@johntgs wrote:

I am testing conference bridge feature on Freepbx. I am using a cloud server, a Vultr and Freepbxhosting. Both are very basic, minimal instances. My question is how many conferees can I have before I would most likely notice a degradation in quality. I would like to suggest a maximum of 10 participants. I am not certain how I can test on my own beyond the three phones I have. So wondering what your experience has been?

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@glsarto wrote:

Hello All,
I am setting up a new PBX, last distro available, and while I can call, incoming calls crash with the error message above.
I setup the PBX with firewall disabled to minimize initial setup issues, anyway we have one on our LAN.
I chose chan_sip because all the extensions were previously setup with that, setup one trunk, inbound, outbound route and one extension.
Here is the log:
[2020-04-29 18:38:01] VERBOSE[2940][C-0000000b] netsock2.c: Using SIP RTP TOS bits 184
[2020-04-29 18:38:01] VERBOSE[2940][C-0000000b] netsock2.c: Using SIP RTP CoS mark 5
[2020-04-29 18:38:01] VERBOSE[30629][C-0000000b] pbx.c: Executing [s@from-sip-external:1] GotoIf(“SIP/sip.messagenet.it-0000000d”, “1?setlanguage:checkanon”) in new stack
[2020-04-29 18:38:01] VERBOSE[30629][C-0000000b] pbx_builtins.c: Goto (from-sip-external,s,2)
[2020-04-29 18:38:01] VERBOSE[30629][C-0000000b] pbx.c: Executing [s@from-sip-external:2] Set(“SIP/sip.messagenet.it-0000000d”, “CHANNEL(language)=en”) in new stack
[2020-04-29 18:38:01] VERBOSE[30629][C-0000000b] pbx.c: Executing [s@from-sip-external:3] GotoIf(“SIP/sip.messagenet.it-0000000d”, “1?noanonymous”) in new stack
[2020-04-29 18:38:01] VERBOSE[30629][C-0000000b] pbx_builtins.c: Goto (from-sip-external,s,5)
[2020-04-29 18:38:01] VERBOSE[30629][C-0000000b] pbx.c: Executing [s@from-sip-external:5] Set(“SIP/sip.messagenet.it-0000000d”, “TIMEOUT(absolute)=15”) in new stack
[2020-04-29 18:38:01] VERBOSE[30629][C-0000000b] func_timeout.c: Channel will hangup at 2020-04-29 18:38:16.814 CEST.
[2020-04-29 18:38:01] VERBOSE[30629][C-0000000b] pbx.c: Executing [s@from-sip-external:6] Set(“SIP/sip.messagenet.it-0000000d”, “receveip=recvip”) in new stack
[2020-04-29 18:38:01] VERBOSE[30629][C-0000000b] pbx.c: Executing [s@from-sip-external:7] Log(“SIP/sip.messagenet.it-0000000d”, "WARNING,“Rejecting unknown SIP connection from 212.97.59.76"”) in new stack
[2020-04-29 18:38:01] WARNING[30629][C-0000000b] Ext. s: “Rejecting unknown SIP connection from 212.97.59.76”
[2020-04-29 18:38:01] VERBOSE[30629][C-0000000b] pbx.c: Executing [s@from-sip-external:8] Answer(“SIP/sip.messagenet.it-0000000d”, “”) in new stack
[2020-04-29 18:38:02] VERBOSE[30629][C-0000000b] pbx.c: Executing [s@from-sip-external:9] Wait(“SIP/sip.messagenet.it-0000000d”, “2”) in new stack
[2020-04-29 18:38:04] VERBOSE[30629][C-0000000b] pbx.c: Executing [s@from-sip-external:10] Playback(“SIP/sip.messagenet.it-0000000d”, “ss-noservice”) in new stack
[2020-04-29 18:38:04] VERBOSE[30629][C-0000000b] file.c: <SIP/sip.messagenet.it-0000000d> Playing ‘ss-noservice.slin16’ (language ‘en’)
[2020-04-29 18:38:09] VERBOSE[30629][C-0000000b] pbx.c: Executing [s@from-sip-external:11] PlayTones(“SIP/sip.messagenet.it-0000000d”, “congestion”) in new stack
[2020-04-29 18:38:09] VERBOSE[30629][C-0000000b] pbx.c: Executing [s@from-sip-external:12] Congestion(“SIP/sip.messagenet.it-0000000d”, “5”) in new stack
I must be missing something here…
Thanks for your time,
-Gian

