How to disable SIP ALG

SOURCE:
http://www.voip-info.org/wiki/view/Routers+SIP+ALG

Many of today’s commercial routers implement SIP ALG (Application-level gateway), coming with this feature enabled by default. While ALG could help in solving NAT related problems, the fact is that many routers’ ALG implementations are wrong and break SIP.

CheckPoint
login to Smart Dashboard
click Smart Defence tab
expand Application Intelligence
expand VoIP
disable all features on H.323

Cisco
(config)# no ip nat service sip tcp port 5060
(config)# no ip nat service sip udp port 5060

ASA
(config)# policy-map global_policy
(config)# no inspect sip

Cyberoam
> cyberoam system_modules sip unload

D-Link
Open a browser and enter the router’s IP address in the address bar. Go to “Firewall Settings” under the “Advanced” item.
Uncheck the box to disable SPI – usually, directly below this item are options for “NAT Endpoint Filtering” that must be changed to “Endpoint Independent” for both TCP and UDP.
Next, find the “Application Level Gateway (ALG) Configuration” area and uncheck the box for SIP.
Save these settings and reboot the device if requested

FortiGate
disable SIP ALG
# config system settings
# set sip-helper disable
# set sip-nat-trace disable
# end
verify
# show full-configuration system settings
delete sip
# config system session-helper
(session-helper) # show
config system session-helper
edit 1
set name pptp
set protocol 6
set port 1723
next
edit 2
set name h323
set protocol 6
set port 1720
next
edit 3
set name ras
set protocol 17
set port 1719
next
edit 4
set name tns
set protocol 6
set port 1521
next
edit 5
set name tftp
set protocol 17
set port 69
next
edit 6
set name rtsp
set protocol 6
set port 554
next
edit 7
set name rtsp
set protocol 6
set port 7070
next
edit 8
set name rtsp
set protocol 6
set port 8554
next
edit 9
set name ftp
set protocol 6
set port 21
next
edit 10
set name mms
set protocol 6
set port 1863
next
edit 11
set name pmap
set protocol 6
set port 111
next
edit 12
set name pmap
set protocol 17
set port 111
next
edit 13
set name sip
set protocol 17
set port 5060
next
edit 14
set name dns-udp
set protocol 17
set port 53
next
edit 15
set name rsh
set protocol 6
set port 514
next
edit 16
set name rsh
set protocol 6
set port 512
next
edit 17
set name dcerpc
set protocol 6
set port 135
next
edit 18
set name dcerpc
set protocol 17
set port 135
next
edit 19
set name mgcp
set protocol 17
set port 2427
next
edit 20
set name mgcp
set protocol 17
set port 2727
next
end
(session-helper) # delete 13
(session-helper) # end

Juniper
https://kb.juniper.net/InfoCenter/index?page=content&id=KB7078&actp=search
# set security alg sip disable
# commit and quit

Mikrotik
> ip firewall service-port set sip disabled=yes

Netgear
From Wan Setup Menu, NAT Filtering, uncheck the box next to “Disable SIP ALG”

PaloAlto
https://live.paloaltonetworks.com/t5/Configuration-Articles/How-to-Disable-SIP-ALG/ta-p/60637
# set shared alg-override application sip alg-disabled yes

Peplink
go to http://<router.LAN.IP>/cgi-bin/MANGA/support.cgi
Click the “Disable” button under “SIP ALG Support”

SonicWall
in GUI, go to VOIP>Settings>General Settings
tick Enable consistent NAT
untick Enable SIP Transformations

SpeedTouch
telnet router
> connection unbind application=SIP port=5060
> saveall

Zyxel
telnet router
Menu option “24. System Maintenance”.
Menu option “8. Command Interpreter Mode”.
ip nat service sip active 0

3cx v15 Installation

Download 3cx v15 from
http://www.3cx.com/phone-system/download-phone-system/

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Open “3cx Management Console” on the Desktop
Open Settings>Internet Options>Security>Trusted Sites>Sites
add both https://sip1.3cx.asia and https://sip1.poc.local
click Reload

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click Finish
Extensions

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-adding 3cx Windows client

download and install 3cx Windows client given by Extension “Send Welcome Email”
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-adding 3cx Mac client
download and install 3cx Mac client given by Extension “Send Welcome Email”

-adding 3cx Android client
download and install 3cx Android client given by Extension “Send Welcome Email”

