@edlentz wrote:

I want to disable/enable ALL trunks using fwconsole trunk commands. I have tried fwconsole trunk disable all and fwconsole trunk enable all with no results. Is there a command to do this? Or even if I have to do the trunk individually that is OK too.

Thanks

Posts: 4

Participants: 2

Read full topic

@admartz wrote:

Hello,

I am in the process of setting up the initial configuration of our new FreePBX Phone System 60 running Distro 13.0.195.13

We have 18 PJSIP extensions set up all feeding calls to Grandstream GXP2160 phones. We also have 6 incoming POTS lines connected to 8-port FXO analog card (2 unused for future growth). The extensions appear to register properly with the phones and we can currently make internal calls between the extensions.

The problems we are currently experiencing are as follows:

-Though I have used the dial pattern wizard to set ‘Dialed Number Manipulation Rules’ for an outbound trunk as well as the dial patterns for ‘Outbound Routes,’ we still have to dial a 1 when placing outbound calls.

-If I do call an outbound number the call is dropped after 25 seconds consistently

-If an inbound call is answered and the call is ended in office, the connection to the outside phone is sustained for several minutes.

I am very much a novice when it comes to phone systems. I have done all configuration in the system’s web UI and have primarily been using a combination of the FreePBX documentation, mirroring some settings from our currently working, but much older, Trixbox system and the FreePBX forums to get me to this point.

Any help would be very much appreciated and I am happy to retrieve whatever logs or config files may be necessary to help approach these issues.

Thank you for your time.

Posts: 1

Participants: 1

Read full topic

@TimCavman wrote:

Hi There,

so were using Snom 821’s with Freepbx and ive been tasked to provide a solution with internal and external ring tones, i’ve modified the XML for ringer alert-text external/internal but unfortunately no joy. There are options showing “VIP, Work,Family” etc within the phones themselves but unsure on how to configure and mass provision these settings so different ringers alert the users to where the call is coming from.

Any ideas would be great - Thanks

Posts: 1

Participants: 1

Read full topic

@PCS wrote:

Hi all,

I’m running 6.12.65-32 with Parking 13.0.19.18, and EPM 13.0.97.9. I’m seeing unexpected behavior with park BLFs with my Polycom VVX 400 phones.

The lot is set up at 70, with 3 slots. The Polycom button “Park” is BLF-XFER 70, and the lots themselves are BLF 71, etc.

When parking a call, both the Park and the Lot 71 BLFs are lit. I’m not expecting the Park BLF to be active, since the call was paced in the lot.

Is this normal behavior? We usually put the Park button on a horizontal key, which obviously doesn’t have a lamp, but in this case the client wanted the Park button on the BLF soft keys.

Thanks.

Posts: 2

Participants: 2

Read full topic

@GSnover wrote:

https://www.bloomberg.com/news/features/2018-10-04/the-big-hack-how-china-used-a-tiny-chip-to-infiltrate-america-s-top-companies

I wonder if this has anything to do with it…

Posts: 3

Participants: 3

Read full topic

@Spyder13337 wrote:

Good Morning,

i have a Yealink t52s

when i recived an inbound call i need to transfer that call but i need ti show caller ID of the originating caller not mine but

my yea link firmware is

Firmware Version 70.84.0.10
Hardware Version 74.0.0.0.0.0.0

i am testing on elastix 4.0 and asterisk 11.25.3

i have the trustrpid = yes
and sendrpid = i tried both send p=asserted and send remote party ID

and set up pai-prid-from on the yealink

but still no luck

any help much appreciated

Posts: 1

Participants: 1

Read full topic

@fewpcs wrote:

I am new to FreePBX and have been setting up a new system.
I am not positive where to find out why voipfirewalld is using so much CPU.
I’ve tried rebooting the system and the CPU is still high.

ps output:
root 1834 87.5 0.1 395480 1320 ? R 15:04 57:47 voipfirewalld (Monitor thread)

top output:
PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND
1834 root 20 0 395480 1320 412 R 93.8 0.1 56:36.27 php

Thanks.

