@dmanolis79 wrote:

Hi,

I have a S705 phone with the latest firmware and latest update of Freepbx 14. I have configured the phone to use TLS for registration and SRTP for the media. Everything works for about 2 minutes then the phone looses registration. In order for it to work again i need to reboot it. This does not happen when not using TLS.

I also have a bunch of Polycom’s VVX’s under the same server and dont have that issue. The server is at a remote location.

Posts: 3

Participants: 3

Read full topic

@sipsepp wrote:

Hi FPBX Users and Admins!

In the WebGUI of former FreePBX versions there was the possibilty to add additional SIP configurations under:

Astersik SIP settings > Advanced settings > Other SIP Settings

I’m running FreePBX 14 and there is no Other SIP Settings section.

Do you know where it has been moved or if it has been replaced?

Related post:

Thank you very much!

Regards,
Sipsepp

Posts: 4

Participants: 3

Read full topic

@ddussaultce wrote:

Hi

I’m testing FreePBX and have issues with the IVR, I am trying to have a main recording that can change depending of the date/hour and right after have another IVR that plays.

it looks like this: welcome to… (main message, allows to dial extension) - If you know the… (2nd IVR)

I tried to put a timeout of 0 retries and timeout even changing to 1 second, the system sends a message telling there wasn’t any valid response (or something like this, working with the French version) even if the timeout message is at None.

I have put the timeout destination to the 2nd IVR but never reaches it.

Guess I have something wrong in my config but can’t figure it out.

image.png769x1008 48.9 KB

Posts: 6

Participants: 2

Read full topic

@DennisT wrote:

I’m trying to figure out what version of AsteriskNow(?) I have and how to upgrade. This is on a VMware VM. The main page when I log onto the web admin page says FreePBX and has an *NOW icon on the upper button bar.
System Overview shows FreePBX 12.0.76.6
Asterisk Info shows Asterisk (Ver. 11.16.0)
System Admin shows PBX Firmware as 6.12.65-26 & PBX Service Pack: 1.0.0.0

If I do a download of support files (FreePBX Versions) I see
asterisk-cli: 2.11.0.3
asteriskinfo: 12.0.2
(not sure what else is relevant)

So, what version am I running and how do I upgrade? Since I’m not running any digium hardware can I upgrade to straight FreePBX?

Posts: 1

Participants: 1

Read full topic

@rwllr wrote:

I want to have a custom script run before calling for a specific Outbound Route that does a lookup and CID lookup.
When using the macro-dialout-trunk-predial-hook it works fine and does as I’d expect, but when I change the context to outrt-5-custom (the Outbound Route I’m trying to apply to, it seems to do nothing.

[macro-dialout-trunk-predial-hook]
;[outrt-5-custom]
exten => s,1,Verbose(breakout_reached)
 same => n,Read(TICKET,"goodbye",,,3,5)
; same => n,Set(CALLERID(num)=${TICKET})
 same => n,Set(CHANNEL(hangup_handler_push)=out_ticket_hangup_handler,s,1) ;
 same => n,AGI(csm_number_check.php,${TICKET})

Posts: 2

Participants: 2

Read full topic

@sentinelace wrote:

Even buying the company that created asterisk the stock is still down and dirt cheap. I know nothing about stock, but the company has more than doubled in size since we started using them. Did the get a lot of debt?

Posts: 2

Participants: 2

Read full topic

@wizworks wrote:

I have two PBX in different locations:

1. US
2. Paris

There are conferences hosted on the US server and there is an IAX trunk between both servers. Extension to extension calls flow across fine and with caller ID etc. No problems there.

The problem is when user on Paris PBX tries to call a conference (302) on US server. They dial 302 and hear “Your call cannot be completed as dialed”

I have the right match in the outbound route and I can see the call progressing across the trunk and reaches the US server but yet cannot complete? what gives?

Here’s the CLI debug output on the US server:
VERSION : 2
CALLED NUMBER : 302
CODEC_PREFS : (ulaw|alaw|gsm)
CALLING NUMBER : 24001
CALLING PRESNTN : 0
CALLING TYPEOFN : 0
CALLING TRANSIT : 0
CALLING NAME : K1WIZ John
LANGUAGE : en
FORMAT : 4
FORMAT2 : ulaw
CAPABILITY : 14
CAPABILITY2 : Unknown
ADSICPE : 2
DATE TIME : 2018-09-14 22:27:20
CALLTOKEN : 51 bytes

Please help - what can I do?

