Knowledge Base

Informational

Q. What is VoIP?
A. VOIP is an acronym for Voice over Internet Protocol, or in more common terms phone service over the Internet. If you have a reasonable quality Internet connection you can get phone service delivered through your Internet connection instead of from your local phone company.

Q. What is an IP PBX?
A. PBX (Private Branch Exchange) is a system that connects telephone extensions of a company to outside public telephone network as well as to mobile networks. An IP (Internet Protocol) PBX is a PBX that provides audio, video, and instant messaging communication through the TCP/IP protocol stacks for its internal network and interconnects its internal network with the PSTN (Public Switched Telephone Network) for telephony communication.

Q. What is an IP Phone?
A. IP Phones are the same thing as VoIP Phones or Soft Phones. These are telephones that use VoIP technologies for making calls over an IP Network or the traditional PSTN networks.

Q. What is SIP? 
A. SIP (Session Initiation Protocol) is a signaling communications protocol, widely used for controlling multimedia communication sessions such as voice and video calls over IP networks.

Q. What is IAX?
A. IAX (Inter-Asterisk Exchange) is a communications protocol native to the Asterisk PBX software, and is supported by a few other soft switches, PBX systems, and soft phones. It is used for transporting VoIP telephony sessions between servers and to terminal devices. IAX now most commonly refers to IAX2, the second version of the IAX protocol.

Q. Comparison between SIP and IAX2
A. IAX2 protocol carries both signaling and media on the same port 4569 (by default). SIP signaling port is 5060 (by default), but media goes through other random ports. For IAX2 you only have to make sure port 4569 is clear then the phone calls can be guaranteed, but for SIP, besides port 5060 also other RTP ports need to be guaranteed too for stability of communication. Otherwise you'll encounter no audio or one-way voice problems, especially between different networks.
From the functionality perspective, IAX2 cannot support variety functions like SIP. SIP is the most widely used VoIP protocol so IP phone manufacturers are willing to develop call features based on SIP instead of IAX2.

Q. What is T.38 fax?
A. T.38 is an ITU recommendation for allowing transmission of fax over IP networks in real time. ZYCOO® IP PBX supports T.38 fax pass through.

Q. What is Distinctive Ringtone?
A. Some IP Phones provide different ring tones based on call source (internal or external) and there's text field to enter Alert-Info so that distinctive ring can be used for that destination.

Q. What is DISA?
A. DISA (Direct Inward System Access) is a mechanism available on some PBX’s that permits inbound calls to be answered and immediately presented with system dial tone. The caller is then able to dial a number that the PBX uses to decide how to forward the call. It is like the caller is able to dial twice – first to reach the PBX, then a second time to reach the final destination using the facilities of the PBX.

Q. What is Smart DID?
A. Define patterns to match the numbers you call, then those numbers will be stored into Asterisk database and be associated with the extensions which made the calls. If those numbers call in again, the calls will automatically call on the extensions that associated previously.

Q. What is BLF? Does ZYCOO® IP PBX support it?
A. BLF (Busy Lamp Field), typically a collection of lights or indicators on a phone that indicate who is talking on other phones connected to the same PBX. Used by a receptionist or secretary to aid in routing incoming calls.
Yes, ZYCOO® IP PBX supports BLF by default, and you don’t have to configure on the IP PBX, just configure the BLF key on the IP phone.

Q. What is FXO lifeline to FXS? Does ZYCOO® IP PBX support this feature?
A. FXO lifeline to FXS is a feature that provided by our FXOS and 2FXOS modules, the FXO port can allow a PSTN call to ring/call the phone connected to the FXS port on the same module so it serves as a life line in case of power outage. If the mentioned modules are installed then that IP PBX can support this feature.

Q. What is call parking? How does it work?
A. Call parking is a feature of Asterisk that allows a user to put a call on hold at one extension and continue the conversation from any other extensions.
The number used to park a call on ZYCOO® IP PBX is 700, and the parking extensions are configured from 701 to 720. While in a conversation, press *2 or transfer button on the phone to initiate a transfer, then dial 700.Asterisk will announce the parking extension, most probably 701 or 702. Now hang up - the caller will be left on hold at the announced extension. Walk up to a different phone, dial the extension number announced- the conversation can be continued. If a caller has been parked for a longer time than the specified time limit then Asterisk will again ring the originally dialed extension.

Q. What is Phone Provisioning?
A. Phone provisioning or auto provisioning is an easy and time-saving way to configure IP phones for IP PBXs. With phone provisioning, all user information can be entered via the IP PBX web interface. Required is the MAC address of the IP Phone, the desired extension and the caller ID which is displayed on the called party phone display. The IP Phone receives the configuration via the local network. This offers many advantages, i.e. it is possible to configure phones centralized, which saves a lot of time and money.

