[Asterisk-Dev] Dial the SIP phone to the PSTN phone with FXO interface

[elvisda]

Did anybody ever know how to configure the Asterisk server with FXO interface
so that we can dial the sip phone to PSTN phone?

The attached file is my configuration for sip.conf.
Would you please give me some suggestion?

I cannot complete the registration of the PSTN part.
It will show the following messages:

*CLI> sip show registry
Host                  Username     Refresh State
140.123.107.115:5060  +886-5-272       160 Request Sent
140.123.107.116:5060  elvisda          160 Registered

*CLI> Mar  5 02:39:00 NOTICE[49159]: chan_sip.c:3150 sip_reg_timeout: Registration for '+886-5-2720411 at 140.123.107.115' timed out, trying again
Mar  5 02:39:00 NOTICE[49159]: chan_sip.c:5585 handle_request: Registration from '<sip:+886-5-2720411 at 140.123.107.115>' failed for '140.123.107.115'
Mar  5 02:39:00 NOTICE[49159]: chan_sip.c:5010 handle_response: Failed to authenticate on REGISTER to '<sip:+886-5-2720411 at 140.123.107.115>;tag=as3f1480d8'


My country code is 886 so that I set up it as +886 begining instead of the American outgoing call with the +1 begining.
I guess I have to modify the Asterisk source code in chan_sip.c.
I will very appreciate if you give me some hint.
On the other hand, my Asterisk server IP address is 140.123.107.115.

Thanks, anyway...

Best Regards,

elvisda
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