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@Benn5325 wrote:

We have a odd issue with some FMFM extensions

Example
Extension 123 which has a DID of 123-456-7890
FMFM is set to go that that users cell phone 987-654-3210

A caller from outside or inside the company calls 987-654-3210
That user gets the call on their cell no problem

A caller from inside the company dials Ext 123
The call immediately goes to the Cellphone voicemail and the cell does not ring at all.
Anyone ever had this issue before

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@dan_ce wrote:

I’m trying to edit my dialplan so that BOTH the incoming caller’s number AND name are sent as an MQTT message. At the moment no matter whether I use CALLERID(name) or CALLERID(all), I just get the phone number, HOWEVER…Asterisk itself is well aware of the callerID name, but I think it’s set many commands AFTER the mosquitto trysystem is done.

[2020-04-29 21:37:43] VERBOSE[26210][C-0000001b] pbx.c: Executing [1@disa:9] Set("SIP/SIPGATE-00000017", "CALLERID(all)="Dave Mobile"") in new stack

–

[from-trunk]
exten => _X.,1,trysystem(mosquitto_pub -h 192.168.2.110 -t 'calls/in/${EXTEN}' -m '${CALLERID(num)}' )
exten => _X.,n,trysystem(mosquitto_pub -h 192.168.2.110 -t 'calls/name≈/${EXTEN}' -m '${CALLERID(name)}' )
exten => _X.,n,Goto(from-pstn,${EXTEN},1)

thanks for any insights!

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@GrandPoobah1 wrote:

Hey all,

I’m currently using FreePBX v15.0.16.49 and am running into an issue where my incoming chan SIP trunk status remains “unknown” between to PBX servers. I’m fairly new and have been trying to find documentation on the configurable parameters & their meaning, but am currently at a loss.

The outbound Trunk group remains online from both servers and my current configurations are as follows:

PBX server 1 = 172.154.10.20

( outbound trunk name= server1

username=server1
fromuser=server1
type=peer
secret=joy444
qualify=yes
nat=no
host=172.54.20.20
dtmfmode=2833
context=from-trunk
allow=ulaw&alaw&g729outgoing

( incoming trunk name= server2)

type=user
secret=joy444
qualify=yes
port=5060
nat=no
insecure=very
host=172.154.20.30
dtmfmode=2833
context=from-trunk
allow=ulaw&alaw&g729

PBX server 2 = 172.154.20.20

( outbound trunk name= server2)

username=server2
fromuser=server2
type=peer
secret=joy444
qualify=yes
nat=no
host=172.54.10.20
dtmfmode=2833
context=from-trunk
allow=ulaw&alaw&g729outgoing

( incoming trunk name= server1)
type=user
secret=joy444
qualify=yes
port=5060
nat=no
insecure=very
host=172.154.10.30
dtmfmode=2833
context=from-trunk
allow=ulaw&alaw&g729

Any assistance would be greatly appreciated and a any information or user guides regarding trunk configuration parameters would be helpful.

Thanks

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@nsumner wrote:

I know this has been a long time coming. But since FreePBX had their blog post pushing everyone to PJSIP. I though maybe with the office being closed right now this is the perfect time to transition to PJSIP. But of course as the saying goes the devil is in the details.

The phones shouldn’t be too hard (some offsite phones might be a pain but even so that is a small number). What does seem to be an issue is trunks. I primarily have 3 different trunks.

1: Partner Israel who I’m probably the first to experiment with but the only problem I know I’ll face is that I must make PJSIP use port 5060 or it won’t work with them. They for some reason can’t work if you are on any other port.

2: VOIPInnovations. You would think now being owned by Sangoma they might have instructions for PJSIP but alas they don’t. Their guide for FreePBX still instructs you to use Chan_SIP.

3: voip.ms. Again I find a couple posts but nothing major and apparently they won’t support you if you are using PJSIP.

