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scottdube

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[20/Aug/2012 21:43:17] Asterisk: Authentication failed from host 66.x.x.x (uri:<sip:19782xxxxxx<_at_>66.x.x.x:5060>). Attempts: 12.

Calls do not ring they quickly disconnect, this entry appears in the security log. I can't find anything that points to where to go to resolve this issue. Is it a sip trunk issue or an operator issue. Using nexvortex as the provider, yse the proxy and username/password they provided is correct.

Thanks
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Vladimir Toncar (Kerio)

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Hi,

If you have more than 10 attempts, then the remote address has already blocked itself out. Maybe it's your SIP provider, maybe it's just some random attacker. Check the security tab in Advanced Options.

You may want to check the KB article at http://kb.kerio.com/article/how-to-configure-kerio-operator- to-connect-to-nexvortex-871.html

Vladimir

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scottdube

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The ip was not blocked by security. I used that article to set it up in the first place. I am using operator behind Kerio Control, RTP is not a service in the services choices in traffic rules by the way. Again I can make outbound calls but cannot get inbound calls. It does not even ring.
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scottdube

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Ok so it was the control box. I had the operator traffic coming in on an additional ip configured on the control box nic.

The config was
xx.xx.xx.225 control nic
xx.xx.xx.229 was the ip to be routed to the operator box

The sip trunking company nexvortex reported


Your PBX Server "xx.xx.xx.225" is responding back to NexVortex's "INVITE" with a "404 - Not Found". This usual means that the DIDs are not configured on your PBX.

Notice the last octet it should be 229 not 225.

So I moved the sip trunk to .225 and got it to work, ell at least the pbx answered the phone. But it will not respond to keypad inputs and the following is in the warning log of operator:

21/Aug/2012 15:07:20] asterisk[1531]: WARNING[1656]: chan_sip.c:3680 in retrans_pkt: Hanging up call CXC-119-7f7ac010-c9d5ed0-13c4-5033dc48-8f167625-64359c6<_at_ >xx.xx.xx.10 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissio ns).
[21/Aug/2012 15:07:20] {ami} Internal Asterisk Connection: Got empty channel

I have since moved it out from behind control to see if that was the issue but it acts the same way.
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scottdube

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UPDATE

Got a different sip account set up with a different company, works fine right out of the box! Company A. was Kerio certified partner. So far have been no help! Called support emailed support all they can ask is if the operator box has rfc 2833 enabled. Company B was much easier to set up and works right away... I thought the kerio partner would be the easiest route that's why I selected them as the first test company... I was not correct.
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Vladimir Toncar (Kerio)

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Hi,

I will have NexVortex re-tested and will also look into their certification claims. If you opened a ticket with our support, could you send me the ticket number in a private message?

Vladimir
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scottdube

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I reloaded the entire operator on the box... That seemed to solve the problem. Still not sure what the issue was.
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