[Asterisk-Users] calling to sip provider

[Carles Pina i Estany]

Hello,

I am new user of Asterisk. Yesterday I was trying to call from softphone
to Asterisk, and that Asterisk routes this call to sipphone.com provider.

I have found information on internet about how to register to sipphone
and it seems that I have done. "sip show status" (or similar
command) in CLI was showing me that I was registered.

To call was not working, and on Asterisk's logs appeared:

------
  == Parsing '/etc/asterisk/enum.conf': Found
Asterisk Ready.
    -- Registered SIP '200' at 192.168.1.121 port 5060 expires 900
    -- Saved useragent "Linphone-1.2.0/eXosip" for peer 200
    -- Got SIP response 481 "Subcription Does Not Exist" back from 192.168.1.121
    -- Executing SetCallerID("SIP/200-0e5a", ""Name" 17476304480") in new stack
    -- Executing Dial("SIP/200-0e5a", "SIP/1747blabla at proxy01.sipphone.com|20|r") in new stack
    -- Called 1747blabla at proxy01.sipphone.com
    -- Got SIP response 500 "I'm terribly sorry, server error occured (1/SL)" back from 198.65.166.131
    -- SIP/proxy01.sipphone.com-8a47 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
  == Auto fallthrough, channel 'SIP/200-0e5a' status is 'CONGESTION'
------

Calling using linphone or other softphones was working, so it is not
"circuit-busy" error.

I tried lot of configurations (in sip.conf and extensions.conf). Call
is getting the correct route, but connection it is not working.

Asterisk is behind NAT, without any redirected port. I was using
externip and nat directives in configuration file. I think that I shouldn't
need redirected ports because I was trying to call, not to receive calls.

And NAT problem should be that I can listen but not talk (or vice-versa...)

Any idea about what I can check? Any suggestion?

Tahnk you very much,

-- 
Carles Pina i Estany		GPG id: 0x8CBDAE64
	http://pinux.info	Manresa - Barcelona