So therefore my question is are we really ready to move over to PJSIP. Since at the end of the day I need my providers to be 100% ready before I change the ports around.

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@Infinit0s wrote:

Hello everyone,

As per the title, I am having issues with the Firewall module on my FreePBX installation. It keeps on crashing, yet the Dashboard shows the status of Firewall as ‘online’

The log file in /tmp/ shows the following error message:

rfw rule 3 not valid (Is '-m recent --rcheck --seconds 86400 --hitcount 1 --name ATTACKER --mask ffff:ffff:ffff:ffff:ffff:ffff:ffff:ffff --rsource -j fpbxattacker', should start with '-m recent --rcheck --seconds 10 --hitcount 50 --name REPEAT --rsource')
THIS MAY BE A KERNEL ISSUE. IF THIS KEEPS OCCURRING REBOOT YOUR MACHINE URGENTLY.
1588197287: Wall: 'Firewall Rules corrupted! Restarting in 5 seconds
More information available in /tmp/firewall.log
' returned 0
Redirecting to /bin/systemctl stop fail2ban.service
Starting firewall.
1588197305: Monitoring parent (voipfirewalld) died. Shutting down!

I can see that most likely this fails because of the ‘error’ message up in the log which reads:

1588197251: /sbin/ip6tables -w5 -W10000 -A fpbxratelimit -m recent --rcheck --seconds 86400 --hitcount 200 --name REPEAT --rsource -j fpbxattacker
ip6tables: Invalid argument. Run `dmesg' for more information.
1588197251: /sbin/iptables -w5 -W10000 -A fpbxratelimit -m recent --rcheck --seconds 86400 --hitcount 200 --name REPEAT --rsource -j fpbxattacker
iptables: Invalid argument. Run `dmesg' for more information.
1588197251: /sbin/ip6tables -w5 -W10000 -A fpbxratelimit -m recent --rcheck --seconds 300 --hitcount 100 --name REPEAT --rsource -j fpbxattacker
ip6tables: Invalid argument. Run `dmesg' for more information.
1588197251: /sbin/iptables -w5 -W10000 -A fpbxratelimit -m recent --rcheck --seconds 300 --hitcount 100 --name REPEAT --rsource -j fpbxattacker
iptables: Invalid argument. Run `dmesg' for more information.
1588197251: /sbin/ip6tables -w5 -W10000 -A fpbxratelimit -m recent --rcheck --seconds 60 --hitcount 50 --name REPEAT --rsource -j fpbxshortblock
ip6tables: Invalid argument. Run `dmesg' for more information.
1588197251: /sbin/iptables -w5 -W10000 -A fpbxratelimit -m recent --rcheck --seconds 60 --hitcount 50 --name REPEAT --rsource -j fpbxshortblock
iptables: Invalid argument. Run `dmesg' for more information.

I have been doing some digging to ensure that everything is installed/enabled.

Iptables version:
[root]# iptables --version
iptables v1.4.21

ipt_recent or xt_recent loaded:
[root]# lsmod | grep xt_recent
xt_recent 4242 -2

FreePBX version:
FreePBX 15.0.16.49

Linux version:
[root]# cat /etc/os-release
NAME=“CentOS Linux”
VERSION=“7 (Core)”
ID=“centos”
ID_LIKE=“rhel fedora”
VERSION_ID=“7”
PRETTY_NAME=“CentOS Linux 7 (Core)”
ANSI_COLOR=“0;31”

CENTOS_MANTISBT_PROJECT="CentOS-7"
CENTOS_MANTISBT_PROJECT_VERSION="7"
REDHAT_SUPPORT_PRODUCT="centos"
REDHAT_SUPPORT_PRODUCT_VERSION="7"

Kernel version:
[root]# uname -r
3.10.0

The system runs on OpenVZ 7.

Unfortunately, I have run out of ideas on what could cause this - is it a bug?

Thanks for looking!
Maciej

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@roney019 wrote:

My Freepbx Paging and intercom is not working properly as i tried to use it.When i tried to dial *80 then extension number for intercom calls then calls are ended.I found that i dont have the Paging and intercom option in Feature code page but in my other freepbx box intercom is working fine and i have the paging and intercom section on Feature code page.

