Solved

Asterisk Incoming calls failing

Posted on 2009-04-03
5
1,005 Views
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})
0
Comment
Question by:sallwine
  • 4
5 Comments
 
LVL 25

Expert Comment

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

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

should be ...

exten => _X.,1,Dial(SIP/2000)
0
 
LVL 25

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 ?
0
 
LVL 25

Accepted Solution

by:
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.
0
 
LVL 3

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.
0
 
LVL 25

Expert Comment

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

-x
0

Featured Post

Is Your Active Directory as Secure as You Think?

More than 75% of all records are compromised because of the loss or theft of a privileged credential. Experts have been exploring Active Directory infrastructure to identify key threats and establish best practices for keeping data safe. Attend this month’s webinar to learn more.

Question has a verified solution.

If you are experiencing a similar issue, please ask a related question

The Zaptel people (www.zaptel.com) got kind of annoyed with the fact that they were getting bombarded with searches for the zaptel driver system for Asterisk (not to mention they own the trademark on zaptel). So, they kindly requested that Digium ch…
Article by: user_n
How Sip Phone (User Agent) works and communicates with sip servers 1.  There is a sip server and a sip registrar.  The sip server and sip registrar can be one server or two different servers. The sip registrar is the server on which it is record…
Windows 10 is mostly good. However the one thing that annoys me is how many clicks you have to do to dial a VPN connection. You have to go to settings from the start menu, (2 clicks), Network and Internet (1 click), Click VPN (another click) then fi…
Video by: Mark
This lesson goes over how to construct ordered and unordered lists and how to create hyperlinks.

867 members asked questions and received personalized solutions in the past 7 days.

Join the community of 500,000 technology professionals and ask your questions.

Join & Ask a Question

Need Help in Real-Time?

Connect with top rated Experts

20 Experts available now in Live!

Get 1:1 Help Now