我在想，bubblestar和ckleea师兄的Asterisk都可以，估计是某些settings与我的Asterisk Server （GUI）有些不同，现在我用TS-119的Asterisk Server也出现同样的情况（打不出，503 out of service), 打入一接通就断线。这个问题已经发生于好几位members身上。
Att.: It is possible to use OPTION to try to solve NAT problem in order to keep open the connection from Asterisk to the peer behind NAT. I will write about SIP Pbx protected by Firewall/NAT in future posts.
If qualify is set to "yes" then, by the looks of it, Asterisk will use the information about round trip time to decide whether or not to bother registering. If the SIP OPTIONS packet doesn't receive a response, it assumes the server is unreachable and probably doesn't bother trying to register. Switching off "qualify" is obviously necessary with such providers.
It's also generally a good idea to have qualify=no for softphones and maybe some hardphones. the OPTIONS packets can cause problems with them.
Purpose of qualify=yes
On the other hand, one of the main benefits of qualify=yes is to detect network problems with peers.
We send a lot of calls via a service provider using SIP but we have qualify-yes set so that if it becomes unreachable the dial fails
immediatly without having to wait for a timeout which enables us to
seamlessly failover to an ISDN or other connection.
發表於 2012-10-1 00:36
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