeting Message—>IVR----->on “press 1”—round-robin(12 extensions total)---->Extension101 Picked Up the Call------>Play audio to caller (smth like “Greetings. Mr.Green is going to answer your call, please standby”)---->Conversation---->Extensions101 Hangup the Call----->Play audio to caller (smth like “Thank you for choosing us”)---->Finish.

So if Extensions102 Picked Up the Call the audio file would be different, and so on, for every extension we need different file.

So, basically i can write the script to do that, but I need something easier, I think i may have missed some parameter somewhere.
So any ideas would be welcomed!
Thanks in advance!

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@asyscom wrote:

Hello to all,
I have a problem that blocking the project of putting into operation a new pbx for a company.
My configuration is freepbx FreePBX 14.0.1.20 + patton SN4171

The situation is simple, incoming calls works regularl, direct calls via pass-through and IVR, the problem is the outgoing calls, from the dialplan the instructions are routed correctly only that the error is the following, the classic 503

Error in Diaplan
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
– Called SIP/gatewaypri1/01875295100
– Got SIP response 503 “Service Unavailable” back from 192.168.10.132:5060
– SIP/gatewaypri1-0000009b is circuit-busy

This my configuration

Patton:
#----------------------------------------------------------------

Patton Electronics Company

Wizard generated config file

Name:

Trinity SN4170 / SN4970 / SN4971 Basic Setup

Description:

This sets up your Trinity SN4170 or SN4970/71 with either an IPPBX or an ITSP SIP Trunk.

#----------------------------------------------------------------
cli version 4.0

clock local default-offset +01:00

profile aaa DEFAULT
method 1 local rule required
method 2 none rule required

console
use profile aaa DEFAULT

telnet-server
use profile aaa DEFAULT
no shutdown

ssh-server
use profile aaa DEFAULT
no shutdown

snmp-server
shutdown

web-server http
use profile aaa DEFAULT
no shutdown

system
clock-source 1 e1t1 0 0

ntp
server 0.patton.pool.ntp.org
server 1.patton.pool.ntp.org
server 2.patton.pool.ntp.org
server 3.patton.pool.ntp.org
no shutdown

dns-client
name-server 192.168.10.10

profile tls DEFAULT
no authentication incoming
no authentication outgoing
private-key pki:private-key/DEFAULT
own-certificate 1 pki:own-certificate/DEFAULT

profile call-progress-tone IT_Dialtone
play 200 425 -12
pause 200
play 600 425 -12
pause 1000

profile call-progress-tone IT_Alertingtone
play 1000 425 -12
pause 4000

profile call-progress-tone IT_Busytone
play 500 425 -12
pause 500

profile tone-set default

profile tone-set IT
map call-progress-tone dial-tone IT_Dialtone
map call-progress-tone ringback-tone IT_Alertingtone
map call-progress-tone busy-tone IT_Busytone
map call-progress-tone release-tone IT_Busytone
map call-progress-tone congestion-tone IT_Busytone

profile voip DEFAULT
codec 1 g711alaw64k rx-length 20 tx-length 20
codec 2 g711ulaw64k rx-length 20 tx-length 20
fax transmission 1 relay t38-udp
fax transmission 2 bypass g711alaw64k rx-length 20 tx-length 20
fax transmission 3 bypass g711ulaw64k rx-length 20 tx-length 20
fax bypass-method signaling
modem transmission 1 bypass g711alaw64k rx-length 20 tx-length 20
modem transmission 2 bypass g711ulaw64k rx-length 20 tx-length 20
modem bypass-method signaling

profile pstn DEFAULT

profile sip DEFAULT

context ip ROUTER

interface WAN
ipaddress WAN 192.168.10.132 255.255.255.0

routing-table DEFAULT
route 0.0.0.0/0 gateway 192.168.10.85 metric 0

profile ppp DEFAULT

context bridge

context cs SWITCH
no shutdown

routing-table called-e164 RT_ISDN_TO_SIP
route T2 dest-interface IF_SIP

interface isdn IF_ISDN_00
route call dest-table RT_ISDN_TO_SIP
call-reroute emit
diversion emit
user-side-ringback-tone
caller-name

