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@proxy01.sipphone.com|20|r") in new stack
-- Called 1747blabla@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
