@smwein wrote:

I’m trying to setup Asternics Call Center Stats but I’m not getting any data in there. When I look at my queue_log it is empty. How do make sure data is write to that.

Thanks

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@faisalkhan wrote:

hi Guys,

I have a pbx running with one NIC which is configured in a way that it’s on Private Lan and gateway is my Internet Firewall so all my carriers like FR, SR are configured properly on this network card.

However Now I have to add another card with IP scheme from My other Sip provider P2P.

This provider has provided me with bunch of ip addresses and gateway and Sip Proxy.

I have to configure the network card with it’s ip address and gateway then I will be able to reach out to this Sip Provider network to make sip trunk.

Scenario is Internet Firewall gateway is for All the international Traffic or Trunks with International providers and this second interface with P2P network my end users can dial Local Calling to my country.

Now I need to add the second Nic with gateway and subnetmask and it should be accessible from the phones on the private network.

Please guide me.

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@salvy12477 wrote:

After restarting the asterisk service in FreePBX command line the service now shows;

root@TOWER-NYSAU-PBX-02 ~]# service asterisk status
asterisk dead but subsys locked
[root@TOWER-NYSAU-PBX-02 ~]#

(The commands I used to restart was service asterisk restart)
I’m assuming that there’s a process thats stuck and or locked out but am unable to locate it. Also I am fairly new to linux and only recently have been put incharge of maintaining the phone system.

I’ve also started asterisk using; asterisk -vvvc with the below output;

Asterisk 13.18.2, Copyright © 1999 - 2014, Digium, Inc. and others.
Created by Mark Spencer markster@digium.com
Asterisk comes with ABSOLUTELY NO WARRANTY; type ‘core show warranty’ for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type ‘core show license’ for details.

