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DPH-80 Phone User Manual   

 

Configuring

 the SIP phone 

 

 

Upload Filename (up to 6 characters): 

This is the filename to upload the configuration 

parameters from the phone to the TFTP server. It may be 6 characters long at 

maximum. It should start with a letter and should consist of digits, letters and 
underscore.  

Download Filename (up to 6 characters): 

This is the filename to download the 

configuration parameters from the TFTP server to the phone. It may be 6 characters 
long at maximum. It should start with a letter and should consist of digits, letters and 
underscore.  

Adaptive Jitter: 

If this is enabled then Jitter Buffer will be adaptive, otherwise it will use 

a fixed buffer of a size specified in 

Maximum Buffer Size.

Maximum Buffer Size: 

If adaptive jitter is disabled, the phone will use this static value 

for Jitter Buffer size. This should be in the range of 0-300 in ms. 

Log Server:

 This flag is turned on in case the user wants to log all debug messages for 

viewing.  

Log Server Address: 

This has the IP address of the machine where all the log messages 

should be sent. It must be in dot-separated form. An illegal IP address won't be allowed 

for this field. 

Log Server Port:

 This is the port number on the log server to which the log messages are 

to be sent. It should be a valid port number in the range of 0-65335. The user should 

make sure that it is not one of the reserved port numbers. 

Microphone Gain:

 This will show the microphone gain in the range of -14 to 14 (unit of 

dB) 

Speaker Gain: 

This will show the speaker gain in the range of -14 to 14 (unit of dB) 

Access Settings:

 The following three key sequences should be unique. 

Factory Default: 

This is the key sequence the user should dial on the phone to get the 

phone to use all the default values of the parameters. After entering this key sequence 

on the SIP phone it will restore the parameters to default upon next restart. 

Production Key: 

This is the key sequence the user should dial on the phone to get to 

production-test mode. After entering this key sequence, the SIP phone will start in 
production-test mode upon next restart. Reboot after the production test is complete to 

start functioning in the SIP phone mode. 

TFTP upload: 

This is the key sequence the user should dial on the phone to start the 

TFTP software update. After getting the new image, the phone will start itself using the 

new image. 

Click on SIP Configuration. 

 

 16

Summary of Contents for DPH-80

Page 1: ...DPH 80 IP Phone User s Guide...

Page 2: ...as a copy of the dated purchase invoice must be provided If Purchaser s circumstances require special handling of warranty correction then at the time of requesting RMA number Purchaser may also prop...

Page 3: ...e D Link s obligation under this warranty shall be a reasonable effort to provide compatibility but D Link shall have no obligation to provide compatibility when there is fault in the third party hard...

Page 4: ...HAS BEEN ADVISED OF THE POSSIBILITY OF SUCH DAMAGE IF YOU PURCHASED A D LINK PRODUCT IN THE UNITED STATES SOME STATES DO NOT ALLOW THE LIMITATION OR EXCLUSION OF LIABILITY FOR INCIDENTAL OR CONSEQUEN...

Page 5: ...OKMARK NOT DEFINED MGCP TROUBLESHOOTING ERROR BOOKMARK NOT DEFINED MGCP PRODUCTION TEST ERROR BOOKMARK NOT DEFINED DPH 80 SIP SESSION INITIATION PROTOCOL 12 IP PHONE CONFIGURATION 12 INFRASTRUCTURE RE...

Page 6: ...al and long distance voice services and data services over a single network connection This easy to use IP Phone simply plugs right into the local area network through a standard RJ45 interface The DP...

Page 7: ...a Gateway Control Protocol MGCP Session Initiation Protocol SIP and the H 323 protocol These protocols are used for signaling maintaining and tearing down voice calls The D Link IP phone allows voice...

Page 8: ...power adapter into the appropriate wall outlet 3 Plug the power adapter plug into the power jack Basic Configuration IP Address In order to use a Web browser to configure the DPH 80 IP phone you must...

Page 9: ...DPH 80 Phone User Manual Basic Configuration In the General Tab click on Internet Protocol TCP IP 8...

Page 10: ...0 1 1 xx range Do not use the IP address 10 1 1 80 that address is already in use by the DPH 80 as a default address If necessary change the Subnet mask and the Default gateway to match the values sho...

Page 11: ...the three main Internet telephony protocols MCGP SIP and H 323 Please choose the appropriate protocol and follow the directions indicated in the Web based configuration utility Appendix_A MGCP Media...

Page 12: ......

Page 13: ...support SIP to make and receive calls from outside your LAN network A TFTP server is required to support remote software upgrading The TFTP server should have the two software image files dph80v1 tfp...

Page 14: ...r both will be dlink These values can be changed later using the Change Login Name and Password Page User Name This is case insensitive with a maximum of 20 characters Password This is case sensitive...

Page 15: ...f ff ff ff ff It can be modified once to a non default value Once modified it will be grayed out and cannot be changed again Country Code This is a drop down menu Select the appropriate country This f...

Page 16: ...allowed for this field Net Mask This will have the Net Mask of the network to which the IP phone is connected It must be in dot separated form An illegal IP address mask won t be allowed for this fie...

Page 17: ...ed form An illegal IP address won t be allowed for this field Log Server Port This is the port number on the log server to which the log messages are to be sent It should be a valid port number in the...

Page 18: ...e Phone Number This will store any character string up to 20 characters long Phone Port This is the port number at which the phone will open the socket to send and receive SIP messages Proxy Server Ad...

Page 19: ...odec2 and Codec3 are drop down menus This allows selecting what codecs to be used by the phone It also specifies the priority of the codec while negotiating for the codec to use in any call Codec1 wil...

Page 20: ...st Number of packets that have been lost in the network Data Under Run Count This is the jitter buffer under run count for the entire call Maximum Jitter This is the estimated maximum jitter in the ne...

Page 21: ...DPH 80 Phone User Manual SIP Configuration On clicking Yes the following screen will appear You will be returned to the main page Configuration Upload 20...

Page 22: ...one to the TFTP server as the upload filename The TFTP server and filename are set in the General Configuration Click no on the warning page to return to the main page Configuration Download When clic...

Page 23: ...login user name it should be entered here Otherwise enter the same user name This is case insensitive and may be 20 characters long at maximum Old Password This is the login password used to access th...

Page 24: ...lick yes to save all the updated parameters to the flash memory and restart the phone so that the latest changes take effect The You have been successfully logged out page will be displayed The phone...

Page 25: ...the server any other user accessing the MGCP phone s configuration will get the Server Busy page and will not be allowed access Using the SIP Phone If the SIP phone is configured properly and if the...

Page 26: ...one plays dial tone Then dial the new party s number to transfer the call The SIP phone transfers the call and plays busy tone Flashing the hook twice before dialing the number will restore the call t...

Page 27: ...p The phone will recover on detecting a SIP server Error tone if the called party is not registered SIP Troubleshooting Some of the common error situations are described below Power up No tone on powe...

Page 28: ...It the browser is idle for more than 10 minutes the SIP phone will terminate the session You must restart the browser Other functions Entered factory default key sequence no response You must restart...

Page 29: ...erly In this test the user must speak into the microphone and that is heard after some finite delay in both the handset and speaker simultaneously This goes on till the user doesn t interrupt the test...

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