<div dir="ltr">They don't require authentication of invites which I do need <div><br></div><div><br clear="all"><div dir="ltr">Regards,<br>Zohair Raza<div><br></div><div><br></div></div>
<br><br><div class="gmail_quote">On Thu, Mar 22, 2012 at 3:20 PM, Leandro Dardini <span dir="ltr"><<a href="mailto:ldardini@gmail.com">ldardini@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div class="HOEnZb"><div class="h5"><div class="gmail_quote">2012/3/22 Zohair Raza <span dir="ltr"><<a href="mailto:engineerzuhairraza@gmail.com" target="_blank">engineerzuhairraza@gmail.com</a>></span><br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div dir="ltr">Hi, <div><br></div><div>How to allow registered sip users to call without re-authentication</div><div><br></div><div>insecure =yes/very are deprecated in 1.8 </div><div><br></div><div>I want to avoid fromuser= in peer configuration. When I add this in peer asterisk, my asterisk accepts call otherwise it says username mismatch.</div>
<div><br></div><div>Please help</div><div><br></div><div><br clear="all"><div dir="ltr">Regards,<br>Zohair Raza<div><br></div></div>
</div></div>
</blockquote></div><br></div></div><div>There are other options, like "invite" and "port" to be used when you trust the IP of the caller. </div><span class="HOEnZb"><font color="#888888"><div><br></div>
<div>Leandro</div>
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