sallwine
asked on
Asterisk Incoming calls failing
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=192.168.0.0/255.2
register => <userid>:<pwd>@<host>/<use
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
exten => 2000,n,Dial(SIP/2000,30)
exten => 2000,n,Hangup()
exten => 2001,1,Verbose(1,Extension
exten => 2001,n,Dial(SIP/2001,30)
exten => 2001,n,Hangup()
exten => 2002,1,Verbose(1,Extension
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/vo
[local_outgoing_calls]
exten => _NXXNXXXXXX,1,Dial(SIP/voi
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=192.168.0.0/255.2
register => <userid>:<pwd>@<host>/<use
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
exten => 2000,n,Dial(SIP/2000,30)
exten => 2000,n,Hangup()
exten => 2001,1,Verbose(1,Extension
exten => 2001,n,Dial(SIP/2001,30)
exten => 2001,n,Hangup()
exten => 2002,1,Verbose(1,Extension
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/vo
[local_outgoing_calls]
exten => _NXXNXXXXXX,1,Dial(SIP/voi
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 ?
What distro are you running ?...do you have SELinux enabled ?..Ip tables setup ?
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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.
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.
glad I could help.
-x
-x
exten => _X.,1,Dial(SIP,2000)
should be ...
exten => _X.,1,Dial(SIP/2000)