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
-upgrade Firmware
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
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
-set event and notification
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
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
-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
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
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
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
-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
-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
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
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
-Define business hours
-Call Detail Records
open CDR and Recordings
-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
In this example, I will set lagu9 as a tone for incoming call from GSM
In this example, I will set lagu9 as a tone for incoming call from GSM
-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
On old U Series, you need to do these
- create 1 extension
2. connect through SIP Phone
3. connect through IAX Phone
-QueuesQueues 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
Basic SettingsNumberUse this number to dial into the queue, or transfer callers to this number to put them into the queueNameGive this queue a brief name to help you identify itPasswordYou can require agents to enter a password before they can login to this queue.Ring StrategyThis 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 DestinationSet the failover destinationStatic AgentsThis 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 TimeoutThe 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.AgentAnnouncementAnnouncement played to the Agent prior to bridging in the callerWrap-up TimeHow 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 delayRing In UseIf set to “no”, unchecked, the queue will avoid sending calls to members whose device are known to be “in use”RetryThe 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 SettingsMusic On HoldSelect the “Music on Hold” playlist for this QueueCaller Max Wait TimeSelect 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 unlimitedLeave When EmptyIf enabled, callers already on hold will be forced out of a queue when no agents availableJoin EmptyIf enabled, callers can join a queue that has no agentsJoin AnnouncementAnnouncement played to callers once prior to joining the queueAnnounce PositionAnnounce position of caller in the queueAnnounce Hold TimeEnabling 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.FrequencyHow often to announce queue position and estimated hold timePromptSelect a prompt file to play periodically-Music on Holddownload and install audacity from http://audacity.sourceforge.netdownload a mp3run audacity and click File/Import/Audio and choose a mp3click Tracks>Stereo Track to Monoclick Tracks>Resampleset 8000Hzclick File>Export AudioFile name: must have .wavSave as type: Other uncompressed filesHeader: WAV (Microsoft)Encoding: U-Law
open Yeastar GUI
-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”
-Tone RegionIf 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 ExtensionThis 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
Nbctcp's Weblog 