@yeasindsi wrote:

I am using freePBX on an EC2 Instance. I am using Freepbx 13. I haven't configured any sip trunking yet. All I am doing is, Created few extensions & configured them on soft phones (3cx, zoiper & xlite). I can make successful calls among the extensions while using the softphones. But when I try to make call from UCP WebPhone, the soft phones are getting the call but with 2 problems:

1. The call is dropping after 30 sec
2. No voice is heard from any of the sides.

But that same extension is functional while using the soft phones. I have tried using Firefox, Chrome. I have SSL certificate configured as well.

Can some one please guide me what are check points I need to check inorder to make this webPhone call working. Thanks In advance

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@Username4488 wrote:

Hi
I installed the OS/ Software on my Virtual Pc and it Works.
The Wiki here is down what I see.
So how can I get ~3 IP Phones to work just for a an Internal "Demo".
For your Home I need just an Internal System to call my Granny and see if she is rite or not.
Thanks

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@Tanakay wrote:

Hello, I am trying to implement a python script to react to channel states and initiate system actions. I have followed the Asterisk tutorials but I can't seem to figure where to drop my script... or if that's a non-issue, why the ARI module cannot load.

Thanks and regards.

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@MSchub wrote:

Trying to use a Cisco 2851 ISR to be the PRI-to-SIP interface for my new FreePBX installation. Could someone who has done this please take a look at my setup and tell me if something is obviously wrong? Here is the router config:

Current configuration : 4112 bytes
!
! Last configuration change at 20:17:47 GMT Wed Sep 13 2017 by XX_admin
!
version 15.1
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname phonerouter
!
boot-start-marker
boot system flash:c2800nm-adventerprisek9_sna-mz.151-3.T4.bin
boot-end-marker
!
!
logging buffered 51200 warnings
!
no aaa new-model
!
clock timezone GMT -4 0
!
dot11 syslog
dot11 phone
ip source-route
!
!
ip cef
!
!
ip domain name xxx.local
ip name-server 192.168.0.20
ip name-server 192.168.0.21
no ipv6 cef
!
multilink bundle-name authenticated
!
!
voice rtp send-recv
!
voice service voip
 allow-connections h323 to h323
 allow-connections h323 to sip
 allow-connections sip to h323
 allow-connections sip to sip
 redirect ip2ip
 signaling forward unconditional
sip
  bind control source-interface GigabitEthernet0/1
  bind media source-interface GigabitEthernet0/1
!
!
voice-card 0
 dspfarm
!
crypto pki token default removal timeout 0
!
crypto pki trustpoint TP-self-signed-838144
 enrollment selfsigned
 subject-name cn=IOS-Self-Signed-Certificate-838144
 revocation-check none
!
!
crypto pki certificate chain TP-self-signed-838144
 certificate self-signed 01
  30820227 30820190 A0030201 02020101 300D0609 2A864886 F70D0101 04050030
  ...
  F70D0101 04050003 8181004E 23883DD9 AA8320E4 5BDB9470 4E4B3BEC DAC313D9
  7835005D 37353FBC 77DC5CB9 7AE15796 2A05EC40 8F200F3C DF09F22D 76E49294
  A3A55790 2E150E7B 35EBAE4E 214BFAC1 B9EB0170 2BA26E0A 03C0E10F 420CA910
  2213413E 0D94ED34 BD8D8F5E 642E02C1 52120FFD B9BCD955 BD1BD5C6 CC8A601B
  5B05103C F71325AD B48AA6
    quit
!
!
license udi pid CISCO2851 sn FTX124XXX4Q
username XX_admin privilege 15 secret 5 $1$XGR3$zjan13hpDseun3DCYRonw0
!
redundancy
!
!
interface GigabitEthernet0/0
 no ip address
 duplex auto
 speed auto
!
interface GigabitEthernet0/1
 ip address 192.168.0.223 255.255.255.0
 duplex auto
 speed auto
!
interface Serial0/0/0
 no ip address
!
!
ip forward-protocol nd
ip http server
ip http access-class 23
ip http authentication local
ip http secure-server
ip http timeout-policy idle 60 life 86400 requests 10000
!
!
logging esm config
access-list 23 permit 192.168.0.0
!
!
control-plane
!
!
mgcp profile default
!
!
dial-peer voice 1 voip
 destination-pattern ^[2-9]......$
 session protocol sipv2
 session target ipv4:192.168.0.226:5160
 session transport udp
 dtmf-relay rtp-nte
 codec g711ulaw
!
!
gateway
 timer receive-rtp 1200
!
sip-ua
 no remote-party-id
 retry invite 3
 retry response 3
 retry bye 3
 retry cancel 3
 timers trying 1000
 sip-server ipv4:192.168.0.226
!
!
banner exec ^CC
 ---------------------------------------------------------
^C
banner login ^CC
Telephone traffic ONLY
---------------------------------------------------------
^C
!
line con 0
 login local
line aux 0
line vty 0 4
 access-class 23 in
 privilege level 15
 login local
 transport input telnet ssh
line vty 5 15
 access-class 23 in
 privilege level 15
 login local
 transport input telnet ssh
!
scheduler allocate 20000 1000
end