Posts: 1

Participants: 1

Read full topic

@ELindemann wrote:

Hello @all,

i have been reading a lot of helps/hints in this forum, all other places in the www.
But none of a solution fits for my problem.

I compiled a on

$lsb_release

Distributor ID: Debian
Description: Debian GNU/Linux 8.10 (jessie)
Release: 8.10
Codename: jessie

$uname -a
Linux XXXX 3.16.0-4-amd64 #1 SMP Debian 3.16.51-3 (2017-12-13) x86_64 GNU/Linux

$asterisk -rvvvv

Asterisk certified/13.21-cert2, Copyright © 1999 - 2014, Digium, Inc. and others.
---- no links allowed for new users
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.

Running as user ‘asterisk’
Running under group ‘asterisk’
Connected to Asterisk certified/13.21-cert2 currently running on XXXX (pid = 6468)

As you see, not running as root.

###########################
/etc/asterisk/manager.conf

[general]
enabled = yes
port = 5038
bindaddr = 0.0.0.0
displayconnects=no ;only effects 1.6+

[admin]
secret =b56f82085735733e7a850e8893084ea0
deny=0.0.0.0/0.0.0.0
permit=127.0.0.1/255.255.255.0
permit=192.168.XXX.0/255.255.255.0
read = system,call,log,verbose,command,agent,user,config,command,dtmf,reporting,cdr,dialplan,originate,message
write = system,call,log,verbose,command,agent,user,config,command,dtmf,reporting,cdr,dialplan,originate,message
writetimeout = 5000

;include manager_additional.conf
;include manager_custom.conf
###########################

/etc/amportal.conf

Asterisk Manager Password

Default Value: amp111

AMPMGRPASS=b56f82085735733e7a850e8893084ea0

Asterisk Manager User

Default Value: admin

AMPMGRUSER=admin

Asterisk Manager Host

Default Value: localhost

ASTMANAGERHOST=localhost

Asterisk Manager Port

Default Value: 5038

ASTMANAGERPORT=5038
###########################

/etc/freepbx.conf

<?php $amp_conf['AMPDBUSER'] = 'freepbxuser'; $amp_conf['AMPDBPASS'] = 'ec9be7540d0b79f14142d271b27fc151'; $amp_conf['AMPDBHOST'] = 'localhost'; $amp_conf['AMPDBNAME'] = 'asterisk'; $amp_conf['AMPDBENGINE'] = 'mysql'; $amp_conf['datasource'] = ''; //for sqlite3 require_once('/var/www/html/admin/bootstrap.php'); ?>

Starting with
$fwconsole -vvv start

delivers
=>
…
…
Setting /bin/mydir to permissions of: 755
Setting /var/www/html/admin/modules/core/agi-bin/list-item-remove.php user owner to: asterisk
Setting /var/www/html/admin/modules/core/agi-bin user owner to: asterisk
Setting /var/www/html/admin/modules/recordings/agi-bin to permissions of: 755
Setting /var/www/html/admin/modules/recordings/agi-bin/recordings.agi to permissions of: 755
Setting /var/www/html/admin/modules/recordings/agi-bin/recordings.agi user owner to: asterisk
Setting /var/www/html/admin/modules/recordings/agi-bin user owner to: asterisk
Setting /etc/foo.bar to permissions of: 644
Setting /etc/mydir to permissions of: 755
Setting /tmp/foo/ to permissions of: 644
Setting /bin/mydir to permissions of: 755

Finished setting permissions
Starting Asterisk…
[------------->--------------] 41 mins
[-------------------->-------] 48 mins

I also did:

