是这样设置的吗？

Required Settings
SIP Server: voip.freephoneline.ca
Alternative SIP Server: voip2.freephoneline.ca
Transport: UDP
Port: 5060
Notes:
It is always best to use the DNS name for your SIP server as our infrastructure is always expanding/changing/being maintained.
The IP addresses which you register to will change over time.
Use of Fongo SIP Servers that are not listed in this document will result in your account being suspended.
Registration Interval: 3600 seconds (1 hour)
Registration Expiry: 3600 seconds (1 hour)
Failed Registration Re-Try Interval: 120 seconds
Recommended Settings
STUN/ICE: Disable
NAT Mapping Enabled:Yes
NAT Traversal:
Enable sending Keep-Alives only:
on Grandstream HT-701 ATAs this setting is “no, but send keep-alive”
Keep Alive Message:
NOTIFY or a UDP PING Packet
For Linksys/Cisco devices, use ‘Nat Keep Alive Msg’ = $NOTIFY or $PING
Never use REGISTER as your Keep Alive message
Keep Alive Interval: 20 seconds*
*Audio may be affected if this value is adjusted
Notes:
The above settings can be used to configure your SIP client to function in common home network configurations. Since there a
thousands of home network configurations, it is impossible for us to provide a single set of parameters that will always work. As a
VoIP Key purchaser, it’s expected that you have knowledge of your network and how to configure your SIP client properly.
Freephoneline does not offer STUN server. However, you may use a public one if your wish.
RTP Settings
Supported Codecs: g711-u (uLAW) and g729
Suggested RTP Packet size
(psize):
0.020 - This ensures audio packets every 20 milliseconds, achieving better quality (trade-off:
bandwidth)