This is my trunk setup in FreePBX:
Outgoing

Trunk name: PRI
host=192.168.0.223
type=friend
context=from-internal
qualify=yes
nat=no
insecure=port,invite
dtmfmode=rfc2833
disallow=all
allow=ulaw

Incoming

User context: from-internal
type=friend
context=from-trunk
host=192.168.0.223
dtmfmode=rfc2833
disallow=all
allow=ulaw&alaw
nat=no
canreinvite=no
qualify=yes

SIP SHOW PEERS shows both "PRI" and "from-internal" as "OK"
SIP SHOW REGISTRY shows 0 SIP registrations

In FreePBX, Chan_SIP is on 5160 and PJ_SIP is on 5060. I tried connecting on both ports to no avail.

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@chaser wrote:

Hi,

Current PBX Version:14.0.1.4
Current System Version:12.7.3-1708-1.sng7
Current Asterisk Version: 13.17.1

Just finished an Admin/Updates/System Update today and I've started seeing some strange behaviour on incoming calls from my external PBX exchange.
I have one DAHDi trunk, which points to 'Analog Channel 1' in the DAHDI Trunk Settings. Oddly, once this has been set, the 'Analog Channel 1' selection disappears from the drop down menu (but I don't think that's the main problem).
Destination in the Inbound Route is set to Ring Group, which rings all internal extensions
If I use my mobile to make a call into the system via the DAHDi trunk, all extensions ring as expected - correctly showing the callerid. If I then hangup the call from my mobile (without picking up any of the extensions) the extensions stop ringing, but on some occasions the extensions will start ringing again for a couple of rings (callerid now displays unknown) and then stops.

I've tried moving the DAHDi trunk over to 'Analog Channel 4' and get the same result. (my card has 2 FXO & 2 FXS ports).
I have the same issue if I point the inbound route to a single extension instead of a Ring Group

Any ideas?

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@cloudnetmgmt wrote:

I had a perfectly working FreePBX 13 server using PJSIP, for at least 4 months. Today, I decided to upgrade from 10.13.66-20 to 10.13.66-21 using the System Admin module. After this update, I no longer have audio in either direction. For example, I can make a call to my cell phone sitting right next to me and everything appears to be OK, I answer the call, but no audio in either direction. When making the call, I also do not hear the typical ringing when a call is made. A packet trace on my firewall shows no blocked traffic and has absolutely no RTP traffic flowing. Any help in troubleshooting this would be appreciated. Thanks.

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@MIKEK wrote:

Freepbx 13

SQLSTATE[HY000]: General error: 145 Table './asteriskcdrdb/cel' is marked as crashed and should be repairedSQL -
SELECT * FROM asteriskcdrdb.cel WHERE uniqueid = '1505604683.0' OR linkedid = '1505604683.0' ORDER BY eventtime, id::

I tried

fwconsole util tablefix

but that did not handle this problem.

Any idea?

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@Pentium5 wrote:

Hi all,

I am using FreePBX13 paired with Asterisk13 running PJSIP driver only (chan_sip is disabled). My VoIP provider requires sending custom contact and auth user in SIP registration string. In the "old" chan_sip driver this was not an issue as these parameters could have been configured in one string (parameter "register string") and transparently provisioned to sip.conf in Asterisk.

However, I have not found any way to provision these parameters in PJSIP trunk settings.
Moreover, I tried to specify them in pjsip.registration_custom.conf and pjsip.registration_custom_post.conf files, but they do not seem to be read.
I used the following syntax:
[Trunk_name](+)
type=registration
contact_user=value

Does anyone know if there is any way to provision these parameters in UI?
If otherwise, how to specify them in custom files correctly?

Thank you in advance.