[ Initializing Custom Configuration Options ]
== Parsing ‘/etc/asterisk/asterisk.conf’: Found
Couldn’t find manager DBGet in XML documentation
Couldn’t find manager DBGet in XML documentation
== Manager registered action DBGet
Couldn’t find manager DBPut in XML documentation
Couldn’t find manager DBPut in XML documentation
== Manager registered action DBPut
Couldn’t find manager DBDel in XML documentation
Couldn’t find manager DBDel in XML documentation
== Manager registered action DBDel
Couldn’t find manager DBDelTree in XML documentation
Couldn’t find manager DBDelTree in XML documentation
== Manager registered action DBDelTree
PBX UUID: f846665d-a4e5-4879-8ab7-5b06f24da8f3
== Registered ‘audio’ codec ‘g723’ at sample rate ‘8000’ with id ‘1’
== Created cached format with name ‘g723’
== Registered ‘audio’ codec ‘ulaw’ at sample rate ‘8000’ with id ‘2’
== Created cached format with name ‘ulaw’
== Registered ‘audio’ codec ‘alaw’ at sample rate ‘8000’ with id ‘3’
== Created cached format with name ‘alaw’
== Registered ‘audio’ codec ‘gsm’ at sample rate ‘8000’ with id ‘4’
== Created cached format with name ‘gsm’
== Registered ‘audio’ codec ‘g726’ at sample rate ‘8000’ with id ‘5’
== Created cached format with name ‘g726’
== Registered ‘audio’ codec ‘g726aal2’ at sample rate ‘8000’ with id ‘6’
== Created cached format with name ‘g726aal2’
== Registered ‘audio’ codec ‘adpcm’ at sample rate ‘8000’ with id ‘7’
== Created cached format with name ‘adpcm’
== Registered ‘audio’ codec ‘slin’ at sample rate ‘8000’ with id ‘8’
== Created cached format with name ‘slin’
== Registered ‘audio’ codec ‘slin’ at sample rate ‘12000’ with id ‘9’
== Created cached format with name ‘slin12’
== Registered ‘audio’ codec ‘slin’ at sample rate ‘16000’ with id ‘10’
== Created cached format with name ‘slin16’
== Registered ‘audio’ codec ‘slin’ at sample rate ‘24000’ with id ‘11’
== Created cached format with name ‘slin24’
== Registered ‘audio’ codec ‘slin’ at sample rate ‘32000’ with id ‘12’
== Created cached format with name ‘slin32’
== Registered ‘audio’ codec ‘slin’ at sample rate ‘44100’ with id ‘13’
== Created cached format with name ‘slin44’
== Registered ‘audio’ codec ‘slin’ at sample rate ‘48000’ with id ‘14’
== Created cached format with name ‘slin48’
== Registered ‘audio’ codec ‘slin’ at sample rate ‘96000’ with id ‘15’
== Created cached format with name ‘slin96’
== Registered ‘audio’ codec ‘slin’ at sample rate ‘192000’ with id ‘16’
== Created cached format with name ‘slin192’
== Registered ‘audio’ codec ‘lpc10’ at sample rate ‘8000’ with id ‘17’
== Created cached format with name ‘lpc10’
== Registered ‘audio’ codec ‘g729’ at sample rate ‘8000’ with id ‘18’
== Created cached format with name ‘g729’
== Registered ‘audio’ codec ‘speex’ at sample rate ‘8000’ with id ‘19’
== Created cached format with name ‘speex’
== Registered ‘audio’ codec ‘speex’ at sample rate ‘16000’ with id ‘20’
== Created cached format with name ‘speex16’
== Registered ‘audio’ codec ‘speex’ at sample rate ‘32000’ with id ‘21’
== Created cached format with name ‘speex32’
== Registered ‘audio’ codec ‘ilbc’ at sample rate ‘8000’ with id ‘22’
== Created cached format with name ‘ilbc’
== Registered ‘audio’ codec ‘g722’ at sample rate ‘16000’ with id ‘23’
== Created cached format with name ‘g722’
== Registered ‘audio’ codec ‘siren7’ at sample rate ‘16000’ with id ‘24’
== Created cached format with name ‘siren7’
== Registered ‘audio’ codec ‘siren14’ at sample rate ‘32000’ with id ‘25’
== Created cached format with name ‘siren14’
== Registered ‘audio’ codec ‘testlaw’ at sample rate ‘8000’ with id ‘26’
== Created cached format with name ‘testlaw’
== Registered ‘audio’ codec ‘g719’ at sample rate ‘48000’ with id ‘27’
== Created cached format with name ‘g719’
== Registered ‘audio’ codec ‘opus’ at sample rate ‘48000’ with id ‘28’
== Created cached format with name ‘opus’
== Registered ‘image’ codec ‘jpeg’ at sample rate ‘0’ with id ‘29’
== Created cached format with name ‘jpeg’
== Registered ‘image’ codec ‘png’ at sample rate ‘0’ with id ‘30’
== Created cached format with name ‘png’
== Registered ‘video’ codec ‘h261’ at sample rate ‘1000’ with id ‘31’
== Created cached format with name ‘h261’
== Registered ‘video’ codec ‘h263’ at sample rate ‘1000’ with id ‘32’
== Created cached format with name ‘h263’
== Registered ‘video’ codec ‘h263p’ at sample rate ‘1000’ with id ‘33’
== Created cached format with name ‘h263p’
== Registered ‘video’ codec ‘h264’ at sample rate ‘1000’ with id ‘34’
== Created cached format with name ‘h264’
== Registered ‘video’ codec ‘mpeg4’ at sample rate ‘1000’ with id ‘35’
== Created cached format with name ‘mpeg4’
== Registered ‘video’ codec ‘vp8’ at sample rate ‘1000’ with id ‘36’
== Created cached format with name ‘vp8’
== Registered ‘video’ codec ‘vp9’ at sample rate ‘1000’ with id ‘37’
== Created cached format with name ‘vp9’
== Registered ‘text’ codec ‘red’ at sample rate ‘0’ with id ‘38’
== Created cached format with name ‘red’
== Registered ‘text’ codec ‘t140’ at sample rate ‘0’ with id ‘39’
== Created cached format with name ‘t140’
== Registered ‘audio’ codec ‘none’ at sample rate ‘8000’ with id ‘40’
== Created cached format with name ‘none’
== Registered ‘audio’ codec ‘silk’ at sample rate ‘8000’ with id ‘41’
== Created cached format with name ‘silk8’
== Registered ‘audio’ codec ‘silk’ at sample rate ‘12000’ with id ‘42’
== Created cached format with name ‘silk12’
== Registered ‘audio’ codec ‘silk’ at sample rate ‘16000’ with id ‘43’
== Created cached format with name ‘silk16’
== Registered ‘audio’ codec ‘silk’ at sample rate ‘24000’ with id ‘44’
== Created cached format with name ‘silk24’
== Sorcery registered wizard ‘bucket’
== Sorcery registered wizard ‘bucket_file’
Cannot update type ‘bucket’ in module ‘core’ because it has no existing documentation!
Failed to register ‘bucket’ object type in Bucket sorcery
Bucket API initialization failed. ASTERISK EXITING!
== Manager unregistered action DBGet
== Manager unregistered action DBPut
== Manager unregistered action DBDel
== Manager unregistered action DBDelTree

