C56 VoIP Phone User Manual Cortelco 1703 Sawyer Road Corinth, MS 38834 USA www.cortelco.com Tel: (662)287-5281 Fax:(662)287-3889 Version 1.
Safety Notices Please read the following safety notices before installing or using this phone. They are crucial for the safe and reliable operation of the device. Please use the external power supply that is included in the package. Other power supplies may cause damage to the device, affect the behavior or induce noise. Before using the external power supply, please be sure it is for use with your power voltage. Incorrect power voltage may cause fire and damage.
Table of Contents 1 INTRODUCING C56 VOIP PHONE............................................................................... 5 1.1 THANK YOU .......................................................................................................................... 5 1.2 BOX CONTENTS .................................................................................................................... 5 1.3 KEYPAD ...........................................................................................
5.3.1.2.3 PPPoE ................................................................................................................................ 16 5.3.1.2.4 Quick SIP Settings ............................................................................................................. 17 5.3.1.3 Call Log ............................................................................................................................. 18 5.3.2 Network .......................................................
1 Introducing C56 VoIP Phone 1.1 Thank you Thank you for purchasing the C56(P) Voice Over Internet Protocol (VoIP) telephone. The C56(P) is a fully featured telephone that provides voice communication over the data network. This phone has all the features of a traditional telephone and gives access to many data service features. This guide will help you easily use the various features and services available on your phone. 1.2 Box Contents The following items should be packed with your telephone.
Redial Speaker phone Indicator light When off hook, this will dial the last called number. In stand-by mode, it will check the Outgoing Call. Activate speakerphone mode. This light blinks to indicate a missed call. SYSINFO Displays phone settings such as phone number, IP address, gateway address, etc. ENTER Used to enter next menu or confirm settings MWI Accesses voice mail system. TRANSFER Performs blind or attended transfers. See Section 3.1.4 for more details.
1.4 Input/Output Ports Port Port name Description Power switch Input: 5V AC, 1A WAN 10/100M Connect to Network LAN 10/100M Connect to PC Handset Port type: RJ-9 connector 2 Initial Connection and Setting 2.1 Connecting the phone 1. Connect to the network. Use the Ethernet cable in the package to connect the WAN port on the back of your phone to an Ethernet port. The following two figures show connection options. 2.
4. The phone’s LCD screen displays “WAIT LOGON”. Later, a ready screen displays the date, time and current network mode. If your LCD screen displays different information from the above, more information may need to be entered. Please refer to the next section. If your phone registers into your IP telephony Server, it is ready to use. If not, continue to read for more configuration information. 2.2 Network Settings DHCP is supported by default.
to switch to PPPoE mode. When the PPPoE icon at the top of the LCD stops blinking, the mode change is complete. 2.2.2 1. 2. 3. 4. 5. 6. 7. 8. 9. 10. 11. 12. 13. 14. 15. 16. 17. 18. 19. 20. 21. 22. 23. 24. 25. 26. 27. 28. 29. 30. 31. 32. 33. 34. 35. 36. 37. 38. 39. 40. Static IP Mode Press and hold 1 for three seconds. Press ENTER to confirm. Press ENTER. The LCD will display “INPUT PASSWORD”. Input the password (default is 123). Press ENTER. The LCD will display” NETWORK”. Press ENTER.
41. 42. 43. 44. 45. 2.2.3 Press ENTER. The LCD will display “ARE YOU SURE”. Press ENTER. The LCD will display “SAVING NOW” and then display “SAVE”. Press EXIT twice. Press and hold 1 until the LCD displays “ARE YOU SURE”. Press ENTER. The LCD will display “CHANGING”. This means the phone is trying to switch to static IP mode. When the STATIC icon at the top of the LCD stops blinking, the mode change is complete. DHCP Mode 1. Press and hold 2 until the LCD displays “ARE YOU SURE”. 2. Press ENTER.
confirming it with the # button. 3. Pressing the HOLD button during the second call will resume the first call. 3.4 Call Waiting 1. When a third party calls during an established call, the LCD will display the incoming call number. Press the HOLD key to place the established call on hold and answer the incoming call. 2. Press # to hang up the established call and answer the incoming call. NOTE: Call Waiting service must be enabled. 3.5 Call transfer 3.5.
