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

-Distinctive Ringtone

-BLF (Busy Lamp Field)

-DNIS (Dialed Number Identification Service)

> core set debug 2
> core set debug 0
> 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): 13
BER value (0,7): 0
SIM IMSI: 510890170771570
SIM SMS Center Number: +6289644000001

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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-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

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

SIP Server Auto Provisioning

I want to have SIP Server name or ip address automatically provided by DHCP server into SIP Client for example IPPI, Zoiper or other IP Phone

First we need to prepare for example Windows 2008R2 DHCP Server
-click Start/Administrative Tools/DHCP
-right click IPv4/Set Predefined Options
-click Add
follow below

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-right click Scope Options/Configure Options
-tick 120 SIP Server
SIP Client Settings
  1. IPPI
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  1. Zoiper
    I can’t test because only available for Zoiper Business