Trouble with incoming calls

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Trouble with incoming calls

Postby aerys » Fri Feb 02, 2007 4:06 am

Hello,

We have a problem with the incoming calls by SIP, and never we receives the call, in the asterisk console we can read this:

Feb 2 10:45:47] WARNING[2562]: chan_sip.c:8023 check_auth: username mismatch, have <trunk_2>, digest has <s>
[Feb 2 10:45:47] NOTICE[2562]: chan_sip.c:13200 handle_request_invite: Failed to authenticate user "Antonio" <sip:709@192.168.0.207>;tag=as08bd2bd2

I know there is a bug, and my file incoming.html is the like this:

http://svn.digium.com/view/asterisk-gui ... 277&r2=278

But we still have the problem, Do you know if I have to do something more? At this moment I have the AsteriskNOW Beta 4, and I do "sudo conary update asterisk-gui" and there isn't more updates.

I probe the Asterisk GUI ot the svn also, and its the same.

Thanks.
aerys
Newsterisk
 
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Joined: Mon Jan 29, 2007 7:45 am

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