4.2 MWI(Message Waiting Indication) This LED will flash to indicate a new voicemail. Pressing the MWI key will access the voicemail if the key has been configured correctly. 4.3 Redial / Unredial If B is on a call when A calls, A will get busy tone. If A wants to connect to B as soon as B is available, he can use the redial function. To use this feature, A dials a prefix and then B’s number. When the redial function is activated, A will check B’s calling status every 60 seconds.
5.1.2 Password Configuration There are two levels of access: root level and general level. A user with root level access can browse and set all configuration parameters, while a user with general level can set all configuration parameters except server parameters for SIP or IAX2. Default user with general level: Username: guest Password: guest Default user with root level: Username: admin Password: admin The default password for the phone screen menu is 123. 5.
5.3 Configuration via WEB 5.3.1 BASIC 5.3.1.1 Status Field Name Network Accounts Explanation Shows the configuration information for WAN and LAN port, including connection mode of WAN port (Static, DHCP, PPPoE), MAC address, IP address of WAN port and LAN port, DHCP server status for LAN port (ENABLED or DISABLED). Shows the phone numbers and registration status for the 2 SIP LINES.
5.3.1.2 Wizard Select the appropriate network mode. The phone supports three network modes: 1 Static: The parameters of a Static IP connection must be provided by your ISP. 2 DHCP: In this mode, network parameter information will be obtained automatically from a DHCP server. 3 PPPoE: In this mode, you must enter your ADSL account and password. Refer to Section 2.2 for detailed information about configuring the network parameters.
5.3.1.2.1 Static IP If Static IP is selected, this screen will be displayed. Information provided by the ISP should be entered. Click Back to return to the Wizard screen. Click Next to go to Quick SIP Settings 5.3.1.2.2 DHCP After selecting DHCP and clicking NEXT, the Quick SIP Settings screen will appear. Click Back to return to the Wizard screen. Click Next to go to the Summary screen. 5.3.1.2.3 PPPoE If PPPoE is selected, this screen will appear. Enter the information provided by the ISP.
5.3.1.2.4 Quick SIP Settings Field Name Explanation Display Name The name shown in caller ID. Server Address SIP server address either IP address or URI. Server Port SIP server port (usually 5060). Authentication User Login name or Authentication ID. Authentication Password SIP password. SIP User Phone number. Enable Registration Submits registration information. Normally checked. Click Back to return to the IP Address screen. Click Next to see summary screen.
5.3.1.3 Call Log Outgoing call logs can be seen on this page. Field Name Start Time Duration Dialed Calls Explanation Start time of the outgoing call Duration of the outgoing call. Account, protocol, and line of the outgoing call. 5.3.2 Network 5.3.2.1 WAN Config Field Name Active IP Address Current Subnet Mask Current IP Gateway MAC Address MAC Timestamp Explanation The current IP address of the phone. The current Subnet Mask. The current Gateway IP address. The MAC address of the phone.
5.3.2.1.1 Static IP If Static IP is chosen, the screen below will appear. Enter values provided by the ISP. 5.3.2.1.2 DHCP If DHCP is chosen, all configuration information will be provided by a DHCP server. Contact the ISP to determine if DHCP is used. 5.3.2.1.3 PPPoE If PPPoE is chosen, the screen below will appear. Enter the information provided by the ISP.
Chart 1 shows a network switch with no VLAN. Any broadcast frames will be transmitted to all other ports. For example, and frames broadcast from Port 1 will be sent to Ports 2, 3, and 4. Chart 2 shows an example with two VLANs indicated by red and blue. In this example, frames broadcast from Port 1 will only go to Port 2 since Ports 3 and 4 are in a different VLAN. VLANs can be used to divide a network by restricting the transmission of broadcast frames.
Field Name Explanation Enable LLDP Packet Interval Enable or Disable Link Layer Discovery Protocol (LLDP) The time interval for sending LLDP Packets Enable Learning Function Enables the telephone to synchronize its VLAN data with the Network Switch. The telephone will automatically synchronize DSCP, 802.1p, and VLAN ID values even if these values differ from those provided by the LLDP server.
5.3.2.3 Service Port Set the port values for Telnet/HTTP/RTP on this page. Field Name Web Server Type HTTP Port Telnet Port RTP Port Range Start RTP Port Quantity Explanation Specify Web Server Type – HTTP or HTTPS Port for web browser access. Default value is 80. To enhance security, change this from the default. Setting this port to 0 will disable HTTP access. Example: The IP address is 192.168.1.70 and the port value is 8090, the accessing address is http://192.168.1.70:8090. Port for Telnet access.