Posts: 2

Participants: 2

Read full topic

@GSnover wrote:

Ok - As the following two posts show, moving directly from FreePBX-13/Sangoma 6.5 to FreePBX-14/Sangoma 7 is no longer directly possible…

But there IS a path to do a backup and restore so that ALL of your settings, recordings, and most importantly End Point Manager is preserved - Read on!

1. Create a Backup of the existing machine.
2. Scratch-Load 10.13.66 - 64-Bit in a VirtualBox VM - Make sure you clone the MAC address of the old machine - You will also need to do this on a seperate LAN to make sure you don’t interfere with the running machine.
3. Restore your Backup to the VirtualBox Machine.
4. Update to the latest 10.13.66 - 22 is current.
5. Upgrade the VirtualBox VM to Sangoma 7 using the script here: https://wiki.freepbx.org/display/PPS/Upgrading+from+FreePBX+10.13.66+to+SNG7
6. Run a Complete Backup on the New Machine.
7. Load a Hyper-V copy of Sangoma-7 - again, preserve the MAC address for activation later.
8. Restore your Backup - You now have upgraded from 13/6.5 to 14/7.5 with ALL your settings preserved.

I would not go this route for a simple box, but for a complex box, it is WAY better than recreating all the things that the Migration script misses.

Posts: 1

Participants: 1

Read full topic

@connect wrote:

Hello

I am new here and also new to FreePBX. I successfully got FreePBX working with a Snom320 extension (calls and everything works fine). Now I am trying to get a Panasonic KX-UDS124 Multi-Cell DECT system with a KX-UDT121 handset connected to my FreePBX. Note that the basic Air setup is working fine, i.e. the handset is registered to the UDS124.

My local freepbx is reachable inside my network through the host name freepbx.local

These are the basic voip settings of the UDS124 (base station):

image.png603x908 37.7 KB

Handset settings:

My extension (202) is CHAN_SIP and listening on Port 5160 (UDP). All configs are applied.

I tried various settings, but am always getting either DNS error or a 403 response.

Has anyone gotten a Panasonic UDS124 setup working with FreePBX and would be willing to share their config?

Posts: 1

Participants: 1

Read full topic

@phirephoto wrote:

So I was attempting to upgrade my PHP version… I can’t recall exactly what I did - want to say tried to upgrade from source, that didn’t work, tried a yum update, same thing. Finally uninstalled via yum, rebuilt from source. Had to configure with apxs2 support. So now PHP is installed, but the FreePBX admin no longer works. I’ve been trying to get a hold of my hosting support, but not having much luck. Near as I can tell I’m running
freepbx.x86_64 2.11.0-4_centos6
Linux version 2.6.32-642.4.2.el6.x86_64 (mockbuild@worker1.bsys.centos.org) (gcc version 4.4.7 20120313 (Red Hat 4.4.7-17) (GCC) ) #1 SMP Tue Aug 23 19:58:13 UTC 2016

So after getting a few errors I was able to resolve, when I try to access the web admin for my freepbx, I get
Fatal error : Class ‘DB’ not found in /usr/local/lib/php/DB.php on line 3
Any suggestions?

Thanks!

Posts: 1

Participants: 1

Read full topic

@bajramia wrote:

Hi All,
I have install zulu mobile app and connected it works i can make call and receive calls but no audio

When i look the asterisk logs i get this

0x7faae8ed7d40 – Strict RTP learning after remote address set to: 76.23.183.29:53036
[2018-09-15 21:08:44] ERROR[14071]: pjproject:0 <?>: icess0x7faae8eef5d8 …Error sending STUN request: Network is unreachable
[2018-09-15 21:08:44] ERROR[24304]: pjproject:0 <?>: icess0x7faae8eef5d8 …Error sending STUN request: Network is unreachable
[2018-09-15 21:08:44] ERROR[24304]: pjproject:0 <?>: icess0x7faae8eef5d8 …Error sending STUN request: Network is unreachable
[2018-09-15 21:08:44] ERROR[24304]: pjproject:0 <?>: icess0x7faae8eef5d8 …Error sending STUN request: Network is unreachable
[2018-09-15 21:08:44] ERROR[24304]: pjproject:0 <?>: icess0x7faae8eef5d8 …Error sending STUN request: Network is unreachable

Thank you for your help.

Posts: 2

Participants: 2

Read full topic

@sentinelace wrote:

When I record a call on a transfer to my extension it records fine. If a call is parked, the recording does not work. Am I missing a setting?

Posts: 2

Participants: 2

Read full topic

@PBXUser1234 wrote:

Hello,

I’m new to Asterisk/FreePBX… I setup a server running FreePBX 14 and Asterisk 13.