$mysql> select * from admin where variable = ‘version’;
±---------±--------+
| variable | value |
±---------±--------+
| version | 13.0.60 |
±---------±--------+
1 row in set (0.00 sec)

and

$fwconsole ma list | grep framework

| framework | 13.0.195.4 | Aktiviert | GPLv2+

and
netstat -tulpen | grep asterisk
=>
tcp 0 0 0.0.0.0:5038 0.0.0.0:* LISTEN 130 32447 6468/asterisk
udp 0 0 0.0.0.0:5060 0.0.0.0:* 130 32457 6468/asterisk
udp 0 0 0.0.0.0:52865 0.0.0.0:* 130 33002 6468/asterisk
udp6 0 0 :::57590 :::* 130 33003 6468/asterisk

lsoflisten
=>

apache2/:443/TCP
apache2/:8080/TCP
apache2/:80/TCP
apache2/:8443/TCP
asterisk/:5038/TCP
cupsd/127.0.0.1:631/TCP
cupsd/[::1]:631/TCP
master/[::1]:25/TCP
master/127.0.0.1:25/TCP
mysqld/127.0.0.1:3306/TCP
postgres/127.0.0.1:5432/TCP
postgres/[::1]:5432/TCP
rpcbind/:111/TCP
rpc.statd/:33027/TCP
rpc.statd/:41669/TCP
sshd/:22/TCP
xrdp/:3390/TCP
xrdp-sesm/127.0.0.1:3350/TCP
(---- no links allowed for new users =>, therefor no ., only _)
nslookup dns_quad9_net
=>
Server: 192.168.XXX.4
Address: 192.168.XXX.4#53

Non-authoritative answer:
(---- no links allowed for new users =>, therefor no ., only )
Name: dns_quad9_net
Address: 149.112.112.112
Name: dn.quad9_net
Address: 9.9.9.9

also working.

Freepbx-Webgui ist very slow, one click triggers a long time nothing, then after this long time the gui delivers something, but (cause of no connect), it stands for it self.

/var/log/asterisk/freepbx.log
=>
…
[2018-Aug-22 03:03:42] [CRITICAL] (admin/bootstrap.php:258) - Connection attmempt to AMI failed
[2018-Aug-22 03:05:19] [CRITICAL] (admin/bootstrap.php:258) - Connection attmempt to AMI failed

I do not have any other logs; was looking for them with
$locate asterisk| grep .log
also for freepbx.

This machine is a kvm-machine on (also) debian without any hw-cards needed, only Lan/VOIP, no analog stuff, no (german) ISDN, pure voip, nothing complicated.

Do not know, what to do. Nope. Appreciate any help.

What else could i deliver for any appreciated help to solve this problem, that FreepbxGUI can not connect to Asterisk, although asterisk is running.

Thanks in advance
ELindemann

Posts: 2

Participants: 2

Read full topic

@carlosriosg wrote:

Hello everyone, I’m very new to this FreePBX 14.0.3.19, I have it installed in my office on a Raspberry Pi and it works very well.

I have 2 SIP accounts and 2 annexes

I want to leave an attachment when making a call use a SIP account and the other annex that uses the other SIP account, now the two annexes leave for the same SIP account

How can that rule be established?

I hope you can help me

Thank you

Posts: 1

Participants: 1

Read full topic

@MichaelCollis01 wrote:

Good Afternoon All,

We are have a strange issue with Our FreePBX configuration, We are using a combination of SIP URI and Sip Trunks, We use the SIP URI on for phone numbers to be used as DID’s and the route into the phone system the phone rings like normal and will kick into voicemail but then will not display onto the users phone I have contacted the phone provider to see if they said that they send all the calls direct to our IP Address,

I ran a sip trace on the call to see if i could see anything strange but i’m not able to, i also sent the same logs to the telephone provider but they said it was out of their scope to even look at the trace.

The Voicemails work internally fine and also work when you call the extensions through the IVR but does not seem to work using URI, i must say this has worked in the past but recently stopped working, i have checked all the settings i can see possible and also updated the system, On the trace i can see the phone is ringing and the system has passed it off to the voicemail where it has created a temp recording but after that it never displays on the system.