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@dstu wrote:

Hello,

I have an old FreePBX v2.11.0.43 (asterisk version 11.25.2) at one end (sys1) and a FreePBX v14.0.13.6 (asterisk version 13.20.0) in the other (sys2), using PJSIP on port 5060. Both are connected behind firewalls and have static public IP addresses (external IP and local network are properly configured on both ends).

I’ve been struggling for many hours with trying to make calls between them.

I was trying initially to create trunks without authentication, but sys1 kept rejecting the calls coming from sys2:

SIP Peer ACL: Rejecting ‘xx.xx.xx.xx’ due to a failure to pass ACL ‘(BASELINE)’

Then, I tried to add a username and secret to the trunk (on both ends), but that didn’t solve the problem either. I also disabled in sys2’s trunk pjsip advanced settings the field “Permanent Auth Rejection”, but that also didn’t solve the problem.

Can anyone suggest trunk settings for both ends to enable site to site communications?

Thank you very much in advance for your kind assistance.

David

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@stevensedory wrote:

Hi All,

We recently converted a server from the chan_sip driver to only pjsip.

Everything is working, but we’re having one way audio issues with an FXO, an Obi 110.

We have this device setup via a Trunk entry. Here’s what we had on chan_sip:

username=8880
type=friend
transport=tcp
secret=somepassword
qualifyfreq=25
qualify=yes
nat=yes
host=dynamic
dtmfmode=auto
context=from-pstn
canreinvite=no

Note the nat and dynamic parts.

I cannot figure out how to replicate this on the pjsip trunk, and therefore we’re having one way audio issues.

I read on this wiki that I need to set the following settings to replicate chan_sip’s “nat=yes”, but I can’t find them in the FreePBX UI under Trunks, only under Extensions, and this device needs to be a Trunk.

https://wiki.asterisk.org/wiki/display/AST/Migrating+from+chan_sip+to+res_pjsip

rtp_symmetric=yes
force_rport=yes
rewrite_contact=yes

Anyone know how I can have a NAT’d dynamic host on pjsip?

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@Red wrote:

Need some ideas. How to get pbx system to jump to other WAN if primary goes down.

two modems----Pfsense—FreePBX

PBX has address of primary Wan but if that goes down, how does it know to use backup Wan? Is there anything within pfsense that would allow phone system to keep working?

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@GRZMRC wrote:

Is it possible to add cisco 7911?
many thanks

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@tomas12343 wrote:

Hello community! Does anyone know how to cancel the attended transfer call to another extension? One scenario is that an inbound call is made to extension 1 and extension 1 transfers the call (attended transfer-*2+extension number) to extension 2. As extension 2 is ringing, extension 1 wants to cancel the transfer and return to the inbound call. How do we do that?