Q. How do I configure phone provisioning with ZYCOO® IP PBX?
A. First, you need to enable DHCP service on ZYCOO® IP PBX, and please make sure there’s no other DHCP service in your subnet. Also please fill in TFTP server address with ZYCOO® IP PBX IP address if you don’t use other TFTP server.
Now please configure phone settings, choose manufacturer of the phone and the model, then fill in MAC address and choose an extension number for this phone.
After all done, please attach the IP phones to your switch and power them on then they can be auto provisioned.

Q. What is PnP? How does it work?
A. PnP refers to plug and play of the IP phones, it's another more efficient way of provisioning the IP phones, just enable this feature and power on the IP phones then they can be provisioned.

Q. What does PIN Set do in a dial rule?
A. For example, this rule you configured that can dial international numbers, you don't want everyone to use this dial rule, you can enable PIN Set, so if someone want to call out using this dial rule, he/she needs to enter one of the PIN codes in the PIN Set first, then he/she can call out. Also the PIN number can be traced in the call logs.

Q. What are the advantages of an IP PBX comparing to a legacy PBX?
A. See below:
1. No more phone wiring, it uses you LAN
2. Much easier of installation and management
3. Many more features but much cheaper than a legacy PBX
4. Remote extensions and branch offices, saving your long distance telephone expense
5. Significant cost savings using VoIP providers



CooVox Series IP PBX
 

Q. Which Asterisk version are CooVox series based on?
A. CooVox series are based on Asterisk 1.8.

Q. What's new on CooVox series compared to ZX series? 
A. See below: 
1. Enhanced hardware
2. Newly designed dismountable modules (except U20)
3. More PBX features, such as FOP (Flash Operator Panel), Smart DID, Call Back, Virtual Fax and so on
4. More security features, UDP/TCP/TLS SIP, SRTP, firewall, Web Manager and so on
5. More network features, IPv6, VLAN, VPN Server, TR069 and so on
6. USB storage supported, Hardware Echo Cancellation module supported (U50 and U100)

Q. Which IP Phones can be used with ZYCOO® IP PBX?
A. All ZYCOO® IP Phones can work perfectly with our IP PBX, as well as other IP Phones that support standard SIP/IAX2 protocol, e.g.: CISCO, Grandstream, Yealink, Polycom, Siemens, Snom, etc.

Q. Which modules can I choose to install on U20?
A. FXO-200, FXS-200, FXOS, 1GSM/3G UMTS modules

Q. Can I install echo cancellation module on U20?
A. Echo cancellation modules can be only installed on U50 and U100.

Q. Which modules can I choose to install on U50?
A. 4FXO, 4FXS, 2FXOS, 2GSM, 4GSM, 1PRI, 4BRI, hardware echo cancellation, 3G UMTS modules

Q. Which modules can be installed on Slot1 of U50? What about Slot2?
A. Available modules on each slot: 
Slot1: FXO/FXS/GSM/PRI/BRI modules
Slot2: FXO/FXS/GSM modules

Q. Which modules can be installed on U100?
A. 4FXO, 4FXS, 2FXOS, 2GSM, 4GSM, 1PRI, 4BRI, hardware echo cancellation, 3G UMTS modules

Q. Can I load my customized firmware on U20/U50?
A. All applications and drivers need to be compiled under our developing environment so you cannot load your own firmware to U20/U50.

Q. What system can be installed on U100/Asterisk Appliance P2 platform?
A. You can choose to install ZYCOO® system, Elastix and also other Linux distribution with Asterisk installed.

Q. How do I register IAX2 extensions?
A. We do not enable IAX2 protocol for the extensions by default, if you want to register IAX2 extensions you need to create new extension numbers with IAX2 protocol enabled, then you can register.



D30 IP Phone
 

Q. How many SIP lines can be supported?
A. D30 can register 2 SIP lines at the same time.

Q. Does D30 support IAX2 protocol?
A. Yes, D30 supports IAX2 protocol, and there’s IAX2 line key to choose this line to make phone calls.

Q. Does D30 support BLF?
A. No, D30 cannot support BLF.

Q. Does D30 support PoE?
A. D30 does not support PoE but D30P does.

Q. How do I make a 3-way conference call on D30? 
A. See below: 
1. Press softkey Conf during an active call.
2. The first call is placed on hold. Then you will hear a dial tone. Dial the number to conference in, then press Send key.
3. When the call is answered, press Conf and add the first call to the conference.
4. If you want to release the conference, press Split key.