Field Name Enable SNTP Enable DHCP Time Primary Server Secondary Server Time Zone Resync Period 12 -Hour Clock Date Format Date Separator Enable Offset(minutes) Month Explanation Simple Network Time Protocol (SNTP) Settings Enable or Disable SNTP If this is enabled, phone will synchronize time with DHCP server. IP address of Primary SNTP Server IP address of Secondary SNTP Server Local Time Zone Time between resync to SNTP server. Default is 60 seconds. If checked, clock is 12 hour mode.
Week Day Hour Minute Start and end week for DST Start and end day for DST Start and end hour for DST Start and end minute for DST Manual Time Settings Enter the values for the current year, month, day, hour and minute. All values are required. Note: Be sure to disable SNTP service before entering manual time and date. 5.3.3 VOIP 5.3.3.1 SIP Configuration Configure a SIP server on this page.
Field Name Explanation Choose the sip line to configured (SIP 1 – SIP 2). Click the dropdown arrow to select the line. Status Shows registration status. Will show “Registered” if registered or “Unapplied” if not registered. Server Address SIP server IP address or URI. Server Port SIP server port. Default is 5060. Authentication User SIP account name (Login ID). Authentication Password SIP registration password. SIP User Phone number assigned by VoIP service provider.
Display Name Enable Registration Domain Realm Proxy Server Address Proxy Server Port Proxy User Proxy Password Backup Server Address Backup Server Port Server Name Set the display name. This name is shown on Caller ID. Check to submit registration information. SIP Domain if different than the SIP Registrar Server. SIP proxy server IP address or URI（This is normally the same as the SIP Registrar Server) SIP Proxy server port. Normally 5060. SIP Proxy server account. SIP Proxy server password.
Subscribe For MWI MWI Number Subscribe Period Conference Type Conference Number Registration Expires Enable Service Code DND On Code DND Off Code Always CFwd On Code Always CFwd Off Code Busy CFwd On Code Busy CFwd Off Code No Ans. CFwd On Code No Ans.
Local port Ring type Enable Rport Enable PRACK Enable Long Contact Convert URI Dial Without Registered Ban Anonymous Call Enable DNS SRV Enable Missed Call Log BLF List Number Enable BLF List Server Type RFC Protocol Edition Transport Protocol Anonymous Call Edition Keep Authentication Ans. With a Single Codec Auto TCP Enable Strict Proxy Enable GRUU Enable Displayname Quote Enable user=phone Click to Talk Strict Branch Enable Group Different VoIP Service providers may require different modes. SIP port.
Registration Failure Retry Time 5.3.3.2 Registration failure retry time – If registration fails, the phone will attempt to register again after registration failure retry time. This will affect all lines STUN Config STUN support is configured in this page. STUN – Simple Traversal of UDP through NAT – A STUN server allows a phone in a private network to know its public IP and port as well as the type of NAT being used.
Field Name STUN NAT Transversal Server Address Server Port Binding Period SIP Waiting Time SIP Line Using STUN Use STUN 5.3.3.3 Explanation Shows whether or not STUN NAT Transversal was successful. STUN Server IP address STUN Server Port – Default is 3478. STUN blinding period – STUN packets are sent at this interval to keep the NAT mapping active. Waiting time for SIP. This will vary depending on the network.
Field Name Phone number Destination Port Alias Explanation There are two types of matching: Full Matching or Prefix Matching. In Full matching, the entire phone number is entered and then mapped per the Dial Peer rules. In prefix matching, only part of the number is entered followed by T. The mapping with then take place whenever these digits are dialed. Prefix mode supports a maximum of 30 digits. Set Destination address. This is optional.
Delete Length Sets the number of characters to be deleted. For example, if this is set to 3, the phone will delete the first 3 digits of the phone number. This is optional. Dial Peer Examples Web Interface Explanation Set phone number, Destination, Alias and Delete Length. Phone number is XXXT; Destination is 255.255.255.255 (0.0.0.2) and Alias is del. Any phone number that begins with XXX will be sent via SIP2 after the first several digits are deleted depending on the delete length.
