vicidial.org

Good evenig,

I'm working in a little company in Mauritius, with 25 agents on site and 04 agents working from home in another country. 28 of my agents are using MicroSIP Lite 3.20.5 with vicidial to make and receive calls. BUT...

... My issue is that 01 of my agent who is in Tunis is using an IP Phone Yealink T415 and I need to connect her on the system.

I d'ont know how to add the hard phone in vicidial. For the SIP phones it's all ok, but for the IP Phone, I really d'ont know how to.

Can someone show me the configuration of how to add the Yealink T415 in Vicidial please ?

My Configuration are:
VICIDIAL:
SERVER DESCRIPTION IP ACT LOAD CHAN DISK TIME VER
vicibox-1 Server vicibox-1 0.0.0.0 Y 16 - 4% 9 8% 2021-04-02 03:53:47 3094
/usr/src/astguiclient/trunk
Path: .
Working Copy Root Path: /usr/src/astguiclient/trunk
URL: svn://svn.eflo.net/agc_2-X/trunk
Relative URL: ^/agc_2-X/trunk
Repository Root: svn://svn.eflo.net
Repository UUID: 3d104415-ff17-0410-8863-d5cf3c621b8a
Revision: 3094
Node Kind: directory
Schedule: normal
Last Changed Author: mattf
Last Changed Rev: 3094
Last Changed Date: 2019-04-25 02:02:16 +0200 (Thu, 25 Apr 2019)

ASTERISK:
Asterisk Configuration:
Asterisk 16.2.1, Copyright (C) 1999 - 2018, Digium, Inc. and others
System information as of Fri Apr 2 03:44:41 CEST 2021

System load: 0.0 Processes: 103
Usage of /: 13.5% of 1006.87GB Users logged in: 0
Memory usage: 2% IP address for ens18: 0.0.0.0
Swap usage: 0% IP address for ens19: 0.0.0.0

ACCOUNT ENTRY:
[activsol]
type=friend
context=trunkinbound
secret=@5ter15k1@@u5
username=activsol
host=0.0.0.0
disallow=all
allow=ulaw
insecure=port,invite
fromdomain=0.0.0.0
deny=0.0.0.0/0.0.0.0
permit=0.0.0.0/255.255.255.0


DIALPLAN ENTRY:
exten = _90ZXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten = _90ZXXXXXXXX,2,Dial(${ACTIVTRUNK}/06033${EXTEN:2},,To)
exten = _90ZXXXXXXXX,3,Hangup

exten => _910ZXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _910ZXXXXXXXX,2,Dial(${ACTIVTRUNK}/06033${EXTEN:3},,To)
exten => _910ZXXXXXXXX,3,Hangup

exten=> s,1,AGI(agi://127.0.0.1:4577/call_log)
exten=> s,n,NoOp(No Dial Pattern Matches This Extension)
exten=> s,n,Hangup

I thank you in advance for your help.

Yours sincerely

Rasbenny.

your using asterisk 16?

Hi, good evening,
Yes, it seem to be Asterisk 16. It is what is written on the system, plse see below:

PBX Core settings
-----------------
Version: 16.2.1
Build Options: BUILD_NATIVE, OPTIONAL_API
Maximum calls: Not set
Maximum open file handles: 1024
Root console verbosity: 0
Current console verbosity: 3
Debug level: 0
Maximum load average: 0.000000
Minimum free memory: 0 MB
Startup time: 05:19:52
Last reload time: 05:19:52
System: Linux/4.15.0-45-generic built by root on x86_64 2019-03-03 20:56:54 UTC
System name:
Entity ID: 02:00:00:e3:46:cc
PBX UUID: 20e67dac-f450-46ae-8fd7-b87cf8aae51c
Default language: en
Language prefix: Enabled
User name and group: /
Executable includes: Disabled
Transcode via SLIN: Enabled
Transmit silence during rec: Disabled
Generic PLC: Enabled
Generic PLC on equal codecs: Disabled
Min DTMF duration:: 80
Cache media frames: Enabled
RTP use dynamic payloads: 1
RTP dynamic payload types: 35-63,96-127

* Subsystems
-------------
Manager (AMI): Disabled
Web Manager (AMI/HTTP): Disabled
Call data records: Enabled
Realtime Architecture (ARA): Disabled

* Directories
-------------
Configuration file: /etc/asterisk/asterisk.conf
Configuration directory: /etc/asterisk
Module directory: /usr/lib/asterisk/modules
Spool directory: /var/spool/asterisk
Log directory: /var/log/asterisk
Run/Sockets directory: /var/run/asterisk
PID file: /var/run/asterisk/asterisk.pid
VarLib directory: /var/lib/asterisk
Data directory: /var/lib/asterisk
ASTDB: /var/lib/asterisk/astdb
IAX2 Keys directory: /var/lib/asterisk/keys
AGI Scripts directory: /var/lib/asterisk/agi-bin

Yours Sincerely.
Rasbenny.

im not sure how thats possible on a vicidial server, ast16 is still in vici beta and not really released to the public yet, and from what i can see, youve had your center going since at least 2019 which definitely did not come with a working ast16-vici version, did you change something?

Hi,

Not at all, no change have been made since i'm here (which is not that long, just a couple of months, since September last).

I'm sorry to say that but all I know is only which system they left in my hand.

Yours Sincerely
Rasbenny.

Ok so anyways, for hard phones you need to setup the template for them depending on the hardphone you choose to use

Hi,

First of all, many thanks for your reply and the time you are giving me.

Please could you give me a hint or explain to me how to set up the template?

The hard phone I need to add is a Yealink T415.

I thank you again for your kind help.

Yours Sincerely
Rasbenny.

i dont personally use hard ip phones, however im sure there is someone in the community that will respond with a possible solution, your best bet tho is to try on the asterisk forum for a question like that

Hi,
Ok, I will try on the other forum.

Thanks again for your kinf help.

Yours Sincerely.
Rasbenny.

configuring IP Hardphone similar to softphone configuration.

you need to know your sip extensions username(extension), password(secret) and domain ip (pbx/vicidial)

browse the Yealink IP address in any browser(you can check the ip in phone display) , login with admin /admin or admin'/1234.

then under accounts section enter sip details to register with your pbx or dialer.


according to your posts seems like you are using Vicidial also Freepbx setup, might some custom deployment with both.
so confirm your setup first freepbx or vicidial ?