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@Tylersprice wrote:

I Have long reboot times (2+ mins) on vm if I reboot and the system hangs completely if I shutdown from terminal or GUI. However, if I - >
asterisk - r
core stop now
Reboot
it reboots within 20 seconds, any ideas? I’ve checked the logs and nothing pops out at me. I’ve heard the mariadb hangs causing this due to vm RTC being wrong but I’ve set proxmox to use local time and now the logs confirm that the time is synched everywhere. Any advice is appreciated.

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@VoIPTek wrote:

Since yesterday several instances are losing communications ( not sure they are failed or crashed ) and in hunting through logs I’m seeing some strange firewall errors, first below is messages info, and with there is also this line, looking at several areas without a solid reason for the lockup

HP Warning: Wrong license type, license codes are not matching this host or license text has been altered (license file: /etc/schmooze/schmooze.zl). in /usr/lib/sysadmin/licensed.php on line 0
PHP Warning: License check failed! in /usr/lib/sysadmin/licensed.php on line 0
Starting firewall.
1570920605: Wall: 'Firewall service now starting.

Oct 12 15:39:46 bpbx php: Wall: 'Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012' returned 0

Oct 12 15:40:33 bpbx php: Wall: ‘Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012’ returned 0
Oct 12 15:41:21 bpbx php: Wall: ‘Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012’ returned 0
Oct 12 15:42:08 bpbx php: Wall: ‘Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012’ returned 0
Oct 12 15:42:55 bpbx php: Wall: ‘Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012’ returned 0
Oct 12 15:43:42 bpbx php: Wall: ‘Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012’ returned 0
Oct 12 15:44:29 bpbx php: Wall: ‘Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012’ returned 0
Oct 12 15:45:16 bpbx php: Wall: ‘Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012’ returned 0
Oct 12 15:46:04 bpbx php: Wall: ‘Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012’ returned 0
Oct 12 15:46:51 bpbx php: Wall: ‘Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012’ returned 0
Oct 12 15:47:39 bpbx php: Wall: ‘Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012’ returned 0
Oct 12 15:48:26 bpbx php: Wall: ‘Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012’ returned 0
Oct 12 15:49:13 bpbx php: Wall: ‘Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012’ returned 0
Oct 12 15:50:00 bpbx php: Wall: ‘Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012’ returned 0
Oct 12 15:50:47 bpbx php: Wall: ‘Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012’ returned 0
Oct 12 15:51:34 bpbx php: Wall: ‘Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012’ returned 0
Oct 12 15:52:21 bpbx php: Wall: ‘Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012’ returned 0
Oct 12 15:53:09 bpbx php: Wall: ‘Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012’ returned 0
Oct 12 15:53:56 bpbx php: Wall: ‘Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012’ returned 0
Oct 12 15:54:43 bpbx php: Wall: ‘Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012’ returned 0
Oct 12 15:55:31 bpbx php: Wall: ‘Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012’ returned 0
Oct 12 15:56:18 bpbx php: Wall: ‘Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012’ returned 0
Oct 12 15:57:05 bpbx php: Wall: ‘Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012’ returned 0
Oct 12 15:57:52 bpbx php: Wall: ‘Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012’ returned 0
Oct 12 15:58:39 bpbx php: Wall: ‘Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012’ returned 0
Oct 12 15:59:26 bpbx php: Wall: ‘Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012’ returned 0
Oct 12 19:01:07 bpbx php: Wall: ‘Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012’ returned 0
Oct 12 19:01:56 bpbx php: Wall: ‘Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012’ returned 0
Oct 12 19:02:43 bpbx php: Wall: ‘Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012’ returned 0
Oct 12 19:03:31 bpbx php: Wall: ‘Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012’ returned 0
Oct 12 19:04:19 bpbx php: Wall: ‘Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012’ returned 0
Oct 12 19:05:06 bpbx php: Wall: ‘Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012’ returned 0
Oct 12 19:05:54 bpbx php: Wall: ‘Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012’ returned 0
Oct 12 19:06:41 bpbx php: Wall: ‘Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012’ returned 0
Oct 12 19:07:29 bpbx php: Wall: ‘Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012’ returned 0
Oct 12 19:08:16 bpbx php: Wall: ‘Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012’ returned 0
Oct 12 19:09:04 bpbx php: Wall: ‘Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012’ returned 0
Oct 12 19:09:51 bpbx php: Wall: ‘Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012’ returned 0
Oct 12 19:10:40 bpbx php: Wall: ‘Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012’ returned 0
Oct 12 19:11:27 bpbx php: Wall: ‘Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012’ returned 0
Oct 12 19:12:14 bpbx php: Wall: ‘Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012’ returned 0
Oct 12 19:13:02 bpbx php: Wall: ‘Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012’ returned 0
Oct 12 19:13:49 bpbx php: Wall: ‘Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012’ returned 0