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@proj964 wrote:

wanted to try out freepbx. can't even get update installs to work:

0 info it worked if it ends with ok
1 verbose cli [ '/usr/bin/node', '/usr/bin/npm', 'install' ]
2 info using npm@5.0.3
3 info using node@v8.1.4
4 verbose npm-session ad13a0c183b28ef1
5 silly install runPreinstallTopLevelLifecycles
6 silly preinstall pm2@0.0.1
7 info lifecycle pm2@0.0.1~preinstall: pm2@0.0.1
8 silly lifecycle pm2@0.0.1~preinstall: no script for preinstall, continuing
9 silly install loadCurrentTree
10 silly install readLocalPackageData
11 silly install loadIdealTree
12 silly install cloneCurrentTreeToIdealTree
13 silly install loadShrinkwrap
14 silly install loadAllDepsIntoIdealTree
15 silly fetchPackageMetaData error for pm2@^2.4.4 request to hypertext link s //registry.npmjs.org/pm2 failed, reason: self signed certificate in certificate chain
16 silly fetchPackageMetaData error for pm2@^2.4.4 request to hypertext link s //registry.npmjs.org/pm2 failed, reason: self signed certificate in certificate chain
17 verbose type system
18 verbose stack FetchError: request to hypertext link s //registry.npmjs.org/pm2 failed, reason: self signed certificate in certificate chain
18 verbose stack at ClientRequest.req.on.err (/usr/lib/node_modules/npm/node_modules/pacote/node_modules/make-fetch-happen/node_modules/node-fetch-npm/src/index.js:68:14)
18 verbose stack at emitOne (events.js:115:13)
18 verbose stack at ClientRequest.emit (events.js:210:7)
18 verbose stack at TLSSocket.socketErrorListener (httpclient.js:400:9)
18 verbose stack at emitOne (events.js:115:13)
18 verbose stack at TLSSocket.emit (events.js:210:7)
18 verbose stack at emitErrorNT (internal/streams/destroy.js:62:8)
18 verbose stack at combinedTickCallback (internal/process/nexttick.js:102:11)
18 verbose stack at process.tickCallback (internal/process/nexttick.js:161:9)
19 verbose cwd /var/www/html/admin/modules/pm2/node
20 verbose Linux 3.10.0-514.26.2.el7.x86_64
21 verbose argv "/usr/bin/node" "/usr/bin/npm" "install"
22 verbose node v8.1.4
23 verbose npm v5.0.3
24 error code SELF_SIGNED_CERT_IN_CHAIN
25 error errno SELF_SIGNED_CERT_IN_CHAIN
26 error request to hypertext link s //registry.npmjs.org/pm2 failed, reason: self signed certificate in certificate chain
27 verbose exit [ 1, true ]
===========================================================
so tried to use "Let's Encrypt" to get a signed certificate. Am behind an isp dynamic ip. do have an ddns domain name that matches that ip address. however "Let's Encrypt" expects a running http server at that address, which right now there is not. I am not sure I understand why all this certificate stuff to run a pbx using an ITSP. Guess I don't understand why anyone would need encrypted external access to the freepbx server. I am beginning to think what I wanted to try just isn't possible. Maybe someone can point me in the right direction. Thanks.

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@bobh1950 wrote:

So here is the situation. I have one 20 channel DID for conferencing using Confbridge. Three different groups want to use it a different times. What could I do to set it up so that no one from one group could call in and listen during another group's meeting? Each group values it's privacy.

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@cynjut wrote:

I have one installation (32-bit PIAF running Asterisk 1.4).

I'd like someone to write me a script that converts this to Asterisk 14 AND provides a new driver for the DAHDI card from a supplier that went out of business 10 years ago.

Just kidding - I just wanted to whine for a minute.

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@peter007 wrote:

I just virtualized one FPBX on Hyper-V 2016 and imported the settings from a physical server. Everything ran (more or less) without bigger errors.

One thing is voicemail-notification are not delivered / stored properly. Calls are always shown in the calls list in UCP, but voicemails sometimes are shown with attachments, sometimes neither they nor the attachements are shown in vm-list (and are not delivered via email).