Can you help me on that how can i get working paging intercom feature code?

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@peakpbx wrote:

Hi. I am having an issue with my freepbx install

Have been using my installation with just 1 trunk with 1 DID in it… it works all well with CFC, Time conditions ecc…

I recently added 3 more DIDs for direct to extension calling. I added inbound routes for all numbers to go to all respective extensions…

What is happening is… any number i call… my main number or other 3 new numbers… they still all go to the destination of my mian line CFC>TC>IVR…

If i change destination of main number all numbers go to that destination…

In my Sip Trunk Portal all numbers are added correctly as aliases of the number so free pbx should be doing the work of picking out what number is being dailed and sending to correct destination…

Am i missing something please… maybe in trunk incoming settings… as i only have register string… maybe in other fields to add the other numbers or something.
Thanks

Ryan

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@HELSup wrote:

Hi all!

My FreePBX 15 is up and running behind a Fritz!Box via NAT.
Everything worked fine for some time but not I found that my external “static” IP is not really static and it may be changed.
So I can’t no longer use in SIP Settings -> General SIP Settings -> NAT Settings a fixed “External Address” via “Detect Network settings”.

I want to use our own web server to get my external IP address and no external service.
For example

wget http://some.web.address/cgi-bin/addr.cgi -q -O - | awk ‘/REMOTE_ADDR/ {print$2}’

I looked at the documentation and the Forums but I found no information how to use such a setup.

Any clue?

TIA,
HS

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@kmgsam wrote:

FreePBX 15.0.16.49

I am having trouble understanding how groups work.

I tired to create a group that has access to s shared mailbox for example our evening mailbox that needs to be cleared out each morning. I then assigned a few users to the group for the evening mailbox plus they were in the all users group that i setup. These users are now members of 2 groups. It is only processing the settings from the highest priority group.

I am using LDAP and I have imported my users but doing local group management. I then created several groups. The system allows me to assign a user to more then one group. I have it setup to manage groups locally. in the LDAP settings.

Also if i made a change to an individual user settings and click save it removes them from any groups they are a member of when I click save on the user then i have to go back to the group and add them back to the group. Is this normal?

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@glsarto wrote:

Hello All,
I am setting up FreePBX for a small office that actually runs two different business.
So I setup two Trunks and two Outbound Routes, but how can I divide incoming traffic?
I guess I should setup two Inbound Routes, but I do not really know what to enter…
Thanks for reading,
-G

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@glsarto wrote:

Hello All,
I am registering (chan_sip) some Gigaset and some Grandstream phones.
I am having a very hard time with the Grandstream, they just don’t want to.
If I ping the PBX from the phone, it says no answer, any other host on the LAN will answer.
I managed to register just one, and I notice that the port keeps changing, and is not 5060:
[2020-04-30 09:47:06] VERBOSE[2940] chan_sip.c: Registered SIP ‘12’ at 192.168.1.85:31307
[2020-04-30 12:04:02] VERBOSE[2692] chan_sip.c: Registered SIP ‘12’ at 192.168.1.85:16906
[2020-04-30 14:09:08] VERBOSE[2692] chan_sip.c: Registered SIP ‘12’ at 192.168.1.85:15636
[2020-04-30 14:21:20] VERBOSE[2692] chan_sip.c: Registered SIP ‘12’ at 192.168.1.85:23768
[2020-04-30 14:27:08] VERBOSE[2692] chan_sip.c: Registered SIP ‘12’ at 192.168.1.85:36039
[2020-04-30 14:36:29] VERBOSE[2828] chan_sip.c: Registered SIP ‘12’ at 192.168.1.85:24108
[2020-04-30 15:38:45] VERBOSE[2828] chan_sip.c: Registered SIP ‘12’ at 192.168.1.85:13188
[2020-04-30 15:39:52] VERBOSE[2828] chan_sip.c: Registered SIP ‘12’ at 192.168.1.85:13188
[2020-04-30 16:09:56] VERBOSE[2828] chan_sip.c: Registered SIP ‘12’ at 192.168.1.85:17331
[2020-04-30 16:39:05] VERBOSE[2828] chan_sip.c: Registered SIP ‘31’ at 192.168.1.77:46722
[2020-04-30 16:42:36] VERBOSE[2828] chan_sip.c: Registered SIP ‘15’ at 192.168.1.23:5060
[2020-04-30 17:10:14] VERBOSE[2828] chan_sip.c: Registered SIP ‘35’ at 192.168.1.48:5060
[2020-04-30 17:32:16] VERBOSE[2828] chan_sip.c: Registered SIP ‘18’ at 192.168.1.26:5060

The last three are Gigasets, they show port 5060, and they register in a click.