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@wisadmin wrote:

Well my FreePBX was up and running fine with my old Asus router and port forwarding. LIfe changes took me away from installation for about 2.5 months, switch out asus for PFsense.

Calls don’t work incoming or outgoing (sound). Nothing has changed except updates/patching. I even tried to revert to a snapshot when was working still no sound. I am open for any assistance.

PBX Firmware:

12.7.6-1904-1.sng7

PBX Service Pack:

1.0.0.0

Running asterisk -rvv I am gettting some ast-yyerror* and such. I checked PFsense and it isn’t blocking anything from my SIP provider.

I need a troubleshooting guide as I am new to FreePbx and want to utilize and not just pay SIP provider monthly.

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@Josh9591 wrote:

Hello again.

So I recently had an established IP address of 192.168.2.12 No connection problems what so ever. I had a bug issue where I’d drop calls every 30 second due to a static ip not being set.

I’ve made a reset to the system several times to obtain an ip address of my local Xfinity network. And now, it looks like when I reboot the machine, I see a 50/50 chance of FreePBX not being able to Obtain an IP Address when FreePBX is launched. When I had a connection with freepbx I Could connect to it using Ssh and the GUI just fine. When I go to the system modules. the Page times out. And now i’m unable to make an established ip address to connect to the freepbx using gui. The last error in the console was “unable to write /etc/wanpipe/global.conf disable sangoma DIGIUM or Change Permissions to /etc/wanpipe/global.conf” witch I belive caused to drop connections.

when I did “# fwconsole chown” it re wrote the permissions. But then that was like it.

fwconsole chown

I have FreePBX installed on a eMachine Computer I found running through a cisco 24 port unmanaged switch. and I’m on XFinity comcast internet. Thanks Guys!!

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@defres wrote:

Have an existing Avaya system with extensions setup by dept.

I.E.
1xxx Executives
2xxx Billing Dept
3xxx Sames Dept
4xxx Shipping Dept
6xxx Logistics Dept
Etc. The only numbers extensions do not start with is 9

It works just fine and they dial 9 followed by a 7 10 or 11 digit number for outside calls.

Looking to move to PBXaxt probably with Sangoma or Yealink phones and was wondering if was possible to keep this extension plan in place and if so how that would be done? Currently when they enter the last digit of an internal extension I.E. 2004 as soon as the 4 is pressed it rings the extension immediately, no need to press send or anything, Same when dialing an outside number prefixed with 9, I.E. 912125551212 the outside call initiates immediately and starts to ring.

Outside service will be a dedicated PRI provided by Comcast, All internal extensions reachable over the local campus wide LAN.

Is it possible to replicate this functionality and if so how would this be done? If we can avoid the delay or having to press send to initiate the call so it works just like it does now on the Avaya system that would be ideal.

Thanks in advance for your help / ideas

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@bgray wrote:

Hello FreePBX Community!

How often do you use wiki.freepbx.org to find information when you’re trying to build a PBX solution? Do you wish there were more step-by-step guides? Do you think it needs more short how-to articles that describe how to configure a given setup? We do, too!

We’re looking to make the FreePBX Wiki THE place for documentation and troubleshooting on common issues, modules, and set up for all things FreePBX. To that end, we’re committing to getting new and updated articles published more regularly and filling out gaps in the wiki so it can become a full-fledged knowledge base!

