SIP is circuit-busy

Asterisk Asterisk SIP
1

Hey,

everytime i try to call for anyone, i received this message

 == Using SIP RTP CoS mark 5
    -- Executing [554832781336@1774:1] NoOp("SIP/3301-0000000a", "") in new stack
    -- Executing [554832781336@1774:2] Dial("SIP/3301-0000000a", "SIP/1774") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/1774
[Jul 10 10:42:44] NOTICE[4865][C-00000005]: chan_sip.c:23237 handle_response_invite: Failed to authenticate on INVITE to '"3301" <sip:1774@187.103.111.110>;tag=as03b1ca7d'
    -- SIP/1774-0000000b is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing [554832781336@1774:3] Hangup("SIP/3301-0000000a", "") in new stack
  == Spawn extension (1774, 554832781336, 3) exited non-zero on 'SIP/3301-0000000a'

my sip.conf is

[general]
context=default
allowguest=no
allowoverlap=no
udpbindaddr=0.0.0.0
tcpenable=no
tcpbindaddr=0.0.0.0
qualify=yes
transport=udp
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
disallow=all
allow=ulaw,alaw,g729
canreinvite=no
localnet=192.168.1.0/255.255.255.0
externip=201.76.8.155
register => 
register => 
[1774]

context=1774
type=friend
host=187.103.111.110
fromuser=
secret=
transport=udp
fromdomain=187.103.111.110
port=5060
canreinvite=no
qualify=yes
nat=yes
insecure=port,invite
trustrpid=yes
sendrpid=yes
directmedia=no
keepalive=45
dtmfmode=frc2833

and my extensions.conf is

[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
priorityjumping=no

[1774]
;exten => _X.,1,Dial(SIP/3301${EXTEN}@1774)
exten => _X.,1,NoOp()
exten => _X.,n,Dial(SIP/1774)
exten => _X.,n,Hangup()

pleaaaseeeeeeeee help me

2

A call was sent to the remote side and authentication failed. You would need to confirm username/password is correct and also confirm with the remote side.

3

The log doesn’t show an authentication failure. Unfortunately it doesn’t show the actual original failure, either. The verbosity needs increasing.

4

While the SIP messages aren’t present there is this:

[Jul 10 10:42:44] NOTICE[4865][C-00000005]: chan_sip.c:23237 handle_response_invite: Failed to authenticate on INVITE to '"3301" <sip:1774@187.103.111.110>;tag=as03b1ca7d'

So it was auth related.

5

Oops. I didn’t scroll to see it :-(.

6

the password and the username is correct, cause i manage the remote side too.

7

Have you looked at the remote side to see if it has any detail? If it’s Asterisk have you looked at the output and confirmed that it is matching the appropriate entry in sip.conf or pjsip.conf?

8

On the remote side I only have access to the graphical platform, I can not see in detail if it gave error or not. The guy who works with me said that on the remote side the call comes out, he also said that it’s my Asterisk blocking the call.

9

Could you send me an example of sip.conf and extensions.conf that works?

10

I don’t have an example handy but there should be numerous on the web.

11

You should have the sample config included with the source of your asterisk install.

Look in the configs/samples and configs/basic-pbx directories.