<div dir="ltr"><div class="gmail_default" style="color:rgb(0,0,0)">Hi,<br><br></div><div class="gmail_default" style="color:rgb(0,0,0)">Thank you for your help to debug the issue.<br></div><div class="gmail_default" style="color:rgb(0,0,0)"><br>We are now trying without opensips but still facing authentication issue.<br></div><div class="gmail_default" style="color:rgb(0,0,0)"><br></div><div class="gmail_default" style="color:rgb(0,0,0)">We created two users of freeswitch in asterisk as peer.<br><br>For example, First peer in sip.conf is 456789 and second is 123456.<br><br>[123456]<br>type=friend<br>host=x.x.x.45<br>username=123456<br>secret=abcdef<br>disallow=all<br>allow=g729<br>trustrpid=yes<br>canreinvite=yes<br>sendrpid=yes<br>context=extensions-test<br>qualify=yes<br>directmedia=yes <br><br>Now when we are trying to reach 123456, the request reach to asterisk but not authenticate proper user(123456). It authenticate first peer(456789) defined in sip.conf.<br><br>If we define 123456 before 456789 then it works but we have multiple peers in asterisk so how can we manage that ?<br><br>We also tried by sending user credential direct in bridge,but still it is authenticate the first peer defined in sip.conf .<br><br>{sip_auth_username=123456,sip_auth_password=abcdef,sip_contact_user=123456}sofia/default/NUMBER@IP<br> <br></div><div class="gmail_extra"><div class="gmail_default" style="color:rgb(0,0,0)">Can anyone please explain how to authenticate specific peer ?<br><br></div><div class="gmail_default" style="color:rgb(0,0,0)">Please let me know if you want any other details.</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<div class="gmail_default" style="color:rgb(0,0,0);display:inline"></div><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>
<br><div class="gmail_quote">On Fri, Apr 1, 2016 at 3:56 PM, 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><div>Hi,<br><br></div>I'm not sure about your environment, but on this point I will recommend you to isolate issue. So on this point you should understand if your sip trunk between Asterisk and Freeswitch configured correctly, without involving Opensips and etc. To test SIP trunk you can try to send call directly from freeswitch to Asterisk and see if it works, you can send call just directly from freeswitch console, something similar to:<br><pre>originate sofia/example/<a href="mailto:300@foo.com" target="_blank">300@foo.com</a> 8600</pre><a href="https://freeswitch.org/confluence/display/FREESWITCH/mod_commands" target="_blank">https://freeswitch.org/confluence/display/FREESWITCH/mod_commands</a><br><br></div>If you call will reach Asterisk without any issue, then this means that sip trunk created correctly and issue is somewhere on Opensips side or any other part of your set-up.<br><br></div><div>Second option will be to try to configure sip trunk without password and check if it helps<br></div><br><a href="https://freeswitch.org/confluence/display/FREESWITCH/Asterisk" target="_blank">https://freeswitch.org/confluence/display/FREESWITCH/Asterisk</a><br><br></div>With kind regards,<br></div><div class="gmail_extra"><br clear="all"><div><div><div dir="ltr">Jurijs<br></div></div></div><div><div>
<br><div class="gmail_quote">On Fri, Apr 1, 2016 at 12:54 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:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">Hi Juris, <div><br></div><div>Yes we added peer in asterisk sip.conf.</div><div><br></div><div>Are we still missing any configurations? <div><div><br><br>On Thursday, March 31, 2016, Jurijs Ivolga <<a href="mailto:jurijs.ivolga@gmail.com" target="_blank">jurijs.ivolga@gmail.com</a>> 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>Just a shot in a dark. :)<br><br></div>Did you added sip peer record in sip.conf, something similar to:<br><br><div title="Hint: double-click to select code"><div><code>[freeswitch]</code></div><div><code>type=peer</code></div><div><code>host=IP_ADDRESS_OF_FREESWITCH_SERVER</code></div><div><code>username=<a href="http://HOSTNAME.DOMAIN.COM" target="_blank">HOSTNAME.DOMAIN.COM</a></code></div><div><code>port=5080</code></div><div><code>fromdomain=IP_ADDRESS_OF_FREESWITCH_SERVER</code></div><div><code>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><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>rutu.patel@inextrix.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 style="color:rgb(0,0,0)">Hi Michael<br><br></div><div style="color:rgb(0,0,0)">Thank you for your assistance.<br><br></div><div 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 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 style="color:rgb(0,0,0)"><br></div><div 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>msc@freeswitch.org</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">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>rutu.patel@inextrix.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 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>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>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>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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</blockquote></div></div></div><span><font color="#888888"><br><br>-- <br><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>iNextrix Technologies Pvt. Ltd.</span><span style="font-weight:bold"><span><br></span><span><span style="color:rgb(0,0,0)"><a href="http://www.inextrix.com" target="_blank">www.inextrix.com</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><br>
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<a href="mailto:FreeSWITCH-users@lists.freeswitch.org" target="_blank">FreeSWITCH-users@lists.freeswitch.org</a><br>
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<br>_________________________________________________________________________<br>
Professional FreeSWITCH Consulting Services:<br>
<a href="mailto:consulting@freeswitch.org" target="_blank">consulting@freeswitch.org</a><br>
<a href="http://www.freeswitchsolutions.com" rel="noreferrer" target="_blank">http://www.freeswitchsolutions.com</a><br>
<br>
Official FreeSWITCH Sites<br>
<a href="http://www.freeswitch.org" rel="noreferrer" target="_blank">http://www.freeswitch.org</a><br>
<a href="http://confluence.freeswitch.org" rel="noreferrer" target="_blank">http://confluence.freeswitch.org</a><br>
<a href="http://www.cluecon.com" rel="noreferrer" target="_blank">http://www.cluecon.com</a><br>
<br>
FreeSWITCH-users mailing list<br>
<a href="mailto:FreeSWITCH-users@lists.freeswitch.org" target="_blank">FreeSWITCH-users@lists.freeswitch.org</a><br>
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