[Asterisk-Dev] HOWTO register the zap channel on the SIP-phone web server without the proxy (or registrar) server?

[elvisda]

Dear all,

Recently I try to setup the configuration files for the signalling and communication
between the SIP-phone and PSTN phone. I use no proxy server or registrar server.
I configure the sip.conf, extensions.conf, zapata.conf, and zaptel.conf files.
Because I have no proxy or registrar server, I try to add the zap channel (PSTN part)
to the web server which is embedded in the SIP-phone. (I can modify the phonebook
in the web site which is such as: http://IP address/.) It seems to sound like a little
strange that I register the zap channel on the SIP-phone web server using the SIP-phone's
IP address. Why should I do so? My reason is that the SIP-phone needs to have a certain
IP address for dialing under the condition of having no proxy-kind server.
I use the wildcard FXO port to connect the PSTN line. I have searched and studied the Asterisk
handbook several times but having no solution for this problem.
If I try to register the zap channel with the Asterisk server, I will suffer the bad situation
of failing to authenticate on REGISTER to <sip: pstn phone number at asterisk server's IP address>, for example.
If anyone ever knows about what I did, please give some suggestions.
Is there any solution for it in the open source code of Asterisk?
Or, should I really need another server?

Thanks in advance,

elvisda
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