Incomming always busy

Asterisk Asterisk Support
1

Hi

Im a VERY newbie on Asterisk. Have a problem with my newly setup asterisk server.
Dialing between extensions work fine, dialing out via my sip provider works fine. Dialing from public phone net to my SIP number always get’s a busy tone.

I have made a sip debug and can see that the call hits my server.

Can you gurus take a look at my sip.conf and extension.conf (the only two that i have edited) and point me in the correct direction?

----- SIP.CONF -----
[general]
bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
disallow=all
allow=ulaw
allow=alaw
context=default
callerid=Unknown
;useragent=Asterisk Cellip
deny=0.0.0.0/0.0.0.0
permit=192.168.200.0/255.255.252.0

register => USERNAME:SECRET@sip.provider.com/1000

[mysecretary]
type=peer
username=USERNAME
fromuser=USERNAME
secret=SECRET
host=sip.provider.com
fromdomain=sip.provider.com
;context=cellip-in
context=from-trunk
insecure=port,invite
canreinvite=no

[gert]
type=friend
context=phones
host=dynamic

[nisse]
type=friend
context=phones
host=dynamic



----- EXTENSIONS.CONF -----
[general]
static=yes
writeprotect=no
autofallthrough=yes
include => from-trunk

[globals]

[default]
exten => s,1,Verbose(1|Unrouted call handler)
exten => s,2,Answer()
exten => s,3,Wait(1)
exten => s,4,Playback(tt-weasels)
exten => s,5,Hangup()

exten => _.,1,Dial(SIP/${EXTEN}@mysecretary)
exten => _.,2,Hangup()

[from-trunk]
exten => 1000,1,Dial(SIP/gert)
exten => 1000,2,Hangup()

[internal]
exten => s,1,Verbose(1|Echo test application)
exten => s,n,Echo()
exten => s,n,Hangup()

exten => 2000,1,Verbose(1|Extension 2000)
exten => 2000,n,Dial(SIP/gert,30)
exten => 2000,n,Hangup()

exten => 2002,1,Verbose(1|Extension 2002)
exten => 2002,n,Dial(SIP/nisse,30)
exten => 2002,n,Hangup()

[phones]
include => internal
include => default
2

Better post a log of incoming call. (Not SIP, just verbose log first)

3

:smile:

Leads to another question, were do I find that?

Tried “logger set level VERBOSE on” and then dialed. No output in the console though.

4

“core set verbose 3” in console. Look, what happens, if dial from one endpoint to another. If nothing, you get the wrong console :smile:

5

No output what so ever :frowning:

6

Ok, got it. Restarted asterisk.

Calling my SIP number from my cell, gives busy signal. Below happened.

asteriskCLI> core set verbose 3
Verbosity was 0 and is now 3
== Using SIP RTP CoS mark 5
– Executing [1000@from-trunk:1] Dial(“SIP/mysecretary-00000001”, “SIP/gert”) in new stack
== Using SIP RTP CoS mark 5
[May 4 12:41:57] WARNING[5595]: app_dial.c:1747 dial_exec_full: Unable to create channel of type ‘SIP’ (cause 20 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
– Executing [1000@from-trunk:2] Hangup(“SIP/mysecretary-00000001”, “”) in new stack
== Spawn extension (from-trunk, 1000, 2) exited non-zero on 'SIP/mysecretary-00000001’
asteriskCLI> core set verbose 0

7

Has gert ever registered?

8

Everything is solved.

I havent changed anything, the “service asterisk restart” seems to have solved the issue.
I can now call from outside and have the calls rerouted to both gert and nisse. Works as a charm.

Thanks for letting me waste your time on a no-issue. Next time i will restart first.