<ol style="color:rgb(51,51,51);font-family:monospace;font-size:11px;background-color:rgb(255,255,255)"><li class="li1"><div class="de1" style="font-family:'Courier New',Courier,monospace">recv <span class="nu0">369</span> bytes from udp/<span class="br0">[</span><span class="nu0">202.126</span><span class="nu0">.41</span><span class="nu0">.108</span><span class="br0">]</span>:<span class="nu0">5060</span> at <span class="nu0">22</span>:<span class="nu0">13</span>:<span class="nu0">09.398177</span>:</div>
</li><li class="li2" style="background-color:rgb(244,244,244);background-repeat:initial initial"><div class="de2" style="font-family:'Courier New',Courier,monospace"> ------------------------------------------------------------------------</div>
</li><li class="li1"><div class="de1" style="font-family:'Courier New',Courier,monospace"> SIP/<span class="nu0">2.0</span> <span class="nu0">604</span> Does not exist anywhere</div></li></ol><div><br></div><div>
The NO_ROUTE_DESTINATION clearing cause is due to the 604 sent to you from your SIP provider. It is unrelated to your profile.</div><div><br></div><div>Only your SIP provider can tell you the reason for the 604 - raise it with them. It could be a number of things - an invalid number, unreachable number, internal/upstream error etc.</div>
<div><br></div><div>-Steve</div><div><br></div><div><br></div><br><div class="gmail_quote">On 12 May 2013 23:37, Steven Ayre <span dir="ltr"><<a href="mailto:steveayre@gmail.com" target="_blank">steveayre@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Not sure why you replaced the gateway name with xxxx - it makes it impossible to see which gateway you used without looking through the log. It isn't hiding anything of value.<div>
<br></div><div>As I said on my previous email, but in more detail:</div>
<div><br></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">EXECUTE sofia/internal/<a href="mailto:407@10.15.1.41" target="_blank">407@10.15.1.41</a> bridge(sofia/gateway/globe/0018444485452)</blockquote>
<div><br></div><div>You are dialing out via gateway 'globe'</div><div><br></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">
external::globe gateway <a href="mailto:sip%3A029080340@siphosted.com" target="_blank">sip:029080340@siphosted.com</a> REGED</blockquote><div><br></div><div>The 'globe' gateway is configured on the 'external' profile.</div>
<div><br></div><div>Any call sent OUT via the globe gateway will leave from the profile it is configured on.</div><div><br></div><div>If your call comes IN via internal, it will go OUT via external since the gateway is on that profile. FS lets you dial between profiles (and even different protocols) so what you came in on does not affect what you leave on.</div>
<div><br></div><div>This would be the expected and indeed required behaviour, since the gateway configured on external uses its IP and the <a href="http://siphosted.com" target="_blank">siphosted.com</a> provider may not be reachable from the IP used on other profiles.</div>
<div><br></div><div>-Steve</div><div class="HOEnZb"><div class="h5"><div><br><div><br></div><div><br><br><div class="gmail_quote">On 11 May 2013 06:43, baskar <span dir="ltr"><<a href="mailto:yudha2008@gmail.com" target="_blank">yudha2008@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Hi Steve,<br>
<br>
Thanks for the reply.<br>
<br>
When i try to use only the gateway it pass through external profile why it<br>
is happning.<br>
<br>
SIP trunk has username and password with hard cable connected from service<br>
provider.<br>
<br>
SIP trunk configured in freeswitch server and pass through service provider<br>
through SIP trunk.<br>
<div><br>
sofia/internal/<a href="mailto:407@10.15.1.41" target="_blank">407@10.15.1.41</a> bridge(sofia/gateway/xxxx/0018444485452)<br>
2013-05-08 06:13:09.376846 [NOTICE] switch_channel.c:669 New Channel<br>
sofia/external/0018444485452 [4cdd9992-b763-11e2-bdef-2d40d364119f]<br>
<br>
</div>Can any one guide me procedure to calls passing through the SIP server.<br>
<div><br>
Thanks in advance.<br>
<br>
Regards,<br>
N.Baskar<br>
<br>
<br>
<br>
--<br>
</div>View this message in context: <a href="http://freeswitch-users.2379917.n2.nabble.com/NO-ROUTE-DESTINATION-tp7590466p7590575.html" target="_blank">http://freeswitch-users.2379917.n2.nabble.com/NO-ROUTE-DESTINATION-tp7590466p7590575.html</a><br>
<div><div>Sent from the freeswitch-users mailing list archive at Nabble.com.<br>
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