I am unable to put a copy of the sip trace on here, Please contact me and that can be shared

Thanks ,

Posts: 2

Participants: 2

Read full topic

@wellington103 wrote:

cdr (Cur v. 2.11.0.11) should be upgraded to v. 2.11.0.12 to fix security issues: SEC-2015-001 Added 13 horas, 32 minutos ago (freepbx.VULNERABILITIES)

Solução:

para resolver atualizei dos módulos CDR Reports e o módulo Print Extensions

Posts: 2

Participants: 2

Read full topic

@darrenhollick wrote:

We have Sangoma’s commercial “Queue Plus” product (in case that matters) and we’ve been trying for over a year now to get auto pause working correctly. There are two issues that we’re aware of and we were able to work around the one but I don’t know what to do about the other.

FreePBX Firmware: 10.13.66-22
Queues Module: 13.0.34.10

First Issue

The first, and more serious, issue is that when we have a queue that is set to auto pause members and when all the members are busy and another call enters the queue it systematically (and instantly) pauses every single member that is logged in to that queue. The desired behavior is that it should only auto-pause a member if they fail to answer a call when they are not on an existing call. I have played with with every combination of settings I can think of but I cannot get this to function as desired. Here are my queue settings and what I think should work but I’m hoping that someone can either confirm this as a bug or tell what combination of settings makes this work.

Queue Settings

[7200]
announce-frequency=120
announce-holdtime=once
announce-position=yes
autofill=yes
autopause=yes
autopausebusy=no
autopausedelay=0
autopauseunavail=no
joinempty=yes
leavewhenempty=loose
maxlen=0
memberdelay=0
min-announce-frequency=15
penaltymemberslimit=0
periodic-announce-frequency=90
queue-callswaiting=queue-callswaiting
queue-thankyou=queue-thankyou
queue-thereare=queue-thereare
queue-youarenext=queue-youarenext
reportholdtime=yes
retry=0
ringinuse=no
servicelevel=90
strategy=leastrecent
timeout=20
timeoutpriority=app
timeoutrestart=no
weight=0
wrapuptime=30
context=queuecallback-2
periodic-announce=custom/Esc_Queue_Callback_Offer_
lazymembers=no

Second Issue

The second issue we were able to work around is that the FreePBX feature code to toggle a member’s pause status doesn’t work if they are the member that should receive a call next. For example, an agent is paused for any reason and they want to unpause themselves. However, as they dial the feature code *46*EXT*QUEUE it unpauses them and the queue can immediately try to send them a call which it reports as busy and thus it auto pauses them again. The simple work around (and I propose as a fix) for this is to change the [app-queue-pause-toggle] so that it doesn’t set the agent as unpaused until they call us hung up. So, change this…

Current Dialplan

[app-queue-pause-toggle]
include => app-queue-pause-toggle-custom
exten => s,1(start),Answer
exten => s,n,Wait(1)
exten => s,n,Macro(user-callerid,)
exten => s,n,Set(QUEUEUSER=${IF($[${LEN(${ARG2})}>0]?${ARG2}:${AMPUSER})})
exten => s,n,Set(MEMBR=Local/${QUEUEUSER}@from-queue/n)
exten => s,n,Set(PAUSE_STATE=${QUEUE_MEMBER(${ARG1},paused,${MEMBR})})
exten => s,n,Set(QUEUE_MEMBER(${ARG1},paused,${MEMBR})=${IF($[${PAUSE_STATE}]?0:1)})
exten => s,n,Playback(dictate/pause&${IF($[${PAUSE_STATE}]?de-activated:activated)})
exten => s,n,ExecIf($[${ARG2}]?Return())
exten => s,n,Macro(hangupcall,)