-adding 3cx iPhone client
download and install 3cx iPhone client given by Extension “Send Welcome Email”
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-adding Phones
In this example is using Fanvil C56P
to set ip from DHCP (if you didn’t implement Auto Provision from PBX)
go to Basic>WIZARD>DHCP
click Next
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click Next

click Finish

Groups

Contacts
http://www.3cx.com/docs/manual/phonebook-directory/
Integration with LDAP can only happen in Enterprise License
Open 3cx GUI and click Contacts>LDAP
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Digital Receptionist

Ring Groups

Call Queues

Web Meeting

Backup/Restore
You can do manual or scheduled backup. Backup/Restore file must be in this C:\ProgramData\3CX\Instance1\Data\Backups folderImage.png

NTP Server

CISCO
NTP MASTER
(config)#ip name-server 8.8.8.8
(config)#ntp server id.pool.ntp.org
(config)#clock timezone WIB 7
(config)ntp update-calendar
#show clock detail
#show calendarNTP CLIENT
(config)#clock timezone WIB 7
(config)#ntp server 12.12.12.1
# show ntp associations
# show ntp status
#show clock detail

FORTIGATE
using console, login as admin
config system global
    set fgd-alert-subscription advisory latest-threat
    set hostname “FGT1”
    set timezone 53
end
config system dns
    set primary 208.91.112.53
    set secondary 208.91.112.52
end
config system ntp
    set ntpsync enable
    set type custom
    set syncinterval 360
    config ntpserver
        edit 1
            set server “id.pool.ntp.org”
        next
    end
    set server-mode enable
    set interface “port1”
end

FGT1 # exe date
current date is: 2016-09-17
FGT1 # exe time
current time is: 19:28:04

JUNIPER
NTP MASTER
R1# show
system {
host-name R1;
domain-name poc.com;
time-zone Asia/Jakarta;
root-authentication {
encrypted-password “$1$k1Iyv6h7$zpP9XotU3WcLUU2Kf9baC.”; ## SECRET-DATA
}
name-server {
8.8.8.8;
ntp {
boot-server 202.162.32.12;
server 103.28.56.14;
}
}
interfaces {
em0 {
unit 0 {
family inet {
address 10.0.10.151/24;
}
}
}
em1 {
unit 0 {
family inet {
address 12.12.12.1/24;
}
}
}
lo0 {
unit 0 {
family inet {
address 127.0.0.1/32;
}
}
}
}
routing-options {
static {
route 0.0.0.0/0 next-hop 10.0.10.1;
}
}

# run show ntp associations
     remote           refid      st t when poll reach   delay   offset  jitter
=============================================================

 ns1.matrixgloba 203.123.48.219   2 –   22  512  377   19.156  2995.28 1137.12

NTP CLIENT
R2# show
system {
host-name R2;
domain-name poc.com;
time-zone Asia/Jakarta;
root-authentication {
encrypted-password “$1$ZyPyDk7D$KjTrKc1c61UuNszJ/HplX.”; ## SECRET-DATA
}
name-server {
8.8.8.8;
}
ntp {
boot-server 12.12.12.1;
server 12.12.12.1;
}
}
interfaces {
em0 {
unit 0 {
family inet {
address 12.12.12.2/24;
}
}
}
lo0 {
unit 0 {
family inet {
address 127.0.0.1/32;
}
}
}
}
routing-options {
static {
route 0.0.0.0/0 next-hop 12.12.12.1;
}
}

> show ntp status
> run show ntp associations
remote refid st t when poll reach delay offset jitter
==============================================================================
*12.12.12.1 103.28.56.14 3 – 32 64 77 1.786 1088.36 599.222

 

MIKROTIK
-download and install ntp server package inside “Extra packages from http://www.mikrotik.com/download

 

-check your LAN ip address
> ip address print
Flags: X – disabled, I – invalid, D – dynamic
 #   ADDRESS            NETWORK         INTERFACE
 0   ;;; default configuration

     10.0.46.1/24       10.0.46.0       bridge-local

-set your Mikrotik time zone
> /system clock print
                  time: 17:25:06
                  date: sep/12/2016
  time-zone-autodetect: yes
        time-zone-name: Asia/Jakarta
            gmt-offset: +07:00

            dst-active: no

-set your NTP server
> system ntp server print
              enabled: yes
            broadcast: no
            multicast: no
             manycast: yes

  broadcast-addresses:

-set your Mikrotik sync its time with id.pool.ntp.org
> system ntp client print
          enabled: yes
             mode: unicast
      primary-ntp: 202.162.32.12
    secondary-ntp: 0.0.0.0
  dynamic-servers:

           status: synchronized

-configure firewall to allow ntp traffic
/ip firewall nat print
Flags: X – disabled, I – invalid, D – dynamic
 0    ;;; default configuration
      chain=srcnat action=masquerade out-interface=ether1-gateway log=no
      log-prefix=””
 1    ;;; NTP
      chain=srcnat action=src-nat to-addresses=10.0.46.1 protocol=udp

      src-port=123

-verify current clock
> system clock print
                  time: 17:38:10
                  date: sep/12/2016
  time-zone-autodetect: yes
        time-zone-name: Asia/Jakarta
            gmt-offset: +07:00
            dst-active: no

WIN20012R2 can’t get windows update

This is fresh install of SW_DVD9_Windows_Svr_Std_and_DataCtr_2012_R2_64Bit_English_-4_MLF_X19-82891.ISO.
I did Windows Update and get 208 Updates but nothing updated.
After googling and found this

Tested working. Here the steps

  1. Download and extract PSWindowsUpdate.zip from
    https://gallery.technet.microsoft.com/scriptcenter/2d191bcd-3308-4edd-9de2-88dff796b0bc
    p
    ut it into C:\Program Files\WindowsPowerShell\Modules\
  2. Run PowerShell as Administrator
    >Import-Module PSWindowsUpdate
    >Get-WUInstall
    When it prompt for update, press [A] Yes to All
    It will ask to Reboot after finish Updating

S20 Basic Config

Factory default
IP: 192.168.5.150
L: admin

P: password

NOTE:
-if using Opera, set “Block ads/Manage exceptions/yeastaripaddress”

go to https://192.168.5.150:8088

-change ip address

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-upgrade Firmware

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NOTE:

-Email above is important when you forgot the password, you can click Forgot Password and the system will send link to reset to new password
-Registration Password is your SIP client extension password
-User Password is the password user to login to https://10.0.10.219:8088
L: extentsionnumber

P: userpasswordabove

-there is no root a/c or password in S series
L: support
P: iyeastar
root user is ls@yf
for other model
L: root

P: ys123456

-set firewall to only allow GUI from my pc ip
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NOTE: I must put 10.0.10.114/255.255.255.255 there. I can’t put 10.0.10.114/255.255.255.0 there. Otherwise blocking GUI won’t work
No need to enable Drop All
If someday, you can’t access GUI, ssh or ping. Please change your pc ip and check Settings>System>Security>IP Auto Defence>Blocked IP Address
by default already configured below

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-set event and notification

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If an event occured, it will send email like this
    [2016-08-17 18:48:32] The password of Administrator account has been modified. IP Address: 10.0.10.114.
or SMS
   [2016-08-18 00:00:00] CPU Overload. Current Usage: 98

or Call Extension or Mobile with no sound (call only)

-Backup/Restore config
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to Restore, click Upload then click Restore
-to check GSM signal quality
login as root or support
$ asterisk -vvvr
voip1*CLI> gsm show spans
GSM span 4: Power on, Up, Active, Standard
voip1*CLI> gsm show span 4
D-channel: 4
Status: Power on, Up, Active, Standard
Type: CPE
Manufacturer: SIMCOM_Ltd
Model Name: SIMCOM_SIM900
Model IMEI: 013227005472208
Revision: 1137B12SIM900M64_ST
Network Name: 3
Network Status: Registered (Home network)
Signal Quality (0,31): 20
BER value (0,7): 0
SIM IMSI: 510890170771570
SIM SMS Center Number: +6289644000001
PDD: 0
ASR: 0
ACD: 0
Last event: D-Channel Up
State: READY
Last send AT: AT+CSQ\r\n
-to check SIP logging
> pjsip set logger on
PJSIP Logging enabled
<— Received SIP request (1321 bytes) from UDP:10.0.10.114:5060 —>
INVITE sip:889684094@10.0.10.219 SIP/2.0
Call-ID: 04bda17e91d6502523e9291196f89d07@0:0:0:0:0:0:0:0
CSeq: 1 INVITE
From: “nbctcp” <sip:1002@10.0.10.219>;tag=424e3955
To: <sip:889684094@10.0.10.219>
Via: SIP/2.0/UDP 10.0.10.114:5060;branch=z9hG4bK-3239-af06ce5b1bae054271a8034d09a514a2
Max-Forwards: 70
Contact: “nbctcp” <sip:1002@10.0.10.114:5060;transport=udp;registering_acc=10_0_10_219>
User-Agent: ippi Messenger2.3.2705Windows 8
Content-Type: application/sdp

