Thanks for the reply. Hereš<div><br></div><div>sofia status profile externalš<a href="http://pastebin.freeswitch.org/18754">http://pastebin.freeswitch.org/18754</a></div><div><br></div><div>Hm... You are right... WTF with codecs @ SDP.</div>
<div><br></div><div>originate {absolute_codec_string=&#39;PCMU,PCMA,G729&#39;}sofia/gateway/<a href="http://mtt.ru/89093848124">mtt.ru/89093848124</a> &amp;playback()</div><div><br></div><div>and INVITE i have</div><div><br>
</div><div><div>U 2012/03/27 11:12:45.244216 <a href="http://62.76.180.83:5080">62.76.180.83:5080</a> -&gt; <a href="http://80.75.130.134:5060">80.75.130.134:5060</a></div><div>INVITE <a href="mailto:sip%3A89093848124@sip.mtt.ru">sip:89093848124@sip.mtt.ru</a> SIP/2.0.</div>
<div>Via: SIP/2.0/UDP 62.76.180.83:5080;rport;branch=z9hG4bKy0HX29D6rc86K.</div><div>Max-Forwards: 70.</div><div>From: &quot;&quot; &lt;<a href="mailto:sip%3A100002000696@sip.mtt.ru">sip:100002000696@sip.mtt.ru</a>;transport=udp&gt;;tag=7USXQ5jg6caFB.</div>
<div>To: &lt;<a href="mailto:sip%3A89093848124@sip.mtt.ru">sip:89093848124@sip.mtt.ru</a>&gt;.</div><div>Call-ID: 9ea5c38c-f2a0-122f-fd98-00163e000f5d.</div><div>CSeq: 26087318 INVITE.</div><div>Contact: &lt;sip:100002000696@62.76.180.83:5080;transport=udp;gw=<a href="http://mtt.ru">mtt.ru</a>&gt;.</div>
<div>User-Agent: IP-PBX.</div><div>Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY.</div><div>Supported: timer, precondition, path, replaces.</div><div>Allow-Events: talk, hold, refer.</div>
<div>Content-Type: application/sdp.</div><div>Content-Disposition: session.</div><div>Content-Length: 201.</div><div>X-FS-Support: update_display,send_info.</div><div>Remote-Party-ID: &lt;<a href="mailto:sip%3A0000000000@sip.mtt.ru">sip:0000000000@sip.mtt.ru</a>&gt;;party=calling;screen=yes;privacy=off.</div>
<div>.</div><div>v=0.</div><div>o=FreeSWITCH 1332822661 1332822662 IN IP4 62.76.180.83.</div><div>s=FreeSWITCH.</div><div>c=IN IP4 62.76.180.83.</div><div>t=0 0.</div><div>m=audio 24104 RTP/AVP 0 101 13.</div><div>a=rtpmap:101 telephone-event/8000.</div>
<div>a=fmtp:101 0-16.</div><div>a=ptime:20.</div></div><div><br></div><div><br></div><div>The same... I&#39;m little bit frustrating :)</div><div><br><div class="gmail_quote">27 ÍÁÒÔÁ 2012šÇ. 14:26 ÐÏÌØÚÏ×ÁÔÅÌØ jay binks <span dir="ltr">&lt;<a href="mailto:jaybinks@gmail.com">jaybinks@gmail.com</a>&gt;</span> ÎÁÐÉÓÁÌ:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">your not getting very far at all..<div><br></div><div>try enabling more codecs... there is something in your invite ( or SDP ) that they dont like.</div>
<div>but their logs tell you nothing more than the pcap you provided.</div>

<div><br></div><div>I can assure you MERA DOES work... I send millions of calls to MERA servers every day from my freeswitch boxes.</div><div>so this is something specific to your setup.</div><div><br></div><div>Id also suggest the ITSP should be able to give you more info and be a little more helpful.</div>


<div><br></div><div>Jay<br><br><div class="gmail_quote"><div><div class="h5">On 27 March 2012 20:07, Anton Kvashenkin <span dir="ltr">&lt;<a href="mailto:anton.jugatsu@gmail.com" target="_blank">anton.jugatsu@gmail.com</a>&gt;</span> wrote:<br>
</div></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div class="h5">

Hello guys. I have setup the gateway for outbound calls withšcredentials provided by my ITSP.<div><br></div><div><a href="http://pastebin.freeswitch.org/18749" target="_blank">http://pastebin.freeswitch.org/18749</a>
</div><div><br></div><div>Trying to make test call using, for example, originate command</div><div><br></div><div>originate sofia/gateway/<a href="http://mtt.ru/XXXX" target="_blank">mtt.ru/XXXX</a> &amp;txfax(/tmp/fax.tif)</div>


<div><br>
</div><div>result:</div><div><br></div><div>ITSP proxy sends 500 Gateway is Invalidš<a href="http://pastebin.freeswitch.org/18750" target="_blank">http://pastebin.freeswitch.org/18750</a></div><div><br></div><div>But, trying to make call using asterisk with the samešcredentials suprisenly works. Here dump of success callš<a href="http://pastebin.freeswitch.org/18751" target="_blank">http://pastebin.freeswitch.org/18751</a></div>



<div><br></div><div>I also attach logs that provided mys ITSP. Don&#39;t mind that there is ip address from privite LAN, the first try was from behind NAT. The second, that i provided above (<a href="http://pastebin.freeswitch.org/18750" target="_blank">http://pastebin.freeswitch.org/18750</a>), was from vps with staticly assigned address. For the first glance I thought that the root of this problem was NAT but IMO it&#39;s not.</div>



<div><br></div><div>Also, my external.xmlš</div><div><br></div><div><a href="http://pastebin.freeswitch.org/18752" target="_blank">http://pastebin.freeswitch.org/18752</a>
</div>
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<br></blockquote></div><span class="HOEnZb"><font color="#888888"><br><br clear="all"><div><br></div>-- <br>Sincerely<br><br>Jay<br>
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