Create sip account, call and return 503 service unavailable

I’m use asterisk 11.2.1, freepbx, linphone.

use freepbx create sip account 1000, 1001.
1000 call 1001 is ok.
1001 call 1000 is ok.

manual create sip account 1003 in sip_additinoal.conf.
execute “sip reload” in asterisk cli.
sip show peers is ok.

1003 call 1000 is ok.
1000 call 1003 , sip response is 503 service unavailable.

req:INVITE
res:401
req:ACK
req:INVITE
res:100
res:183 session progress
res:503 service unavailable

why?

Unfortunately, this forum can’t upload attachment file.

1 Like

Please direct freepbx problems to freepbx.org. In any case, without the full SIP trace, it is not possible to guess, as 503 is a pretty generic failure reason. It is quite likely that you will need to look at logging from the remote side.

If you created a SIP entry manually (in sip_additional.conf), you probably need to create a dialplan entry manually (probably in extensions_additional.conf). I do not work with GUI’s so this is just a guess. But it would be the logical thing to try.