to this…

Fixed Dialplan

[app-queue-pause-toggle]
include => app-queue-pause-toggle-custom
exten => s,1(start),Answer
exten => s,n,Wait(1)
exten => s,n,Macro(user-callerid,)
exten => s,n,Set(QUEUEUSER=${IF($[${LEN(${ARG2})}>0]?${ARG2}:${AMPUSER})})
exten => s,n,Set(MEMBR=Local/${QUEUEUSER}@from-queue/n)
exten => s,n,Set(PAUSE_STATE=${QUEUE_MEMBER(${ARG1},paused,${MEMBR})})
exten => s,n,Playback(dictate/pause&${IF($[${PAUSE_STATE}]?de-activated:activated)})
exten => s,n,Macro(hangupcall,)

exten => h,1,Set(QUEUE_MEMBER(${ARG1},paused,${MEMBR})=${IF($[${PAUSE_STATE}]?0:1)})

Ultimately this issue may actually be related to the first issue since the problem is that the agent is getting paused because the queue is trying to send them a call while they are on a call and pausing them. Thus, the answer to the first problem maybe an answer to both. But I still think that the proposed change to resolve my second issue is more reliable fix.

Posts: 1

Participants: 1

Read full topic

@Beachtech wrote:

Is there a device (not a phone) that I can put on my client’s FreePBX 14 system that is just a giant BLF light? I want something very conspicuous to light up so my client knows that it is in after hours mode (and he can manually override if necessary)? I have a red BLF set up on one of the DSS keys of the Yealink T28 phone, but they never see it.

Posts: 3

Participants: 3

Read full topic

@toborgps wrote:

Hi, I now have 3 Cisco phones on my system the 3rd one just added today 1 that runs Chan_SIP (Local box network) and 2 remote one in MN which happily registered with PJSIP and one in NV which is the new one, we set this one up the same how we did in MN and it will sort of connect by telling the system it is unknown. It is unreachable to call out or take calls, I have tried everything and can’t figure out a solution.

Posts: 2

Participants: 2

Read full topic

@shail00 wrote:

Hi Team,

We are using asterisk for our incoming call management and automated voice playback files.
Sometime we have users who dials the IVR number and report that nothing is being played and they hang up the call and then it connects. We tried the call and found out that when the user do not hear anything, in the logs, we are missing following line:

0x7f005c005960 – Probation passed - setting RTP source address to 66.241.99.236:18140

However, when the call is connected, we get this line in the call log.
Can someone please help?

Posts: 2

Participants: 2

Read full topic

@Spaxton wrote:

Dear Friends,

I am having a trouble for days to figure out how to configure sip trunk to register directly to providers IMS. I gave up on further searching and I know that this may be asked often but I am in a dead end.
What I can achieve now is to register it but there is no way I can place a call via that trunk. For incoming calls I get the connection but there is no audio while actually I should hear (and I dont know why but thats what I see in asterisk log files. Asterisk default prompt) “The number You have dialed is not in service. Please check the number and try again”. In the logfiles before playing those promts it stands like this:

[2018-10-07 21:48:24] VERBOSE[1733][C-000004d2] pbx.c: Executing [s@from-sip-external:6] Log(“SIP/ims.telekom.me-000010e9”, "WARNING,“Rejecting unknown SIP connection from 10.179.23.180"”) in new stack

TCPDUMP gives this on registration:

21:48:11.104297 IP (tos 0x60, ttl 64, id 10210, offset 0, flags [none], proto UDP (17), length 718)
10.130.81.40.5160 > 10.179.23.180.5060: [bad udp cksum 0x80dc -> 0x5d83!] SIP, length: 690
REGISTER sip:ims.telekom.me SIP/2.0
Via: SIP/2.0/UDP 10.130.81.40:5160;branch=z9hG4bK5db34251;rport
Max-Forwards: 70
From: sip:+382XXXXXXXX@ims.telekom.me;tag=as7adaf115
To: sip:+382XXXXXXXX@ims.telekom.me
Call-ID: 5be4424715ef0d9113e679be70ac3159@127.0.0.1
CSeq: 103 REGISTER
Supported: replaces, timer
User-Agent: FPBX-14.0.3.18(13.22.0)
Authorization: Digest username="382XXXXXXXX@ims.telekom.me", realm=“ims.telekom.me”, algorithm=MD5, uri=“sip:ims.telekom.me”, nonce=“C2F9B68A0763BA5B000000008DCCAA35”, response=“742a07c6caffdf66d872a347b2a3c2e6”, qop=auth, cnonce=“56d5b7f8”, nc=00000001
Expires: 120
Contact: sip:+382XXXXXXXX@10.130.81.40:5160
Content-Length: 0

