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8.5

Asterisk Incoming calls failing

Asked by sallwine in Voice Over IP, Asterisk Open Source Telephony

Tags: Asterisk

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})
[+][-]04/03/09 09:38 PM, ID: 24065927Accepted Solution

Your question has an Asker Certified™ answer! sallwine verified that this solution worked for them--which means it will likely work for you, too. Click to view the solution free for 30-days now.

About this solution

Zones: Voice Over IP, Asterisk Open Source Telephony
Tags: Asterisk
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Solution Provided By: xuserx2000
Participating Experts: 1
Solution Grade: A
 
[+][-]04/03/09 09:21 PM, ID: 24065900Expert Comment

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[+][-]04/03/09 09:30 PM, ID: 24065915Expert Comment

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[+][-]04/05/09 07:46 AM, ID: 24071637Author Comment

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[+][-]04/05/09 09:35 AM, ID: 24071975Expert Comment

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