<div dir="ltr"><div><div><div>Hi Rutu,<br><br></div>Just a shot in a dark. :)<br><br></div>Did you added sip peer record in sip.conf, something similar to:<br><br><div class="" title="Hint: double-click to select code"><div class=""><code class="">[freeswitch]</code></div><div class=""><code class="">type=peer</code></div><div class=""><code class="">host=IP_ADDRESS_OF_FREESWITCH_SERVER</code></div><div class=""><code class="">username=<a href="http://HOSTNAME.DOMAIN.COM">HOSTNAME.DOMAIN.COM</a></code></div><div class=""><code class="">port=5080</code></div><div class=""><code class="">fromdomain=IP_ADDRESS_OF_FREESWITCH_SERVER</code></div><div class=""><code class="">secret=BOOTH_WAY_PASSWORD</code></div></div><br></div>With kind regards,<br><div><br></div></div><div class="gmail_extra"><br clear="all"><div><div class="gmail_signature"><div dir="ltr">Jurijs<br></div></div></div>
<br><div class="gmail_quote">On Thu, Mar 31, 2016 at 5:19 PM, Rutu Patel <span dir="ltr"><<a href="mailto:rutu.patel@inextrix.com" target="_blank">rutu.patel@inextrix.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div class="gmail_default" style="color:rgb(0,0,0)">Hi Michael<br><br></div><div class="gmail_default" style="color:rgb(0,0,0)">Thank you for your assistance.<br><br></div><div class="gmail_default" style="color:rgb(0,0,0)">We are trying to make call between two freeswitch users.<br><br>One user is registered in softphone and other is registered in Asterisk by register string but asterisk user cannot authenticate properly and freeswitch directly send 480 Temporary Unavailable.<br><br></div><div class="gmail_default" style="color:rgb(0,0,0)">In sip profile we set challenge-realm= auto_to also tried with auto_from but still not getting success.<br></div><div class="gmail_default" style="color:rgb(0,0,0)"><br></div><div class="gmail_default" style="color:rgb(0,0,0)">I am attaching some more logs if anyone can guide in right direction.<br></div><div class="gmail_extra"><br clear="all"><div><div><div dir="ltr"><div><div dir="ltr"><div><div dir="ltr"><div><div dir="ltr"><div><div dir="ltr"><div><div dir="ltr"><div><div dir="ltr"><div><div dir="ltr"><div><div dir="ltr"><div><div dir="ltr"><span style="background-color:rgb(255,255,255)"><span style="color:rgb(11,83,148)"><span style="font-family:verdana,sans-serif"><font size="1"><span style="color:rgb(68,68,68)">--<br>Thanks,<br>Rutu Patel<br></span><span style="font-weight:bold"><span><br></span><br></span></font></span></span></span></div></div></div></div></div></div></div></div></div></div></div></div></div></div></div></div></div></div></div></div></div>
<br><div class="gmail_quote">On Mon, Mar 28, 2016 at 10:23 PM, Michael Collins <span dir="ltr"><<a href="mailto:msc@freeswitch.org" target="_blank">msc@freeswitch.org</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">Try a gateway. Look in conf/sip_profiles/external/example.xml for a heavily-commented sample. For more detailed discussion see this nice doc:<div><a href="https://freeswitch.org/confluence/display/FREESWITCH/Gateways+Configuration" target="_blank">https://freeswitch.org/confluence/display/FREESWITCH/Gateways+Configuration</a><br></div><div><br></div><div>-MSC</div></div><div><div><div class="gmail_extra"><br><div class="gmail_quote">On Fri, Mar 25, 2016 at 11:34 PM, Rutu Patel <span dir="ltr"><<a href="mailto:rutu.patel@inextrix.com" target="_blank">rutu.patel@inextrix.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div style="color:rgb(0,0,0)">Hi,<br><br></div><div style="color:rgb(0,0,0)">Thank you for reply.<br><br></div><div style="color:rgb(0,0,0)">We have registered extension of Freeswitch in Asterisk as below<br><br>register=><a href="http://test-ext:secret@192.168.1.64/test-ext" target="_blank">test-ext:secret@192.168.1.64/test-ext</a></div><div class="gmail_extra"><br><div style="color:rgb(0,0,0)">We want to send call from Freeswitch to Asterisk.<br><br>What authentication parameters we need to set in Freeswitch ?<br></div><br clear="all"><div><div><div dir="ltr"><div><div dir="ltr"><div><div dir="ltr"><div><div dir="ltr"><div><div dir="ltr"><div><div dir="ltr"><div><div dir="ltr"><div><div dir="ltr"><div><div dir="ltr"><div><div dir="ltr"><span style="background-color:rgb(255,255,255)"><span style="color:rgb(11,83,148)"><span style="font-family:verdana,sans-serif"><font size="1"><span style="color:rgb(68,68,68)">--<br>Thanks,<br>Rutu Patel</span></font></span></span></span><br></div></div></div></div></div></div></div></div></div></div></div></div></div></div></div></div></div></div></div></div></div><div class="gmail_quote">On Wed, Mar 23, 2016 at 8:46 PM, Anthony Minessale <span dir="ltr"><<a href="mailto:anthony.minessale@gmail.com" target="_blank">anthony.minessale@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div dir="ltr">You need to supply auth credentials to Asterisk because its set to authenticate calls.