In "voicemail settings", if I try to change something and apply it, I get the followinig error:

exit: 255
Unable to continue. SQLSTATE[23000]: Integrity constraint violation: 1062 Duplicate entry 'hooks-noid' for key 'uniqueindex' in /var/www/html/admin/libraries/BMO/DB_Helper.class.php on line 268

0 /var/www/html/admin/libraries/BMO/DB_Helper.class.php(268): PDOStatement->execute(Array)

1 /var/www/html/admin/libraries/BMO/Hooks.class.php(117): FreePBX\DB_Helper->setConfig('hooks', Array)

2 /var/www/html/admin/libraries/BMO/Hooks.class.php(27): FreePBX\Hooks->updateBMOHooks()

3 /var/www/html/admin/libraries/BMO/DialplanHooks.class.php(159): FreePBX\Hooks->getAllHooks()

4 /var/www/html/admin/libraries/BMO/DialplanHooks.class.php(27): FreePBX\DialplanHooks->getBMOHooks()

5 /var/lib/asterisk/bin/retrieve_conf(854): FreePBX\DialplanHooks->getAllHooks(Array)

6 {main}

edit:

besides importing settings, I installed German sounds via:

mkdir /var/lib/asterisk/sounds/de
cd /var/lib/asterisk/sounds/de
wget -O core.zip https___www.asterisksounds.org/de/download/asterisk-sounds-core-de-sln16.zip
wget -O extra.zip https___www.asterisksounds.org/de/download/asterisk-sounds-extra-de-sln16.zip
unzip core.zip
unzip extra.zip
chown -R asterisk.asterisk /var/lib/asterisk/sounds/de
find /var/lib/asterisk/sounds/de -type d -exec chmod 0775 {} \;´

(don´t really know why FreePBX does not support German sounds by default)

So maybe this is a permission problem?

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@dwsiemens wrote:

I working at moving the mysql db engine from the same host to a cluster running galara. / Mariadb on freepbx 14.

So far so good but have one place were it would be good to change source.

You have 1 tampered files
Module: "Call Event Logging", File: "/var/www/html/admin/modules/cel/Cel.class.php altered"

I changed the value of db_host to 127.0.0.1 from localhost

Would it be possible to have this change in the module upstream done, so it would work in either case?

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@etgllc wrote:

When I first installed and activated FreePBX, I ended up using an email account that I didnt want to use. How do I update FreePBX system to attach my deployment ID to another email address?

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@cmhofmmoss wrote:

In the last month or so we have lost the ability to restart the freePBX application via the web interface. We have the most updated version of the application installed with all current OS updates as well. Has anyone else experienced this and know a fix?

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@perrcla wrote:

Hi!

freepbx 13 with gxp2130.
I configured some virtual-key in my phone as BLF.
Also, in my phone configutation i set to yes "show popup when monitored blf ring".

Extension 300 start ring. In extension 301 a popup appears "UNKNOW is calling 300", but when i pickup the call, the callerid appears normally.

is there a way to pass the callerid before answering?

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@vbman213 wrote:

Does FreePBX support enumerating nested LDAP groups?

Parent Group
....Child Group 1
........Member 1
........Member 2
....Child Group 1
........Member 3
........Member 4

When queried, Parent Group should include Member 1, Member 2, Member 3, and Member 4.

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@dwsiemens wrote:

When trying to install freepbx SystemAdmin module

I am getting a error screen

On the left edge i have the following errors.

admin/modules/sysadmin/install.php17
5
mkdir
/var/www/html/admin/modules/sysadmin/install.php17
4
include_once
/var/www/html/admin/libraries/modulefunctions.class.php2499
3
module_functions _doinclude
/var/www/html/admin/libraries/modulefunctions.class.php2451
2
module_functions _runscripts
/var/www/html/admin/libraries/modulefunctions.class.php1985
1
module_functions install

In the middle I have

admin/modules/sysadmin/install.php
<?php @Zend;
0604;
?>

System Admin 14.0.7.22
Copyright 2017 by Schmoozecom, Inc., All rights reserved

By installing, copying, downloading, distributing, inspecting or using
the materials provided herewith, you agree to all of the terms of use as
outlined in our End User Agreement which can be found and reviewed at
www.schmoozecom.com/cmeula

<?PHP
exit();
_haltcompiler();
?>

2004072203655431092740021x�
�2�}ݏd�uߝ鉵��Ճ
J����g��ǽ�� �^-�

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@vbman213 wrote:

I know I set this to "PJSIP" when I connected to Active Directory.

However, if I go into the directory settings, it says "Don't Create"

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@hakimrs wrote:

Hey guys,

I have one freePBX with 4extensions with an IVR to help chosing which one of them . And i'm trying to forward all the calls coming to ONE of this extensions to AN EXTERNAL phone number.

I used the MISC destination, and added it to the Follow destination.
I used even that with the ring groups module.

but i still can t forward this extention calls

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