While we work to make this a better platform, we need your help!

Uncle-Tango.jpg2000×2469 606 KB

I’d like to announce the FreePBX Community Documentation Project. FreePBX can be implemented in so many ways, and there may be scenarios that even our technical staff haven’t encountered. But looking around this forum, there is a ton of experience and knowledge that can be invaluable to the whole community.

When you create articles for the wiki, we will acknowledge your contributions. Get your name immortalized as a contributor, earn Forum Badges to show your dedication to the community, and even get Sangoma Swag! In fact, we’ll send you a Sangoma coffee mug or a Sangoma USB stick for your first article approved and published!

Additionally, to get us started the first 5 contributors who create an article that gets published will receive a $25 Amazon giftcard!

You can pick a topic from our Most Wanted list here, create an entirely new article, or suggest a topic you’d like to see covered! If you’re interested in helping us with this effort, either reply on this forum post and we’ll reach out to you, or email freepbxdoc@sangoma.com

Have an idea to improve our documentation process? Open a thread here, or email us and we can open a discussion!

Thanks in advance!

BrianG

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@Trovan0 wrote:

Hello everyone, i am getting issue with audio in internal phones conected via OpenVPN.

Freepbx/OpenVPN IP: 192.168.1.250
LocalNetwork: 192.168.1.0/24
VPN Pool: 10.8.0.1/24

The problem is when i turn on iptables the local phones cant hear between two phones, when i turn off, i have audio but dont have network for vpn clients.

There are my iptables:
# Generated by iptables-save v1.4.7 on Fri Oct 11 00:49:51 2019
*nat
:PREROUTING ACCEPT [0:0]
:POSTROUTING ACCEPT [3:779]
:OUTPUT ACCEPT [3:779]
-A POSTROUTING -s 10.8.0.0/24 -o eth0 -j MASQUERADE
COMMIT

Completed on Fri Oct 11 00:49:51 2019

Generated by iptables-save v1.4.7 on Fri Oct 11 00:49:51 2019

*filter
:INPUT ACCEPT [0:0]
:FORWARD ACCEPT [0:0]
:OUTPUT ACCEPT [25:7021]
-A INPUT -i tun+ -j ACCEPT
-A INPUT -i eth0 -p udp -m state --state NEW -m udp --dport 1194 -j ACCEPT
-A INPUT -p tcp -m tcp --dport 22000 -j ACCEPT
-A INPUT -m state --state RELATED,ESTABLISHED -j ACCEPT
-A INPUT -i eth0 -p tcp -m tcp --dport 80 -m state --state NEW,ESTABLISHED -j ACCEPT
-A INPUT -p icmp -j ACCEPT
-A INPUT -i lo -j ACCEPT
-A INPUT -p tcp -m state --state NEW -m tcp --dport 22000 -j ACCEPT
-A INPUT -j REJECT --reject-with icmp-host-prohibited
-A INPUT -p tcp -m tcp --dport 80 -j ACCEPT
-A FORWARD -i tun+ -j ACCEPT
-A FORWARD -i tun+ -o eth0 -m state --state RELATED,ESTABLISHED -j ACCEPT
-A FORWARD -i eth0 -o tun+ -m state --state RELATED,ESTABLISHED -j ACCEPT
COMMIT

Completed on Fri Oct 11 00:49:51 2019

Thanks

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@vcsupport wrote:

Hi,

I have a SIP endpoint connected via a pjsip trunk and a snom 300 phone. When I send a message wait message from the sip trunk to the snom phone I get a 501 error message. Any ideas what config I have missed or is this a bug?