interface sip IF_SIP
bind context sip-gateway GW_SIP
route call dest-interface IF_ISDN_00
remote 192.168.10.130
trust remote

authentication-service AUTH_SRV
username gatewaypri1 password tEtEnewairlTY7j/OmBBeg== encrypted

location-service SER_LOC
domain 1 192.168.10.130
match-any-domain

identity-group DEFAULT

authentication inbound
authenticate 1 authentication-service AUTH_SRV username gatewaypri1

registration inbound

identity gatewaypri1 inherits DEFAULT

context sip-gateway GW_SIP
bind location-service SER_LOC

interface SIP
transport-protocol udp+tcp 5060
no transport-protocol tls

bind ipaddress ROUTER WAN WAN

context sip-gateway GW_SIP
no shutdown

port ethernet 0 0
bind interface ROUTER WAN
no shutdown

port e1t1 0 0
port-type e1
clock auto
framing crc
encapsulation q921

q921
permanent-layer2
uni-side auto
encapsulation q931

q931
protocol dss1
uni-side user
encapsulation cc-isdn
bind interface SWITCH IF_ISDN_00

port e1t1 0 0
no shutdown

Sip User:
[gatewaypri1]
disallow=all
context=from-trunk
type=friend
insecure=port,invite
qualify=yes
sendrpid=yes
trustrpid=yes
dtmfmode=rfc2833
username=gatewaypri1
secret=gatewaypri1
host=192.168.10.132
nat=force_rport,comedia
allow=alaw
allow=ulaw

Trunksip:
context=from-trunk
type=friend
insecure=port,invite
qualify=yes
sendrpid=yes
trustrpid=yes
dtmfmode=rfc2833
username=gatewaypri1
secret=gatewaypri1
host=192.168.10.132
nat=yes
disallow=all
allow=alaw&ulaw

In the final analysis I debugged on patton but I do not see any outgoing calls so the error is not given by carrier but just by patton.

Thanks in advance

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@seancmalloy wrote:

I have both php 7.0 and 5.6 installed on Debian 9.3. Apache 2.4.25 virtual hosts are directed to specified php versions with php-fpm/fastcgi. I have verified that the freepbx 14 host IS going to php 5.6 (via fpm) and that it connects to mariadb 10 and asterisk 13, but I am still receiving the php 7 compatibly errors (whoops…) when I run amportal. Is there a way to force freepbx (amportal) to use specified php version? Perhaps hard code it?

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@avayax wrote:

What happens when I am applying distro updates via sysadmin and during the upgrade process I lose connection to the server because I am accessing it remotely and my internet connection drops, etc?

Will the upgrade still continue fine or be disrupted?

10.13.66, Asterisk 11.

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@foovax wrote:

Purchased a UC100 PBXact appliance along with a D100 Transcoder card.

Was informed this should be a plug and play install but I’m not finding anything in the GUI that would allow me to install this card without performing the CLI routines defined here:

https://wiki.freepbx.org/display/MTC/Asterisk+D100+Single+Server+Installation#space-menu-link-content

NOTE: I have installed this card and had it working 100% on the appliance, however, I wasn’t comfortable with that due to some issues I had to overcome in following the documented process.

Before I go down the road compiling drivers/modules is there a GUI method in FreePBX that will identify and setup the card on the system?

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@rakireply wrote:

Hello all,
Good day,

I am new on Freepbx config’s. I have got an issue where our calls gets Hung in between for couple of seconds. Not sure where is the problem, weather Network side or in an config in PBX. Currently we have a separate dedicated VLAN for VOIP also PBX server has two Nics one to publi IP configured and another was connected to LAN VOIP Vlan.