I setup the SIP trunks properly (I believe) and created two test extensions. I have connected to the two test extensions via X-Lite on two different computers. The extensions can call each other no problem, but when they connect there’s no audio at all from either end.

I can also successfully make an external call and receive a call from external parties, but again no audio at all.

I have tried to disable the firewall in FreePBX, but that does not help - and I have tried to increase the port range for RTP which does not help. I have also ensured that the “Local Networks” under “NAT Settings” is set appropriate with our private address scheme.

Any suggestions for how I can get this working properly?

Thanks in advance!

Posts: 1

Participants: 1

Read full topic

@kwriley87 wrote:

Hello-

I am using FreePBX Distro 14.

I am running into an issue this morning where incoming calls are not working… I have a packet capture from my carrier showing that the 200OK they are receiving from us has the incorrect public IP in the contact field.

The correct public is set within General SIP Settings and Chan SIP Settings. I’ve rebooted the box, etc. and the contact IP is still incorrect.

Where else should I be looking?

Posts: 1

Participants: 1

Read full topic

@echo501 wrote:

Hello,

I don’t have a lot of experience managing Asterisk or FreeBPX. however I noticed the following in my logs:

[2018-09-17 13:06:47] WARNING[17592] pbx_config.c: The use of ‘_.’ for an extension is strongly discouraged and can have unexpected behavior. Please use ‘_X.’ instead at line 5648 of /etc/asterisk/extensions_additional.conf
[2018-09-17 13:06:47] WARNING[17592] pbx_config.c: The use of ‘_.’ for an extension is strongly discouraged and can have unexpected behavior. Please use ‘_X.’ instead at line 5649 of /etc/asterisk/extensions_additional.conf
[2018-09-17 13:06:47] WARNING[17592] pbx_config.c: The use of ‘_.’ for an extension is strongly discouraged and can have unexpected behavior. Please use ‘_X.’ instead at line 127 of extensions.conf
[2018-09-17 13:06:47] WARNING[17592] pbx_config.c: The use of ‘_.’ for an extension is strongly discouraged and can have unexpected behavior. Please use ‘_X.’ instead at line 128 of extensions.conf

I vi’d the 2 files in question and found the lines where there is indeed a “_.”. There is a header at the beginning of each file stating that FreePBX has control of these files.

So my questions are:

1. is something to be worried about these warnings?
2. where in the FreePBX gui do I make the needed changes?

Thanks in advance.

–Kenny

Posts: 4

Participants: 3

Read full topic

@ddussaultce wrote:

I’m trying to setup my system before moving it to production and got a little snag that can’t figure out how to do it.

I have the system in both French and English (main language is French). I have a main welcome message in both languages and want to add 9 for English. after the timeout that is configured in the main welcome message it goes in a complex time condition for holidays, opening hours… but the time condition uses the default language (French). Is there a way to specify a different language in the time condition or there would be a better way to do it ? all my recordings have a French and English version recorded so I can change the language easily.

Now:
Main message (bilingual) ---- time condition (French)

Expecting:
Main message (bilingual) and users press 9 — time condition (English)

Posts: 1

Participants: 1

Read full topic

@thebrucey wrote:

Hi there please can you help me. I get these messages all the time and am just really concerned its someone or something hacking the PBX.

