1.8 supports static peers along with realtime peers. I have tested.<br><br><div class="gmail_quote">On Mon, Feb 21, 2011 at 11:04 PM, Ricardo Carvalho <span dir="ltr"><<a href="mailto:rjcarvalho.lists@gmail.com">rjcarvalho.lists@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><span style="border-collapse: collapse; font-family: arial,sans-serif; font-size: 13px;">Thanks Faisal, in fact I made a test that confirmed that in realtime asterisk doesn’t supported static peers, like you told me.<div>
Do you know if newer versions of asterisk, like 1.8, have this issue already solved?</div><div><br></div><div>Regards,</div><div>Ricardo.</div><font color="#888888"><div><br></div><div><br></div><div><br></div></font></span><div>
<div></div><div class="h5"><br><div class="gmail_quote">
On Wed, Feb 16, 2011 at 6:26 PM, Faisal Hanif <span dir="ltr"><<a href="mailto:faisal@vopium.com" target="_blank">faisal@vopium.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div link="blue" vlink="purple" lang="EN-US"><div><p class="MsoNormal"><span style="font-size: 10pt; color: rgb(31, 73, 125);">I have played a lot on this issue with asterisk config but in realtime it doesn’t supported static peers with version 1.6.2.14.</span></p>
<p class="MsoNormal"><span style="font-size: 10pt; color: rgb(31, 73, 125);"> </span></p><p class="MsoNormal"><b><span style="font-size: 10pt;">From:</span></b><span style="font-size: 10pt;"> Ricardo Carvalho [mailto:<a href="mailto:rjcarvalho.lists@gmail.com" target="_blank">rjcarvalho.lists@gmail.com</a>] <br>
<b>Sent:</b> Wednesday, February 16, 2011 10:21 PM</span></p><div><br><b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br></div><b>Cc:</b> Faisal Hanif<br><b>Subject:</b> Re: [asterisk-users] trunk not working if I register a phone at the same IP as the trunk peer's IP<p>
</p><div><div></div><div><p class="MsoNormal"> </p><p class="MsoNormal">Isn't this a limitation that can be surpassed with some configuration that I'm lacking in my sip.conf or extensions.conf of my asterisk?</p>
<div><p class="MsoNormal"> </p></div><div><p class="MsoNormal">Ricardo.</p></div><div><p class="MsoNormal"> </p></div><div><p class="MsoNormal"> </p></div><div><p class="MsoNormal"> </p></div><div><p class="MsoNormal" style="margin-bottom: 12pt;">
</p><div><p class="MsoNormal">On Wed, Feb 16, 2011 at 4:54 PM, Faisal Hanif <<a href="mailto:faisal@vopium.com" target="_blank">faisal@vopium.com</a>> wrote:</p><div><div><p class="MsoNormal"><span style="font-size: 10pt; color: rgb(31, 73, 125);">Well a quick n easy fix for you is you can configure you call sending peers to use username & secret in INVITE. As far as I know it possible in almost all CISCO, Avaya and all other standard Gateway and SBCs which follows full SIP RFCs.</span></p>
<p class="MsoNormal"><span style="font-size: 10pt; color: rgb(31, 73, 125);"> </span></p><p class="MsoNormal"><span style="font-size: 10pt; color: rgb(31, 73, 125);">If you can’t do it then you need to use curl as realtime engine instead of MySQL. It will call a URL for each SIP request which you can handle with flexibility in your CGI scripts with apache. But be careful as per my experience asterisk 1.6 with curl as realtime engine can handle a max of 120 registration in parallel if registration refresh time is 120 seconds.</span></p>
<p class="MsoNormal"><span style="font-size: 10pt; color: rgb(31, 73, 125);"> </span></p><p class="MsoNormal"><span style="font-size: 10pt; color: rgb(31, 73, 125);">Faisal Hanif</span></p><p class="MsoNormal"><span style="font-size: 10pt; color: rgb(31, 73, 125);"> </span></p>
<p class="MsoNormal"><b><span style="font-size: 10pt;">From:</span></b><span style="font-size: 10pt;"> <a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a> [mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a>] <b>On Behalf Of </b>Ricardo Carvalho<br>
<b>Sent:</b> Wednesday, February 16, 2011 9:41 PM<br><b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br><b>Subject:</b> [asterisk-users] trunk not working if I register a phone at the same IP as the trunk peer's IP</span></p>
<div><div><p class="MsoNormal"> </p><p class="MsoNormal">How should I configure my asterisk server so that I can receive calls from an unregistered peer from whom I also receive registrations of sip phones?</p><div><p class="MsoNormal">
</p></div><div><p class="MsoNormal">I'm asking you this, because with my actual configuration, when I register a contact from that peer's IP, no more inbound calls are accepted from that peer, as my asterisk rejects those INVITEs with "407 Proxy Authentication Required", I assume because they don't carry the registered contact registration!!!</p>
</div><div><p class="MsoNormal">My SIP contacts have type=friend and all inbound calls not coming from my registered phones fall in the default context without authentication, so that someone in the Internet be able to call freely through the Internet anyone in my server's dial plan.</p>
</div><div><p class="MsoNormal"> </p></div><div><p class="MsoNormal">Some ideas?</p></div><div><p class="MsoNormal"> </p></div><div><p class="MsoNormal">Regards,</p></div><div><p class="MsoNormal">Ricardo Carvalho.</p></div>
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