What in FreePBX is generating these headers?
Audio is at 65.60.7.4 port 11622
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 206.123.79.110:5060:
INVITE sip:19567898087@206.123.79.110 SIP/2.0
Via: SIP/2.0/UDP 65.60.7.4:5060;branch=z9hG4bK77e2f121;rport
From: "tcreek" <sip:Unknown@65.60.7.4>;tag=as610c326a
To: <sip:19567898087@206.123.79.110>
Contact: <sip:Unknown@65.60.7.4>
Call-ID: 7d8b94896ce26bc4325682745086bbf4@aerialtelcom.com
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 20 May 2009 04:55:01 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 234I know it is actually produced by Chap_sip, but the FROM line, it is generating "Unknown" which is causing me to get authentication problems from my provider.
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Could it possibly be that
Could it possibly be that you don't have your trunk setup correctly?
All of the Asterisk peer configuration variables are available to you through FreePBX web interface. They are documented here.
The fromuser variable would probably be a good place to start.
Not posting your trunk setup gives us no basis to assist you.
what's the call flow? Is it
what's the call flow? Is it a forwarded/followme-ed call? If so, it may be from how the call came in. You may want to try adding "sendrpid=yes" and see what the header looks like there. (Not that your provider necessarily will use that but it may expose more info).
Philippe Lindheimer - FreePBX Project Leader
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You should do something to
You should do something to protect your server, on a hunch on browsed 65.60.7.4 and it let me right in to FreePBX with no password.
If it is not already hacked, it will be.
Robert Keller - VoIPologist
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Thanks for all the great
Thanks for all the great replies.
I did have a reply, but I edited it all out,
I followed SkyKingOH link to the "fromuser" advice and that is what made it work now.
Thanks so much all for the advice.
Hopefully my box is also secure.
Geekyboy, editing is good.
Geekyboy, editing is good. And I realize I wasn't responding to your question. Just another warning about security. Your box is still hung out the Internet for no good reason. At least close port 80 and learn to ssh tunnel.
Robert Keller - VoIPologist
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cosmicwombat: Yes I know you
cosmicwombat: Yes I know you were not responding to the question. I know all about SSH, but if I close port 80, then the benefit of having >FreePBX is gone. I have have just wall just stuck with manually editing asterisk,. But yes I know what you mean, I could always open it back up when I need it. However I plan to use it for other things that I NEED access to it. I will just have to install more security, such as "fail2ban" etc.
So what do you mean, "still hung out there?" I tried it again?, unless I have answered you.
Thanks.
Well Sir, if you know all
Well Sir, if you know all about ssh then I would guess you are all set. I won't attempt to educate you further on the subject.
Robert Keller - VoIPologist
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Laughing, you can lead a
Laughing, you can lead a horse to water but you can't make them drink.
Robert is telling you to close port 80 to the Internet not to shut the web server down.
Have you ever heard of a firewall or NAT?
I suggest you read about SSH tunneling which is what is being referred to and take a look at the Hamachi VPN package which can be installed via yum (I think it might me an RP/M).
I know what he is
I know what he is saying.
Hamachi VPN is a closed source software, but I will give it a look, though it does seem to have high requirements for CPU usage.
Thanks for the hints and tips