<div class="gmail_quote"><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><div><div class="h5"><div class="gmail_quote">Dear all,<br> I want to setup the incoming calls, that don't use authentication in sip.conf file.<br>
My configurations as follows,<br><br>[2000]<br>
type=peer<br>
host=dynamic<br>
insecure=port,invite ; (both)<br>
context=Testing<br><br>But when I call '2000', I noticed the following message in Asterisk console,<br> <br>"NOTICE[5702]: chan_sip.c:10543 handle_request_invite: Failed to authenticate user "Velusamy" <<a href="mailto:sip%3A727@192.168.1.222" target="_blank">sip:727@192.168.1.222</a>>;tag=yj66acQcycvrN"<br>
</div></div></div></blockquote><div><br>Hi<br><br>I'm not sure about this but I think that it may cause by a bad setting in your softphone or your VOIP phone.<br>"Velusamy" is a terminal that you have configured in sip.conf ?<br>
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