Content-Length: 825

-to reset GSM modem
login as root or support
> cd /ysdisk/support
> asterisk -vvvr
> gsm show spans
> gsm debug span 4

> gsm send at 4 ATZ

-TF card requirement
We recommend that you use a high-performance TF/SD card.
SDHC/SDXC Class 10; UHS-I U3
  • Recommended TF/SD Card (Write Speed >=60MB/s)
  • Sandisk Extreme Pro Series
  • Sandisk Extreme Series
  • TOSHIBA EXCERIA Series
  • Samsung Pro Series
-TF card function
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-Upgrade firmware
  •    check current firmware
click Resource Monitor/Information
Software Version: 30.0.0.40
  •    download new firmware
  •    upgrade firmware
-Ring Groups
A ring group helps you to ring a group of extensions in a variety of ring strategies. For example, you could define all the technical support guys’ extensions in a ring group and ring the support guys one by one
click Settings>PBX>Call Features>Ring Group

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You have ring option to
Ring All=all member ring but if one of them pickup, the ring will stop

Sequentially=Will ring first member then second etc

-Conference

click Settings>PBX>Call Features>Conference
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Wait for Moderator= Conference won’t start without Moderator join but any user can be an admin if they know the admin password
Allow Participant to invite= Participant can invite other extension or telephone number by pressing #
Participant can join the conference from outside office by calling IVR
Internal Participant can invite other external user by pressing # but I have problem with that scenario.
I can invite external user, their telephone ringing, but they can’t hear any sound or voice then after 20s the phone hangup

The problem above only happen in GSM trunk. I don’t have problem with FXO trunk

-Transfer

Blind Transfer= Dial *03 and an extension number to blind transfer the call.
Attended Transfer= Dial *3 and an extension number to transfer the call. Hang up after contacting the destination
-Call Forwarding
Enable Forward All Calls= *71 forward to voicemail
*71XXXX forward to XXXX extension
Disable Forward All Calls= *071
Enable Forward When Busy= *72 forward to voicemail
*72XXXX forward to XXXX extension

Disable Forward When Busy= *072

-Pickup Group

Call pickup allows one to answer someone else’s call. You can add pickup group. The default call pickup for Group Call Pickup is *4. It allows you to pick up a call from a ringing phone which is in the same group as you.

click Settings>PBX>Call Features>Pickup Group

-Call Pickup

Extension Pickup: Dial this *04 code and an extension number to pick up a call that is ringing at the extension.

-Voicemail

Press *2 to hear voicemail
Voicemail for Extension= leave a voicemail to other extensions by dialing feature code on their phone or forward an incoming call to an extension’s voicemail directly.
The default feature code is #.
For example, dial “#501” to leave a message for Ext. 501

Voicemail Main Menu=The feature code *02 that is used to access voicemail main menu

-Speed Dial
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to dial telp number 1 press *991

-Do Not Disturb (DND)
Don’t  Disturb.  When  DND  is  enabled  for  an  extension,  the extension will not be available
to enable *74

to disable *074

-Call Blacklist/Whitelist
go to Settings>PBX>Blacklist/Whitelist
to import blacklist.csv the format must be like this
Name,Number,Type
AdsBoth,089519228111-111,both
AdsOut,087809777222,outbound
Whitelist function is let say the Blacklist support ‘pattern’ to match a batch of numbers, for example, 123. will block all numbers those first three digit is 123.

In this case, if we would like to allow 123456 to call in, we can add 123456 into the Whitelist

-Intercom/Zone Intercom/Paging

go to Settings/PBX/Call Features/Paging Intercom/
click Add

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-Call Parking

Call Parking is a feature that allows the user to put a call on hold at one phone and continue the conversation from any other phone
Call Parking=Dial this *6 code to put a call on hold and park the call at an extension  number  directed  by  the  system.  Any  other phone  can dial this extension number to resume the conversation
Directed Call Parking= Dial this *6 code and an extension number to park the call at
that extension. Any other phone can dial this extension number to resume the conversation.
Note:
Extension number above is extension number in Settings>PBX>General>Call parking>parking Extension Range

if the directed extension number is occupied, the call parking will fail

-Time Conditions

On Time Condition page, you can create time groups. A time group is a list of times against which incoming or outgoing calls are checked. The rules specify a time range, by the time, day of the week, day of the month, and month of the year.
Time conditions can be assigned to an inbound route, which control  the  destination  of  a  call  based  on  the  time.
Time  conditions  can  also  be  assigned  to  an outbound route in order to limit the use of that route.
go to Settings>PBX>Call Control>Time Conditions