In the Grandstream account setup, SIP/Basic/Settings, Local Sip Port is 5060.

I must be missing something obvious…

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@mroberts12345 wrote:

Hi everyone.

Trying to program a softbutton on an EXT board to set the phone to DND (*78) and to forward the calls (*72XXXXXXXXX).

I am trying to make this easy for my users and have just one button be pushed. Anyone know how to accomplish this? I tried a comma between the 2 commands and it still only did the first command and then stopped.

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@we7u wrote:

I followed this Wiki page to configure my Obi110 as an ATA for FreePBX:

https://wiki.freepbx.org/pages/viewpage.action?pageId=4161592

That page doesn’t appear to secure against people coming in on the Obi-to-Obi number. I believe someone came in through this a couple of times last night to my FreePBX to sniff around. They didn’t come in over my DID, and the Obi-to-Obi number is the only way I can imagine they could have come in.

Can someone add instructions to close down the Obi-to-Obi calling method to the above Wiki page, plus detail them here?

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@nightbreed wrote:

I have a customer’s office set up to forward calls to their main number to a cell phone via a Misc Destination.
They are telling me that callers are trying to call their main number and getting a “this phone has been disconnected” message sometimes. One caller got this message three times, and got through the 4th time he tried calling.
I’m not really sure where to troubleshoot this problem? Any guidance would be greatly appreciated.

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@trb037 wrote:

Greetings! Long Time Listener First Time Poster…

As a hobbyist I have deployed FreePBX and registered with Twilio for an Elastic Trunk.

I followed a setup guide for integrating the two (link below if you’re curious) but any test calls (in or out) result in “number not in service” messages. The calls appear in my logs, but for some reason I cannot seem to route properly.

I am happy to provide any logs necessary to assist in troubleshooting, and I am grateful in advanced for any assistance.

Thanks!
Thomas

Setup Guide:

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@ericross04 wrote:

This is a very minor issue.

We’ve moved our conference bridges from another service to FreePBX and it works well. We have an IVR set up for all callers (internal and external) to enter the conference bridge number. The part that confuses people off is that other services have you enter a number and # after it to enter the conference bridge (ex: 1234#). I’ve looked all over at settings and can’t find anything that does that.

Is there a way to do the same thing in FreePBX?

Thanks.

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@asianman wrote:

I have two FreePBX boxes peered together with a SIP trunk but call recording does not work. Can someone direct me to how I might enable this?

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@PitzKey wrote:

Hi guys,

I know this potion can be done with an outbound route and a loop back Trunk. But it’s hard to have the techs remember to keep on adding DIDs to the outbound route as well every time they add an inbound route, so I was thinking of the following:

[macro-dialout-trunk-predial-hook]
exten => s,1,Noop()
exten => s,n,Gotoif($["${DIALPLAN_EXISTS(ext-did-XXXX,${OUTNUM:-10},1)}" = "1"]?ext-did-XXXX,${OUTNUM:-10},1)
exten => s,n,MacroExit

But this does not work. It only works if I specify the DID context, like:

exten => s,n,Gotoif($["${DIALPLAN_EXISTS(ext-did-0012,${OUTNUM:-10},1)}" = "1"]?ext-did-0012,${OUTNUM:-10},1)

Is there a way to make this work, or I need to specify the context for each DID?

Another question:
When using a predial hook with something like

exten => s,n,System(/tmp/script.sh)

And lets say the script tries to reach a server, or sends an email. Is there a possibility for the call to continue in case the script gets stuck? I don’t want calls to get stuck because of that script…
(Running System() asynchronous ?)

Thanks

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