[2018-09-18 13:40:02] SECURITY[2857] res_security_log.c: SecurityEvent=“SuccessfulAuth”,EventTV=“2018-09-18T13:40:02.023+0000”,Severity=“Informational”,Service=“AMI”,EventVersion=“1”,AccountID=“admin”,SessionID=“0x33772f0”,LocalAddress=“IPV4/TCP/0.0.0.0/5038”,RemoteAddress=“IPV4/TCP/127.0.0.1/48666”,UsingPassword=“0”,SessionTV=“2018-09-18T13:40:02.023+0000”
[2018-09-18 13:40:24] SECURITY[2857] res_security_log.c: SecurityEvent=“SuccessfulAuth”,EventTV=“2018-09-18T13:40:24.685+0000”,Severity=“Informational”,Service=“AMI”,EventVersion=“1”,AccountID=“admin”,SessionID=“0x7faea0043fe0”,LocalAddress=“IPV4/TCP/0.0.0.0/5038”,RemoteAddress=“IPV4/TCP/127.0.0.1/48670”,UsingPassword=“0”,SessionTV=“2018-09-18T13:40:24.685+0000”
[2018-09-18 13:41:00] SECURITY[2857] res_security_log.c: SecurityEvent=“SuccessfulAuth”,EventTV=“2018-09-18T13:41:00.058+0000”,Severity=“Informational”,Service=“AMI”,EventVersion=“1”,AccountID=“admin”,SessionID=“0x7faeac4a1370”,LocalAddress=“IPV4/TCP/0.0.0.0/5038”,RemoteAddress=“IPV4/TCP/127.0.0.1/48674”,UsingPassword=“0”,SessionTV=“2018-09-18T13:41:00.058+0000”
[2018-09-18 13:41:02] SECURITY[2857] res_security_log.c: SecurityEvent=“SuccessfulAuth”,EventTV=“2018-09-18T13:41:02.008+0000”,Severity=“Informational”,Service=“AMI”,EventVersion=“1”,AccountID=“admin”,SessionID=“0x7faea8003750”,LocalAddress=“IPV4/TCP/0.0.0.0/5038”,RemoteAddress=“IPV4/TCP/127.0.0.1/48678”,UsingPassword=“0”,SessionTV=“2018-09-18T13:41:02.008+0000”
[2018-09-18 13:41:02] SECURITY[2857] res_security_log.c: SecurityEvent=“SuccessfulAuth”,EventTV=“2018-09-18T13:41:02.014+0000”,Severity=“Informational”,Service=“AMI”,EventVersion=“1”,AccountID=“admin”,SessionID=“0x7faeb4392ba0”,LocalAddress=“IPV4/TCP/0.0.0.0/5038”,RemoteAddress=“IPV4/TCP/127.0.0.1/48682”,UsingPassword=“0”,SessionTV=“2018-09-18T13:41:02.014+0000”
[2018-09-18 13:41:02] SECURITY[2857] res_security_log.c: SecurityEvent=“SuccessfulAuth”,EventTV=“2018-09-18T13:41:02.026+0000”,Severity=“Informational”,Service=“AMI”,EventVersion=“1”,AccountID=“admin”,SessionID=“0x7faeb0c52f60”,LocalAddress=“IPV4/TCP/0.0.0.0/5038”,RemoteAddress=“IPV4/TCP/127.0.0.1/48686”,UsingPassword=“0”,SessionTV=“2018-09-18T13:41:02.026+0000”
[2018-09-18 13:41:35] SECURITY[2857] res_security_log.c: SecurityEvent=“SuccessfulAuth”,EventTV=“2018-09-18T13:41:35.417+0000”,Severity=“Informational”,Service=“AMI”,EventVersion=“1”,AccountID=“admin”,SessionID=“0x7faebc173cf0”,LocalAddress=“IPV4/TCP/0.0.0.0/5038”,RemoteAddress=“IPV4/TCP/127.0.0.1/48690”,UsingPassword=“0”,SessionTV=“2018-09-18T13:41:35.417+0000”
[2018-09-18 13:42:01] SECURITY[2857] res_security_log.c: SecurityEvent=“SuccessfulAuth”,EventTV=“2018-09-18T13:42:01.929+0000”,Severity=“Informational”,Service=“AMI”,EventVersion=“1”,AccountID=“admin”,SessionID=“0x7faeb8bb0920”,LocalAddress=“IPV4/TCP/0.0.0.0/5038”,RemoteAddress=“IPV4/TCP/127.0.0.1/48694”,UsingPassword=“0”,SessionTV=“2018-09-18T13:42:01.929+0000”
[2018-09-18 13:42:01] SECURITY[2857] res_security_log.c: SecurityEvent=“SuccessfulAuth”,EventTV=“2018-09-18T13:42:01.968+0000”,Severity=“Informational”,Service=“AMI”,EventVersion=“1”,AccountID=“admin”,SessionID=“0x33772f0”,LocalAddress=“IPV4/TCP/0.0.0.0/5038”,RemoteAddress=“IPV4/TCP/127.0.0.1/48702”,UsingPassword=“0”,SessionTV=“2018-09-18T13:42:01.968+0000”
[2018-09-18 13:42:01] SECURITY[2857] res_security_log.c: SecurityEvent=“SuccessfulAuth”,EventTV=“2018-09-18T13:42:01.976+0000”,Severity=“Informational”,Service=“AMI”,EventVersion=“1”,AccountID=“admin”,SessionID=“0x7faec002a290”,LocalAddress=“IPV4/TCP/0.0.0.0/5038”,RemoteAddress=“IPV4/TCP/127.0.0.1/48698”,UsingPassword=“0”,SessionTV=“2018-09-18T13:42:01.976+0000”
[2018-09-18 13:42:10] SECURITY[2857] res_security_log.c: SecurityEvent=“SuccessfulAuth”,EventTV=“2018-09-18T13:42:10.756+0000”,Severity=“Informational”,Service=“AMI”,EventVersion=“1”,AccountID=“admin”,SessionID=“0x7faea0043fe0”,LocalAddress=“IPV4/TCP/0.0.0.0/5038”,RemoteAddress=“IPV4/TCP/127.0.0.1/48706”,UsingPassword=“0”,SessionTV=“2018-09-18T13:42:10.756+0000”
[2018-09-18 13:42:46] SECURITY[2857] res_security_log.c: SecurityEvent=“SuccessfulAuth”,EventTV=“2018-09-18T13:42:46.107+0000”,Severity=“Informational”,Service=“AMI”,EventVersion=“1”,AccountID=“admin”,SessionID=“0x7faeac4a1370”,LocalAddress=“IPV4/TCP/0.0.0.0/5038”,RemoteAddress=“IPV4/TCP/127.0.0.1/48710”,UsingPassword=“0”,SessionTV=“2018-09-18T13:42:46.107+0000”
[2018-09-18 13:43:01] SECURITY[2857] res_security_log.c: SecurityEvent=“SuccessfulAuth”,EventTV=“2018-09-18T13:43:01.840+0000”,Severity=“Informational”,Service=“AMI”,EventVersion=“1”,AccountID=“admin”,SessionID=“0x7faea8003750”,LocalAddress=“IPV4/TCP/0.0.0.0/5038”,RemoteAddress=“IPV4/TCP/127.0.0.1/48714”,UsingPassword=“0”,SessionTV=“2018-09-18T13:43:01.840+0000”
[2018-09-18 13:43:01] SECURITY[2857] res_security_log.c: SecurityEvent=“SuccessfulAuth”,EventTV=“2018-09-18T13:43:01.861+0000”,Severity=“Informational”,Service=“AMI”,EventVersion=“1”,AccountID=“admin”,SessionID=“0x7faeb4392ba0”,LocalAddress=“IPV4/TCP/0.0.0.0/5038”,RemoteAddress=“IPV4/TCP/127.0.0.1/48718”,UsingPassword=“0”,SessionTV=“2018-09-18T13:43:01.861+0000”
[2018-09-18 13:43:01] SECURITY[2857] res_security_log.c: SecurityEvent=“SuccessfulAuth”,EventTV=“2018-09-18T13:43:01.884+0000”,Severity=“Informational”,Service=“AMI”,EventVersion=“1”,AccountID=“admin”,SessionID=“0x7faeb0c52f60”,LocalAddress=“IPV4/TCP/0.0.0.0/5038”,RemoteAddress=“IPV4/TCP/127.0.0.1/48722”,UsingPassword=“0”,SessionTV=“2018-09-18T13:43:01.884+0000”

