solution, you can choose this option.
Call Forwarding There are four options for you to choose. Disable is to close
call forwarding function. Always means all the incoming calls
will be forwarded into SIP URL without any reason. Busy
means the incoming calls will be forwarded into SIP URL only
when the local system is busy. No Answer means if the
incoming calls do not receive any response, they will be
forwarded to the SIP URL by the time out.
SIP URL – Type in the SIP URL (e.g., aaa@draytel.org or
abc@iptel.org) as the site for call forwarded.
Time Out – Set the time out for the call forwarding. The
default setting is 30 sec.
Ring Port Set Phone 1 and/or Phone 2 as the default ring port(s) for
this SIP account.
Ring Pattern Choose a ring tone type for the VoIP phone call.
Prefer Codec Select one of five codecs as the default for your VoIP calls.
The codec used for each call will be negotiated with the peer
party before each session, and so may not be your default
choice. The default codec is G.729A/B; it occupies little
bandwidth while maintaining good voice quality.
If your upstream speed is only 64Kbps, do not use G.711
codec. It is better for you to have at least 256Kbps upstream
if you would like to use G.711.
Single Codec – If the box is checked, only the selected Codec
will be applied.
Packet Size The amount of data contained in a single packet. The default
value is 20 ms, which means the data packet will contain 20
ms voice information.
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