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-dhcp auto provisioning

Upload Files= Users  could  upload  phone  configuration  file  to  S-Series  system,  and  apply  the  configurations  to desired IP phones.
Phonebook= You can add contacts on S-Series, or upload a phone book to S-Series, the phone book on S-Series will be applied to the phones during the phone provision process
NOTE: All the existing phonebooks of the IP phone would be deleted automatically if the phonebooks are configured in this way
Firmware Upgrade= IP phone firmware can be uploaded to the S-Series internal storage for firmware upgrade
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Just click Scan to scan any ip phone in your subnet

-SMS to Email and Email to SMS

Make sure you can receive email from Test email and you enable “Send Voicemail to Email” in your extension

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Anyone send SMS to your SIM card in GSM module will be forwarded to your email
Anyone send email to your email with subject
num:87809777888; or
port:3;num:87809777888; or
port:3;num:87809777888;code:123;

(number without country code or 0 in front) will be SMSed to subject number

-VoiceMail to email

Make sure you can receive email from Test email and you enable “Send Voicemail to Email” in your extension

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-Define business hours
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-Call Detail Records

open CDR and Recordings

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-Call Back

Callback feature allows callers to hang up and get called back to Yeastar S-Series Callback feature could reduce the cost for the users who work out of the office using their own mobile phones.
Go to Settings > PBX > Call Features > Callback

 

-Call Waiting
Check  this  option  if  the  extension  should  have  Call  Waiting capability.  If  this  option  is  checked,  the  “When  busy”  call forwarding options will not be available. The call waiting function of IP phone has higher priority than MyPBX call waiting function.
If you want to switch between caller, press HOLD. It will switch to 2nd caller. The 1st caller become in hold

-Distinctive Ringtone

The  system  supports  mapping  to  custom  ring  tone  files.  For  example,  if  you  configure  the distinctive  ringing  for  custom  ring  tone  to  “Family”,  the  ring  tone  will  be  played  if  the  phone receives the incoming call.
Set the name of the tone on the phone
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In this example, I will set lagu9 as a tone for incoming call from GSM
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-SIP Forking
Single SIP call to multiple SIP endpoints with the same extension number, but make sure you restart the phone after every changing.
Set in Concurrent Registration
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On old U Series, you need to do these
  1. create 1 extension
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    2. connect through SIP Phone
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    3. connect through IAX Phone
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    -Queues
    Queues are designed to receive calls in a call center.
    A queue is like a virtual waiting room, in which callers wait in line to talk with the available agent. Once the caller called in S-Series and reached the queue, he/she will hear hold music and prompts, while the queue sends out the call to the logged-in and available agents.
    A number of configuration options on the queue help you to control how the incoming calls are routed to the agents and what callers hear and do while waiting in the line
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    Basic Settings
    Number
    Use this number to dial into the queue, or transfer callers to this number to put them into the queue
    Name
    Give this queue a brief name to help you identify it
    Password
    You can require agents to enter a password before they can login to this queue.
    Ring Strategy
    This option sets the Ringing Strategy for this Queue. The options are:
      Ringing All: ring all available agents simultaneously until one answer.
      Least Recent: ring the agent which was least recently called.
      Fewest Calls: ring the agent with the fewest completed calls.
      Random: ring a random agent.
      Rememory: Round Robin with Memory, remembers where it left off in the last ring pass.
      Linear: rings agents in the order specified in the configuration file.
    Failover Destination
    Set the failover destination
    Static Agents
    This  selection  shows  all  users.  Selecting  a  user  here  makes  them  a dynamic agent of the current queue. The dynamic agent is allowed to log in and log out the queue at any time.
      Dial “Queue number” + “*” to log in the queue.
      Dial “Queue number” + “**” to log out the queue.
    Agent Timeout
    The number of seconds an agent’s phone can ring before we consider it a timeout. If you wish to customize, enter the value in the text box directly.
    Agent
    Announcement
    Announcement played to the Agent prior to bridging in the caller
    Wrap-up Time
    How  many  seconds  after  the  completion  of  a  call  an  Agent  will  have before the Queue can ring them with a new call .If you wish to customize, enter the value in the text box directly. Input 0 for no delay
    Ring In Use
    If set to “no”, unchecked, the queue will avoid sending calls to members whose device are known to be “in use”
    Retry
    The number of seconds to wait before trying all the phones again. If you wish to customize, enter the value in the text box directly.
    Caller Experience Settings
    Music On Hold
    Select the “Music on Hold” playlist for this Queue
    Caller Max Wait Time
    Select the  maximum  number  of seconds  a  caller can  wait  in  a queue before being pulled out. If you wish to customize, enter the value in the text box directly. Input 0 for unlimited
    Leave When Empty
    If enabled, callers already on hold will be forced out of a queue when no agents available
    Join Empty
    If enabled, callers can join a queue that has no agents
    Join Announcement
    Announcement played to callers once prior to joining the queue
    Announce Position
    Announce position of caller in the queue
    Announce Hold Time
    Enabling this option causes PBX to announce the hold time to the caller periodically based on the frequency timer. Either yes or no; hold time will be announced after one minute.
    Frequency
    How often to announce queue position and estimated hold time
    Prompt
    Select a prompt file to play periodically
    -Music on Hold
    download and install audacity from http://audacity.sourceforge.net
    download a mp3
    run audacity and click File/Import/Audio and choose a mp3
    click Tracks>Stereo Track to Mono
    click Tracks>Resample
       set 8000Hz
    click File>Export Audio
       File name: must have .wav
       Save as type: Other uncompressed files
       Header: WAV (Microsoft)