PJSIP trunk is 4000
Snom phone is 4015

Freepbx version: 13.0.197.8
Asterisk Version: 13.12.1

trace from cli:

NOTIFY sip:4015@172.16.2.21 SIP/2.0

From: <sip:4000@172.16.2.21>;tag=ACU-1a6a-a9a4882d

To: <sip:4015@172.16.2.21>

Call-ID: [deb011c5-ef49-11e9-b8b9-b2514979b508@172.16.7.104](mailto:deb011c5-ef49-11e9-b8b9-b2514979b508@172.16.7.104)

CSeq: 22088 NOTIFY

Content-Length: 23

Max-Forwards: 70

Date: Mon, 02 Nov 2009 11:56:07 GMT

Contact: <sip:4000@172.16.2.21>

Event: message-summary

Subscription-State: active

Route: <sip:172.16.2.21:5060;lr>

Content-Type: application/simple-message-summary

Via: SIP/2.0/UDP 172.16.7.104:5060;rport=5060;received=172.16.7.104;branch=z9hG4bKdeb011c6-ef49-11e9-b8b9-b2514979b508

Content-Type: application/simple-message-summary

Content-Length: 23

Messages-Waiting: yes

SIP/2.0 501 Not Implemented

Via: SIP/2.0/UDP 172.16.7.104:5060;rport=5060;received=172.16.7.104;branch=z9hG4bKdeb011c6-ef49-11e9-b8b9-b2514979b508

Call-ID: [deb011c5-ef49-11e9-b8b9-b2514979b508@172.16.7.104](mailto:deb011c5-ef49-11e9-b8b9-b2514979b508@172.16.7.104)

From: <sip:4000@172.16.2.21>;tag=ACU-1a6a-a9a4882d

To: <sip:4015@172.16.2.21>;tag=z9hG4bKdeb011c6-ef49-11e9-b8b9-b2514979b508

CSeq: 22088 NOTIFY

Server: FPBX-13.0.197.8(13.12.1)

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@mvogel4949 wrote:

I have what looks to be an infected system here. I have found 3 php files in particular - (example --> graph.php) that are linked to the infection. How would I run a grep command to search for and delete all the graph.php files? Thank you!

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@Chuckak wrote:

Current PBX Version:14.0.13.4
Current System Version:12.7.6-1904-1.sng7

If I leave the browser, Chrome, Firefox or Edge on the FreePBX dashboard for more than a few hours the Network widget quits updating. Then awhile later I cannot navigate to any tabs. I cannot refresh the browser.
Closing and reopening sometimes works but usually not. Can’t find webpage. If I change browsers it will work for awhile again. If I use the IP address instead of FQDN it will work until it times out again.

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@xentity wrote:

hi there,

may it be possible that someone please dumps his definition of the sql tables:

• ‘asterisk.incoming’
• ‘asterisk.ivr_details’
• ‘asterisk.kvstore_FreePBX_modules_Conferences’

Had some file system issues and the backup was also effected (transferring the backup to another system was an open point… shame on me).
However, I was able to fetch all config files (asterisk is working) and most of the DB tables. But the tables mentioned above were not able to recover.

The following commands

mysqldump --no-data -u root asterisk incoming
mysqldump --no-data -u root asterisk ivr_details
mysqldump --no-data -u root asterisk kvstore_FreePBX_modules_Conferences

should do the trick on a most recent system.

The content of incoming should be restorable by the content of extensions_additional.conf, is that right? I didn’t use IVR nor Conferences, therefore the content of those should be avoidable, right? But the tables are needed for the module updates.

Thx a lot.

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@mudslide567 wrote:

I have a group of phones that are failing to register on the server. In the asterisk logs or realtime asterisk, I see absolutely nothing. However, when I turn on TCP dump, I see the registration request hit the server and a 401 Unauthorized being returned but nothing with the asterisk log.

16:35:43.488317 IP 100.64.17.102.60240 > .sip: SIP: SUBSCRIBE sip:2001@;transport=UDP SIP/2.0
16:35:43.488520 IP .sip > 100.64.17.102.60240: SIP: SIP/2.0 401 Unauthorized

I have checked all extension secrets etc (multiple times) and everything checks out. There are about a dozen phones doing this from one location [100.64.17.102 is their external address]. About 250 other phones are working just fine.

What am I missing?

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