Posts: 2

Participants: 2

Read full topic

@phreakazoid wrote:

Hi, I am new at FreePBX. I moved to FreePBX 3 Month ago. Before that I have used the internal VOIP Service of an AVM Fritz.Box.

Time condition is working as expected. If someone calls after 8 in the evening he will be redirected to voicemail, so my kids will not get awake.
BUT, I didnt find a solution to allow, lets call them VIP numbers, to call through at night mode. For example I am on my way back home from work and my car strikes and I want to call my wife during night mode.
Is there a way to allow specific numbers to call through during active time conditions?

Posts: 1

Participants: 1

Read full topic

@Rick wrote:

Hosted system: Host company restarted equipment and Freepbx came back up but problem with trunks working.
The system showed trunks registered but the trunking company would not register. After working with tech support, found after rebooting the hardware and the pbx is up and running, had to issue a “fwconsole reload” command in order to register trunks with trunking company.
Is there a fix for this?
Not until this command was issued would the trunks connect though pbx showed registered the whole time.

Posts: 1

Participants: 1

Read full topic

@SteveMat11 wrote:

I have 3 Sangoma phones plus a Sangoma cordless set up in my home. I have a weird situation where if I intercom any of the phones, 1 specific phone always displays the call (even if this particular phone isn’t one of the phones called) For example, the kitchen phone is always ringing and displays the call origination caller ID and then switches to a busy signal say if the cordless dials the bedroom. Ive searched in endpoint manager to make sure the users are properly linked to their respective phones. I can not think of anything other reason this could be happening. Any ideas?

Posts: 1

Participants: 1

Read full topic