       Encoding: U-Law

    open Yeastar GUI
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    -IVR (Interactive Voice Response)
    Like most organizations, where possible, we would like to route incoming calls an Auto Attendant.
    You can create one or more IVR (Auto Attendant) on S-Series to achieve it. When calls are routed to an IVR, S-Series will play a recording prompting them what options the callers can enter such as “Welcome to XX, press 1 for Sales and press 2 for Technical Support”

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    -Tone Region

    If your country is not listed, make sure your chosen country region match Busy Tone, Dial Tone, Ringing Tone of your own country

     

    TO DO

    -Mobility Extension
    This function different than Cisco CUCM. In Cisco mean yu can login using your extension number into other user phone as long as you know your password

Continue reading

C56P Basic Setup

-Upgrade Firmware
login to your ip phone GUI
check current firmware version in the GUI
before upgrade Version: 2.3.157.108
go to MAINTENANCE > UPDATE > Web Update

after upgrade Version: 2.3.785.400

-to change default GUI to https
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NOTE: C56 didn’t support HTTPS GUI

-how to allow GUI from certain ip

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-how to hear voicemail after press voicemail button
MWI=Message Waiting Indication
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in Yeastar

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-How to reset lost password
if your LCD password is lost , you could set it in the follow web page . If your web access password is lost , you should do the factory reset in LCD (menu-settings-Advanced Settings – Reset to Default ) . If both are lost
power off
hold #
power on
press *#168 in POST Mode
power off/on

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-to Speed up user dial without have to press #SEND key
20160901 20.05

-how to key in sip userid and password into the phone directly

I am going to other room and I want to login using my userid into that room phone. Once I don’t need that phone, I just need to logout.
Its called Hot Desk. At present only supported by C58, C62, C66, C600
It is a function named Agent . You could get the operation in http://www.fanvil.com/images/user/2014050805.pdf

-to call an extension and force immediate pickup (phone will automatically go to speaker phone). Audio will be two-way
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Integrating AD into Phonebook

I want to have central location of Phonebook for all SIP clients.
In this scenario, I want to store all user ipphone and mobile information in AD
  1. Download and run ADExplorer from https://technet.microsoft.com/en-us/sysinternals/adexplorer.aspx
    T
    his will be useful to locate AD user syntax, such as CN, OU, DN, etc
  2. Configure service account svcphone
    This svcphone will only have READ permission on POC OU
    open ADUC
    create svcphone, user1, user3, user3 account
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  1. Delegate READ permission for svcphone a/c in POC OU
    right click POC OU/Delegate Control
    click Next
    in Users or Groups, click Add, search svcphone
    click OK, click Next 2x
    click Create a custom task to delegate
    click Next
    click Only the following objects in the folder. Tick User objects
    click Next
    in Permissions, tick General, tick Read
    click Next, click Finish
  2. Configure SIP clients
    IPPI
    click Tools/Options/Advanced/Contact sources/LDAP configuration
    click +
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    click Save

    click Tools/Accounts
    click Add
    Network SIP
    SIP id 1001
    Password
    tick Remember password
    click Advanced
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    Now if search user phone
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    YEALINK
    I can’t test yet because I don’t have Yealink phone

    ZOIPER
    I can’t test because I don’t have Zoiper Business