<div>The best thing to do would be to make a gateway pointing to Asterisk with all the credentials and call it using that.</div><div> </div><div><br></div><div><br></div></div><div class="gmail_extra"><div><div><br><div class="gmail_quote">On Wed, Mar 23, 2016 at 10:20 AM, Jurijs Ivolga <span dir="ltr"><<a href="mailto:jurijs.ivolga@gmail.com" target="_blank">jurijs.ivolga@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div dir="ltr"><div><div><div>Hi Rutu,<br><br></div>In logs I can see that user-agent Asterisk is sending INVITE to Freeswitch and Freeswitch replies with "480 Temporarily Unavailable". This a bit differs from what you explained in your mail...<br><br></div><div>First think what pop up in my mind is that you need to configure Asterisk as a trunk on Freeswitch, not sure if this helps, but this is my first guess...<br><br><a href="https://freeswitch.org/confluence/display/FREESWITCH/Asterisk" target="_blank">https://freeswitch.org/confluence/display/FREESWITCH/Asterisk</a><br></div><br></div><div>With kind regards,<br><br></div><div>Jurijs<br></div><div class="gmail_extra"><br><div class="gmail_quote"><div><div>On Wed, Mar 23, 2016 at 3:46 PM, Rutu Patel <span dir="ltr"><<a href="mailto:rutu.patel@inextrix.com" target="_blank">rutu.patel@inextrix.com</a>></span> wrote:<br></div></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div><div><div dir="ltr"><div style="color:rgb(0,0,0)">Hi,<br><br>We are sending call to asterisk PBX but we are getting 480 Temporarily Unavailable immediately. from freeswitch server.<br> <br>Call flow is as below:<br><br>User => opensips => FS => PBX,
<br><br>but call not sending out from FS and drop by FS.<br><br></div><div style="color:rgb(0,0,0)">Here I am attaching both sip logs and freeswitch logs.<br><br></div><div style="color:rgb(0,0,0)">Can anyone please help to sort out the issue.<br clear="all"></div><div><div><div dir="ltr"><div><div dir="ltr"><div><div dir="ltr"><div><div dir="ltr"><div><div dir="ltr"><div><div dir="ltr"><div><div dir="ltr"><div><div dir="ltr"><div><div dir="ltr"><div><div dir="ltr"><span style="background-color:rgb(255,255,255)"><span style="color:rgb(11,83,148)"><span style="font-family:verdana,sans-serif"><font size="1"><span style="color:rgb(68,68,68)">--<br>Thanks,<br>Rutu Patel<br></span><span style="font-weight:bold"><span><span style="color:rgb(0,0,0)"><a href="http://www.inextrix.com" target="_blank"></a></span></span></span></font></span></span></span></div></div></div></div></div></div></div></div></div></div></div></div></div></div></div></div></div></div></div></div></div>
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<a href="http://www.freeswitch.org" rel="noreferrer" target="_blank">http://www.freeswitch.org</a><br></blockquote></div><br><br clear="all"><div><br></div></div></div><span><font color="#888888">-- <br><div><div dir="ltr"><div><div dir="ltr"><div><div dir="ltr">Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬<div><br><div>☞ <a href="http://freeswitch.org/" target="_blank">http://freeswitch.org/</a> ☞ <a href="http://cluecon.com/" target="_blank">http://cluecon.com/</a> ☞ <a href="http://twitter.com/FreeSWITCH" target="_blank">http://twitter.com/FreeSWITCH</a></div><div><div>☞ <a href="http://irc.freenode.net" target="_blank">irc.freenode.net</a> #freeswitch ☞ <u><a href="http://freeswitch.org/g+" target="_blank">http://freeswitch.org/g+</a></u><br><br></div><div>ClueCon Weekly Development Call <br></div><div>☎ <a href="mailto:sip%3A888@conference.freeswitch.org" target="_blank">sip:888@conference.freeswitch.org</a> ☎ <a href="tel:%2B19193869900" value="+19193869900" target="_blank">+19193869900</a> </div><div><br></div></div></div><div><a href="https://www.youtube.com/watch?v=9XXgW34t40s" style="color:rgb(17,85,204);font-size:12.8px" target="_blank">https://www.youtube.com/watch?v=9XXgW34t40s</a></div><div><a href="https://www.youtube.com/watch?v=NLaDpGQuZDA" target="_blank">https://www.youtube.com/watch?v=NLaDpGQuZDA</a><br></div></div></div></div></div></div></div>
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