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Asterisk Incoming calls failing

Posted on 2009-04-03
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Last Modified: 2013-11-12
I have an Asterisk server that is connected via SIP trunk to my provider. Outbound calls work great, but inbound calls give an immediate fast busy.

Peers look registered and fine.

I do not even see messages about the call in the Asterisk console.
If I do a sniffer trace then I do see a 401 unauthorized packet that looks like it from from the incoming call.

originally I had the specific DID in the extension.conf. I changed it to _X. to try and make sure that it was not blocking it, and to have everything route to ext 2000; however, if I understand correctly, then if it was a problem with the extensions.conf file, then wouldn't I get a message at the Asterisk console of ext <DID> could not be found (or something like that).

Here are the important sections of the sip.conf

[general]
localnet=192.168.0.0/255.255.255.0
register => <userid>:<pwd>@<host>/<userid>
canreinvite=no

[2000]
type=friend
context=phones
host=dynamic
secret=<pwd>
qualify=3000

[2001]
type=friend
context=phones
host=dynamic
secret=<pwd>
qualify=3000

[2002]
type=friend
context=phones
host=dynamic
secret=<pwd>
qualify=3000

[voipvoip-outgoing]
type=peer
username=<userid>
secret=<pwd>
nat=auto
insecure=very
host=<host>
fromuser=<userid>
dtmfmode=rfc2833
disallow=all
allow=g729
allow=ilbc
allow=ulaw
allow=alaw
qualify=yes

[voipvoip-incoming]
username=<userid, not DID>
type=friend
secret=<pwd>
nat=no
insecure=very
host=<host>
dtmfmode=rfc2833
disallow=all
allow=g729
allow=ilbc
allow=ulaw
allow=alaw
context=from-trunk
qualify=yes


Here are the important sections of the extensions.conf
[internal]
exten => 2000,1,Verbose(1,Extension 2000)
exten => 2000,n,Dial(SIP/2000,30)
exten => 2000,n,Hangup()

exten => 2001,1,Verbose(1,Extension 2001)
exten => 2001,n,Dial(SIP/2001,30)
exten => 2001,n,Hangup()

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

[phones]
include => internal
include => ld_outgoing_calls
include => local_outgoing_calls
include => demo

[from-trunk]
include => internal
exten => _X.,1,Dial(SIP,2000)

[ld_outgoing_calls]
exten => _1NXXNXXXXXX,1,Dial(SIP/voipvoip-outgoing/${EXTEN})

[local_outgoing_calls]
exten => _NXXNXXXXXX,1,Dial(SIP/voipvoip-outgoing/1${EXTEN})
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Question by:sallwine
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Expert Comment

by:Ron M
ID: 24065900
this is wrong...

exten => _X.,1,Dial(SIP,2000)

should be ...

exten => _X.,1,Dial(SIP/2000)
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Expert Comment

by:Ron M
ID: 24065915
There may still be another problem if you didn't see any errors on the console when dialing in...if it hits asterisk you should see something..  but the above should definitely be changed first.

What distro are you running ?...do you have SELinux enabled ?..Ip tables setup ?
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Accepted Solution

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Ron M earned 500 total points
ID: 24065927
You might also try setting NAT to auto on the incoming sip entry.

And i'm not 100 percent positive here...but I don't think you need to specify the incoming user and secret....for incoming calls from the sip trunk provider....unless the provider specifically told you too.  I think your server is challenging your provider for the user/secret, and they aren't sending one.
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Author Comment

by:sallwine
ID: 24071637
The extension was a misprint, I had changed that already, but that should have been a / instead of a , ... in case someone looks at this in the future.
Also after I removed the authentication then calls started to come in, but said no audio could be negoiated (or something like that), so I moved my codec commands up to the general section, and it worked great.
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Expert Comment

by:Ron M
ID: 24071975
glad I could help.

-x
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