No Incoming / Outgoing SIP Call possible

Asterisk Asterisk Support
1

Hi Mates,

it took about one week to get asterisk 1.2 with Freepbx 2.2.1 running. Now i have a problem wich i cannot solve by myself. I always tried many various combinations of trunkdefinitions and inbound and outbound routes.

Incoming Sip Calls are showed on my console but not extension is ringing. here is the output of such a try:

Retransmitting #5 (no NAT) to 63.214.186.6:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
63.214.186.6;branch=z9hG4bKdf9f8a0d3f5af799019db65cbde33ab1;received=63.
214.186.6
Via: SIP/2.0/UDP 195.226.174.68;branch=z9hG4bK2fa84ca5
From: 01726435398 sip:01726435398@195.226.174.68;tag=as5922fa80
To: sip:bungee@63.214.186.6;tag=as09ce33af
Call-ID: 40fad99a6986c50753a3806a0810e1b8@195.226.174.68
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Proxy-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="1ac95f79"
Content-Length: 0

Is there any errormessage or can somebody see that anything is going wrong?

Please help me to get rid of the Problem.

The Machine is running on a Centos vServer from my provider without nat.

Kind regards,
Enno

2

You should be lookin fot this:

this incoming caller has not authenticated itself yet. Define is sip.conf, then see what the cli says when it tries to register.

3

Hi,

do you know if the message “SIP/2.0 407 Proxy Authentication Required” is really an error message? How can I fix that?

I would be very appreciated if somebody could check if my configuration is ok, I still cannot call in or out. Talks between the extensions are fine:

extension 7000
http://www.t190.com/extension_7000.pdf

inbound_routes
http://www.t190.com/inbound_routes.pdf

outbound_route
http://www.t190.com/outbound_route.pdf

trunk_nikotel
http://www.t190.com/trunk_nikotel.pdf

Thanks and greetings,
Enno