Can you guys help me out to diagnosis, below is my sip

Global Settings:

UDP Bindaddress: 0.0.0.0:5061
TCP SIP Bindaddress: Disabled
TLS SIP Bindaddress: Disabled
Videosupport: No
Textsupport: No
Ignore SDP sess. ver.: No
AutoCreate Peer: Off
Match Auth Username: No
Allow unknown access: Yes
Allow subscriptions: Yes
Allow overlap dialing: Yes
Allow promisc. redir: No
Enable call counters: No
SIP domain support: No
Path support : No
Realm. auth: No
Our auth realm asterisk
Use domains as realms: No
Call to non-local dom.: Yes
URI user is phone no: No
Always auth rejects: Yes
Direct RTP setup: No
User Agent: FPBX-13.0.192.9(13.9.1)
SDP Session Name: Asterisk PBX 13.9.1
SDP Owner Name: root
Reg. context: (not set)
Regexten on Qualify: No
Trust RPID: No
Send RPID: No
Legacy userfield parse: No
Send Diversion: Yes
Caller ID: Unknown
From: Domain:
Record SIP history: Off
Auth. Failure Events: Off
T.38 support: No
T.38 EC mode: Unknown
T.38 MaxDtgrm: 4294967295
SIP realtime: Disabled
Qualify Freq : 60000 ms
Q.850 Reason header: No
Store SIP_CAUSE: No

Network QoS Settings:

IP ToS SIP: CS3
IP ToS RTP audio: EF
IP ToS RTP video: AF41
IP ToS RTP text: CS0
802.1p CoS SIP: 4
802.1p CoS RTP audio: 5
802.1p CoS RTP video: 6
802.1p CoS RTP text: 5
Jitterbuffer enabled: No

Network Settings:

SIP address remapping: Disabled
Externhost:
Externaddr: (null)
Externrefresh: 10
Localnet: 12.210.165.176/255.255.255.248
12.194.24.0/255.255.255.0
192.168.10.0/255.255.255.0

Global Signalling Settings:

Codecs: (ulaw|alaw|gsm|g726|g729)
Relax DTMF: No
RFC2833 Compensation: No
Symmetric RTP: Yes
Compact SIP headers: No
RTP Keepalive: 0 (Disabled)
RTP Timeout: 30
RTP Hold Timeout: 300
MWI NOTIFY mime type: application/simple-message-summary
DNS SRV lookup: No
Pedantic SIP support: Yes
Reg. min duration 60 secs
Reg. max duration: 3600 secs
Reg. default duration: 120 secs
Sub. min duration 60 secs
Sub. max duration: 3600 secs
Outbound reg. timeout: 20 secs
Outbound reg. attempts: 0
Outbound reg. retry 403:0
Notify ringing state: Yes
Include CID: No
Notify hold state: Yes
SIP Transfer mode: open
Max Call Bitrate: 384 kbps
Auto-Framing: No
Outb. proxy:
Session Timers: Accept
Session Refresher: uas
Session Expires: 1800 secs
Session Min-SE: 90 secs
Timer T1: 500
Timer T1 minimum: 100
Timer B: 32000
No premature media: Yes
Max forwards: 70

Default Settings:

Allowed transports: UDP
Outbound transport: UDP
Context: from-sip-external
Record on feature: automon
Record off feature: automon
Force rport: Yes
DTMF: rfc2833
Qualify: 0
Keepalive: 0
Use ClientCode: No
Progress inband: No
Language: en
Tone zone:
MOH Interpret: default
MOH Suggest:
Voice Mail Extension: *97

==========================

Interface em1 IP: 192.168.10.2
Interface em1 MAC: B0:83:FF:D8:2B:98
Interface em2 IP: 12.210.165.178
Interface em2 MAC: B0:73:FD:D1:2B:cldck950:

[root@pbx01 asterisk]# route -n
Kernel IP routing table
Destination Gateway Genmask Flags Metric Ref Use Iface
12.210.165.176 0.0.0.0 255.255.255.248 U 0 0 0 em2
12.194.24.0 12.210.165.177 255.255.255.0 UG 0 0 0 em2
192.168.10.0 0.0.0.0 255.255.255.0 U 0 0 0 em1
169.254.0.0 0.0.0.0 255.255.0.0 U 1002 0 0 em1
169.254.0.0 0.0.0.0 255.255.0.0 U 1003 0 0 em2
0.0.0.0 192.168.10.1 0.0.0.0 UG 0 0 0 em1
[root@pbx01 asterisk]#

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@ngv wrote:

Hi Guys,

We are really stuck with this issue and would greatly appreciate some advice or insight of any kind. We recently swapped our Elastix PBX out with FreePBX. Where we had no issues.

We have our own voip network setup with over 100 extensions and there are no problems with audio with internal calls or external calls on the same subnet (10.21.8.0/22)

We also have a vpn from our office network that we make calls to extesnsion on the local network and there is no issues with audio. (172.16.50.0/24)

There is a cross connect from our network across to another office network that we do not manage, calls on this network can call each other with no audio issues, but if they try call the local network there is no audio. (192.168.114.0/24)

Firewalls have been thoroughly checked and temporarily disabled on the PBX and our Router. Fail2ban has also been checked.

The RTP config is all standard, using ports 10000 – 20000.

We have taken the voip phone from their network and tested in ours and all works ok there.
In a packet capture taken from our network (please see attached), we can see ICMP replies ‘Destination unreachable (port unreachable)’ however we can successfully ping the device from the same PBX to the other office network and get replies.

What is even stranger, we have another site setup with almost an identical configuration and same version of freePBX and do not experience the issues.

Freepbx version 14.0.1.24
PBX Firmware:12.7.4-1712-2.sng7
Asterisk 13.18.4

We are using PJSIP for the extesnions

rtp7900.jpg1349x460 197 KB

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@jessy5765 wrote:

So we just implemented out first LLDP and VLAN phone system today. Everything seems to work great… sort of… Maybe and I dont know why.

LLDPMED is enabled on the switching and handing out VLAN 25 to Voice. The phones boot up and get the right IP address of 192.168.25.XXX. Our data network is working on 192.168.43.XXX and VLAN1 (default).

I have all phone ports tagged 25 for phones and untagged on data LAN. Confirming nothing can talk to each other now. We then added a firewall rule to allow the 2 networks to talk with HTTP(port 80), ping, and 2001(PBXact GUI).

We can now access the phones web interface and ping them but nothing else. So everything is working.

Now the port on the switch that connects to the phone system is trunked. eth0 is on the data network at 192.168.43.15. I created a new interface on eth0.25 (VLAN) and gave it 192.168.25.8.

I cant ping the phone system or access the web interface on port 2001. I did confirm i have the firewall off. I noticed that the VLAN interface kept saying unconfigured and it was acting wonky. So just for testing I did manually modify the Phones Registration IP and it connected!!! The phone was talking to the PBX. However Phone apps weren’t working and I couldnt call out.

After rebooting the PBX, the phones connected and were able to call out as well as phone apps working. I can still ping the phones and access their web interface. I CANNOT ping the PBX on VLAN25 but I can ping the data VLAN side… wtf is going on?! Is this normal.

I am scared I have got some sort of “unstable” working config and this is going to stop working after a few weeks randomly. lol.

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@monyoudom wrote:

I want to develop a system with can interact with FreePBX server. My workflow of this application is :

• step 1 : Mobile call to the Desk phone
• step 2 : IF 45s no one pick up the desk phone.
• step 3 : Then the FreePBX just request to my chat bot with the extension of the desk phone.

how to make a function that enable the FreePBX server request to my chat bot ?

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