Dyna E30 Manuale d'uso - Pagina 9

Sfoglia online o scarica il pdf Manuale d'uso per Telefono Dyna E30. Dyna E30 17. Wifi phone

6. The functions of configuration item "Options" menu are as follows:
Use - use the configuration item
Change - set the configuration item
Rename - rename the configuration item
Note: 1. STUN protocol is set in SIP server currently; you needn't to set in your handset again. If
the SIP you registered doesn't support STUN, you can search for it on internet and set in
your handset.
2. The port number of VOIP protocol is default; you needn't to set in your handset again
except your operator provides you another one. To set it, enter the symbol ":" after the
SIP sever address, and then input the port number.
3. This edition doesn't support voice mail now.

Dial/Answer Call

1. Enter the integrity number, or you can search the number from the phonebook; or simply click
dial key to recall the recent phone number.
2. Press the Answer key to call the number; you can press the up/down key or left/right key to set
the volume of the speaker and microphone.
3. Press Answer key to answer a call, to reject a call press the End key.
Call Hold:
Call Hold is that users on a call can temporarily interrupt the original call (after call hold the
phone is in silent mode). To restore the original call, the user just needs to release the call hold
function,
Hold: Call status > left soft key > Call Options >1 Hold
Unhold: right soft key can release hold directly (or click the left soft key) > 1 Unhold
Call Mute:
Call Mute is that the user which selects Call Mute can only receive but not call.
Mute: Call status > left soft key > Call Options > 2 Mute
Unmute: right soft key can release mute directly (or click the left soft key) > 2 Unmute
Call Transfer:
Call Transfer is that the users on a call can transfer the current call to the online users on
third-party server platform, the transfer user withdraws the current call, and the remaining one
establishes a new call with the third party.
For example: 8001 and 8002 in the call state, 8002 transfer the current call to the user number
8003. At this time 8001 and 8003 establish a new call, and then 8002 withdraw the current call.
Enter SIP server address
51.108.26.31: 5060
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