5.3.4 Phone 5.3.4.1 AUDIO This page configures audio parameters such as voice codec, handset volume, and ringer volume. Field Name First Codec Second Codec Third Codec Fourth Codec Fifth Codec Sixth codec Onhook Time Default Ring Type Handset Input Volume Handset Output Volume Speakerphone Volume Ring Volume G729 Payload Length Tone Standard G722 Timestamps Explanation The first codec choice: G.711A/u, G.722, G.723, G.729, G.726 The second codec choice: G.711A/u, G.722, G.723, G.729, G.
G723.1 Bit Rate Enable VAD DTMF Payload Type 5.3.4.2 Choices are 5.3kb/s or 6.3kb/s Enable or disable Voice Activity Detection (VAD). If VAD is enabled, G729 Payload length cannot be set greater than 20 mSec. The RTP Payload type that indicates DTMF. Default is 101 FEATURE This page configures various features such as Hotline, Call Transfer, Call Waiting, etc.
Field Name Do Not Disturb Enable Call Transfer Semi-Attended Transfer Enable Auto Handdown Auto Handdown Time Enable Auto Redial Auto Redial Interval Auto Redial Times Explanation If enabled, the phone will reject incoming calls. The callers receive busy tone. Outgoing calls may be made. If enabled, Call Transfer is allowed. If enabled, Semi-Attended Transfer is allowed. If enabled in speakerphone mode, the phone will automatically hang up and return to idle when the distant party terminates the call.
Enable Intercom Enable Intercom Tone P2P IP Prefix Turn Off Power Light Emergency Call Number Enable Password Dial Password Dial Prefix Password Dial Length Ban Outgoing Enable Call Waiting Enable 3-way Conference Accept Any Call Enable Call Completion Enable Pre-Dial Enable Silent Mode Hide DTMF Enable Intercom Mute Enable Intercom Barge If enabled, allows intercom calls. If enabled, plays intercom ring tone to alert to an intercom call. Set Prefix for peer to peer IP call.
DND Return Code Busy Return Code Reject Return Code Active URI Limit IP Push XML Server Enable Call Waiting Tone Action URL Settings Block Out Settings 5.3.4.3 outside call. If an intercom call is established, a second intercom call will be rejected. Specify SIP Code returned for DND. Default is 480 - Temporarily Not Available. Specify SIP Code returned for Busy. Default is 486 – Busy Here. Specify SIP Code returned for Rejected call. Default is 603 – Decline.
[] * . Tn Dial Plan Special Characters Specifies a range of digits to match. May be a range, a list of ranges separated by commas, or a list of digits. Match any single digit that is dialed. Match any arbitrary number of digits including none. A time out period before digits are sent of n seconds in length. n is mandatory and can have a value of 0 to 9 seconds. Tn must be the last 2 characters of a dial plan. If Tn is not specified it is assumed to be T0 by default on all dial plans.
5.3.4.4 CONTACT Enter the name, phone number and ring type for each contact here. Field Name Name Office Number Ring Type Name Office Number Ring Type Select File Explanation Phonebook Tables Contact name Contact phone numbers Ring type for this contact Add Contact Contact name Contact phone numbers Ring type for this contact Import Contact List Click the browse button to select the phonebook file to import. Then click the update button and the selected file will be added to the phone.
Export Contact File Export XML Export contacts to xml file. Export CSV Export contacts to csv file. Export VCF Export contacts to vcf file. Blacklist Settings Type Select the blacklist type - number or prefix Value Input number or prefix Line Select the sip line Note: The maximum capability of the phonebook is 500 contacts. Note: “x” and “.” are special characters in the black list. “x” matches any single digit and “.” matches any number of digits.
Memory Key – Select Type as Memory Key and enter the number to be dialed in the Value box. When the key is pressed, the phone will dial the programmed number. Key Event – Select Type as Key Event and then select the SubType from the following options: None Message Wait Indication (MWI) Do Not Disturb (DND) Hold Transfer Phone Book Redial Auto redial on Auto redial off Call Forward History Flash Headset Call Back 5.3.6 Maintenance 5.3.6.
Auto Provision Setting Field Name Explanation Current Config Version Show the current config file’s version. If the version of configuration downloaded is higher than this, the configuration will be upgraded. If the endpoints confirm the configuration by the Digest method, the configuration will not be upgraded unless it differs from the current configuration. Show the common config file’s version. If the configuration downloaded and this configuration are the same, the auto provision will stop.