D60 IP Phone
 

Q. How many SIP lines can be supported?
A. D60 can register 6 SIP lines, 4 line keys can be configured.

Q. Does D60 support IAX2 protocol?
A. Yes, D60 supports IAX2 protocol.

Q. Does D60 support BLF?
A. Yes, D60 supports BLF, there are 8 programmable keys can be configured as BLF key, also you can connect CEP-26 extension module to D60 to extend the BLF keys, one CEP-26 can provide 26 programmable DSS keys.

Q. Does D60 support PoE?
A. D60 supports PoE by default.

Q. How can I synchronize D60 time with my CooVox series IP PBX?
A. You need to configure primary SNTP server with the IP PBX IP address on web page NETWORK-->TIME&DATE.

Q. How do I make a 3-way conference call on D60? 
A. See below: 
1. Press the Conf softkey during an active call.
2. The first call is placed on hold. Then you will hear a dial tone. Dial the number to conference in, then press Send key.
3. When the call is answered, press Conf and add the first call to the conference.
4. If you want to release the conference, press Split key.



Others
 

Q. How does Email to Fax work?
A. First of all, an Email account needs to be configured on the IP PBX with access code. Then others can send Email to this mail address with Fax file attached, and the mail title need to be the format:
access_code=fax_number
Then the built in mail server will check this account's mail, if there're emails with such title, the IP PBX will download the attachments and send fax to the fax number specified.

Q. Can I forward a call to multiple destinations?
A. Yes, on our CooVox series IP PBX you can define a list of numbers for follow feature, and the call will be forwarded to the numbers listed one by one, till it's been answered.

Q. Can I change the FXS port to FXO or FXO to FXS port by software?
A. No, it cannot be changed; it’s the hardware that decides the port type, not software.

Q. Which languages of voice prompts are available?
A. Chinese, English, French, Italian, Portuguese and Spanish are built-in by default; also other voice prompts can be implemented.

Q. If I configure call recording for an extensions it will record all the phone call, but I don’t want all phone calls been recorded, how can I choose which phone calls to be recorded?
A. You can decide which conversation to be recorded during the call, just press feature code *1 then this call can be recorded; we call it one touch recording and is available on our CooVox series IP PBXs.

Q. How can I make my IP PBX automatically reboot every day?
A. You need to access the IP PBX via SSH and issue command:
crontab -e
and at the bottom of the file you need to add a new line:
0 6 * * * /sbin/reboot
Then it will reboot the IP PBX every day at 6:00 am automatically.

Q. How do I reset my U20/U50? 
A. See below:
1. While plugging the power cord, please press RST button till SYS led off then it will be reset to factory defaults. This method resets the whole system.
2. While system is running, SYS led blinks once in 2s. By pressing RST button for over 5 seconds till SYS led turn green and not blinking, and then all configurations done via web interface will be reset to factory defaults.

Q. How could I know a call to my extension that was called my extension directly or called me on the ring group?
A. You can define a label for the ring group. For example, you are the member of the sales ring group, and the label had been defined as “sales”; then if someone calls in and falls on this ring group, your extension rings and it will display sales and the caller number at the same time.

Q. How do I change root password?
A. Just execute command passwd then it will ask for new password, please make sure you can remember new root password or the only way to retrieve root password is reset.



Troubleshooting
 

Q. I can call external number but why internal phone calls are not working? 
A. Please check your dial rules, you need to avoid dial rules listed below:
_. (1 or more than 1 digits)
_X. (2 or more than 2 digits)
_XX. (3 or more than 3 digits)
Cause they would send all numbers even less than 3 digits to the trunks.

Q. Why do I have one-way voice on remote extensions or remote IP PBX? 
A. Usually the one-way voice problem on remote extensions or remote IP PBX is caused by NAT, you need to check NAT Support Settings on page Advanced-->Global SIP Settings. External Host and External IP need to be configured with your public IP address or DDNS domain, and make sure the Local Network is correctly pointed to your subnet.

Q. Why do I have one-way voice on the analogue trunks?
A. Please check if you had enabled polarity switch option on the analogue trunk advanced settings.

Q. Why can I make phone calls from my IP phone but others cannot call me?
A. Maybe you had enabled DND for your extension, please disable DND on the phone or by feature code *074 of our IP PBX.

Q. Why is my customized wave audio file not working?
A. Please make the sound files in mono, 16 bit, 8000 Hz.

Q. Why don't I have Caller ID on the phone calls from PSTN lines?
A. Our IP PBX can support Caller ID signaling Bellcore FSK, DTMF, V23 and V23 Japan. Please ask your Telco what signaling is applied on the PSTN lines and then configure Caller ID signaling on web page Advanced-->Global Analog Settings.