21:48:11.224625 IP (tos 0x60, ttl 125, id 0, offset 0, flags [none], proto UDP (17), length 727)
10.179.23.180.5060 > 10.130.81.40.5160: [udp sum ok] SIP, length: 699
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.130.81.40:5160;received=10.130.81.40;branch=z9hG4bK5db34251;rport=5160
From: sip:+382XXXXXXXX@ims.telekom.me;tag=as7adaf115
To: sip:+382XXXXXXXX@ims.telekom.me;tag=8d6a58f005e9ab4720d35c7b98378bc1
Call-ID: 5be4424715ef0d9113e679be70ac3159@127.0.0.1
CSeq: 103 REGISTER
Content-Length: 0
Contact: sip:+382XXXXXXXX@10.130.81.40:5160;expires=1800

The default port for chan_sip is 5160 and I cannot change it to 5060 so in this case the source port is 5160 but it seems it can work.

My chan_sip trunk configuration is like this:

PEER Details:

username=+382XXXXXXXX
type=peer
secret=PASSWORD
qualify=yes
port=5060
outboundproxyport=5060
outboundproxy=10.179.23.180
nat=yes
insecure=invite,port
host=ims.telekom.me
fromdomain=ims.telekom.me
disallow=all
context=from-sip-external
canreinvite=no
authuser=382XXXXXXXX@ims.telekom.me
allow=alaw,ulaw,g729,g722

Register string:

+382XXXXXXXX@ims.telekom.me:PASSWORD:382XXXXXXXX@ims.telekom.me@10.179.23.180/+382XXXXXXXX

Here are the screenshot how it was done in the huawei router given from provider:

voip2.jpg714x154 22.6 KB

voip1.jpg1001x588 91.2 KB

Is there anyone who can help me with this, explain it and clear my mind about it a little bit?

Best Regards.

Posts: 1

Participants: 1

Read full topic

@travis_farmer wrote:

while working on my FreePBX server, i was checking the option to stream music on hold from my Icecast server, and it worked great, though there was no call load, so i don’t know if it is fully stable. but still, it worked.
now, if i was to plug in a cheap USB sound card, would it be possible to take the audio input, and apply it directly to a MoH category?
i don’t have a problem using wav audio files, but as i like to tinker with things, i thought i would probe the question. seems like it would be a neat feature.

~Travis

Posts: 4

Participants: 3

Read full topic

@EnergyAliance wrote:

Hi
No activation after reboot again
No hardware change
Hardware lock cannot be reset, maximum attempts reached
Please help.
Need activate system…
What email from the technical support?

Posts: 2

Participants: 2

Read full topic

@psqualo wrote:

Hello community,
i would like to know, if i can disable all not needed “feature code” without affect the core freepbx dialplans; in otherwords feature codes are only related to user extension “dials function” or are globally relevent? i don’t need all these enabled by default.
Thanks guys.

Posts: 1

Participants: 1

Read full topic

@psqualo wrote:

Hello community,
i would like to know, if i can disable all the user settings (DND, Call Forward ecc…) widgets on the UPC user page. I need only a page with the voicemail widget and nothing else. I have also see in the User Management all the UPC settings for the selected user, but in Micellaneus tab -> “Allowed Extension Settings” is not present an option for disable it completely.

Posts: 1

Participants: 1

Read full topic