Password Config Encryption Key Common Config Encryption Key If this is blank the phone will use anonymous. Password for configuration server. Used for FTP/HTTP/HTTPS. Encryption key for the configuration file Encryption key for common configuration file DHCP Option Settings Field Name DHCP Option Setting Custom DHCP Option Enable PnP PnP Server PnP Port PnP Transport PnP Interval Server Address Protocol Type Config File Name Update Interval Update Mode 5.3.6.
Level 0 1 2 3 4 5 6 7 Name Emergency Alert Critical Error Warning Notice Informational Debug Description System is unusable. This is the highest debug info level. Action must be taken immediately. Critical conditions. System is probably working incorrectly. Error conditions. System may not work correctly. Warning conditions. System may work correctly but needs attention. Normal but significant condition. Normal daily messages. Debug messages normally used by system designer.
Config Setting Field Name Save Configuration Backup Configuration Clear Configuration 5.3.6.4 Explanation Save the current phone configuration. Clicking this saves all configuration changes and makes them effective immediately. Save the phone configuration to a txt or xml file. Please note to Right click on the choice and then choose “Save Link As.” Logged in as Admin, this will restore factory default and remove all configuration information.
Type Action to be executed by the phone. 1. Application update - download system update file 1. Config file export - Upload config file to FTP/TFTP server. It can then be named and saved. 2. Config file import - Download the config file from FTP/TFTP server. The configuration will be effective after the phone is reset. 3. Phone book export (.vcf, .csv, .xml) - Upload the phonebook file to FTP/TFTP server. It can then be named and saved. 4. PhoneBook import (.vcf, .csv, .
Password Confirm Set the password Confirm the password User Management Select the account and click Modify to modify the selected account. Click Delete to delete the selected account. A General user can only add another General user. 5.3.6.6 Reboot Some configuration modifications require a reboot to become effective. button will cause the phone to reboot immediately. Note: Be sure to save the configuration before rebooting. 5.3.7 Security 5.3.7.
Add Add this filter range to the Web Filter Table Enable Web Filter Select to enable MMI Filter. Apply Make filter settings effective. Note: Once a range is added, it can be modified or deleted. Note: Be sure that the filter range includes the IP address of the configuration computer. 5.3.7.
Dest Address Src Mask Dest Mask Set destination address. It can be a single IP address or use * as a wild card. For example: 192.168.1.14 or *.*.*.14. Set the source address mask. For example: 255.255.255.255 points to one host while 255.255.255.0 points to a C type network. Set the destination address mask. For example: 255.255.255.255 points to one host while 255.255.255.0 points to a C type network. When a connected device tries to access 192.168.1.
6 Appendix 6.1 Specification 6.1.1 Hardware Item Power Adapter Specification LCD Size Operation Temperature Relative Humidity CPU SDRAM Input: 100-240V Output: 5V 1A 10/100Base- T RJ-45 1 PORT 10/100Base- T RJ-45 1 PORT Idle: 1.5W Active: 1.8W 74x28mm 0～40℃ 10～65% Broadcom 8MB Flash 2MB Dimension(L x W x H) Weight 20 X 18.5 X 19.3cm 0.99kg Port WAN LAN Power Consumption 6.1.2 Voice Features Supports 2 SIP servers Supports RFC3261 Codecs G.711A/u G.723.1 high/low G.729a/b G.
DNS Peer to Peer/ IP call Automatic line selection 9 Standard ring tones DTMF SIP info DTMF Relay (In-Band) RFC2833 AUTO SIP applications Call Forward Call Transfer（Blind/Attended） Hold Call Waiting 3 Way Conference Redial paging Intercom Auto Redial Call control features Flexible dial plan Hotline Anonymous Call Reject Black List (Reject Authenticated Call) Approved Caller List Limit Call Do Not Disturb Caller ID CLIR (reject anony
6.1.3 Network Features WAN/LAN Bridge Bridge with port mirror Supports PPPoE for xDSL Supports VLAN 802.1Q 802.1P Supports STUN Wan Port Supports Main DNS and Secondary DNS Supports DNS via DHCP or Static DNS Supports DHCP client on WAN QoS with DiffServ Network Tools in Telnet Server Ping Trace Route Telnet Client 6.1.
6.2 Digit-character map table Keypad Character Keypad 1@ Character 7PQRSpqrs 2ABC ab c 8TUVtuv 3DEFdef 9WXYZwxyz 4GHIghi .
