<div dir="ltr"><div><div>Hello Matthew,<br><br></div>My version is Asterisk 1.6.2.9.<br><br><span id="result_box" class="" lang="en"><span class="">Or</span> <span class="">have you seen</span> <span class="">NAT</span><span>?</span> <span class="">I have no</span> <span class="">NAT on</span> <span class="">my</span> <span class="">network</span><span>.</span> <span class="">Have you seen</span> <span class="">my little</span> <span class="">diagram</span> <span class="">above</span><span>?</span><br>
</span><br><span id="result_box" class="" lang="en"><span class=""><span id="result_box" class="" lang="en"><span class="">Here it is:<br><br></span></span>SIP friends (phones) <-----> Asterisk <-----> SIP gateway to PSTN converter<br>
80.236.215.61 109.69.217.6 internal IP ( <a href="http://10.4.0.10/255.255.255.0">10.4.0.10/255.255.255.0</a> )<br></span></span><br><span id="result_box" class="" lang="en"><span class="">My</span> <span class="">Asterisk</span> <span class="">server has</span> <span class="">two</span> <span class="">NIC/interfaces.</span><br>
<br class=""><span class="">- 1</span> <span class="">interface</span> <span class="">with</span> <span class="">public</span> <span class="">IP</span> (109.69.217.6 </span><span id="result_box" class="" lang="en"><span id="result_box" class="" lang="en"><span class="">to talk with SIP friends</span></span>)<br class="">
<span class="">- 1</span> <span class="">interface</span> <span class="">with</span> <span class="">internal</span> <span class="">ip (</span></span>10.4.0.1 to talk with SIP gateway's)<br><br><span id="result_box" class="" lang="en"><span class="">SIP</span> <span class="">friend</span> <span class="">should not even</span> <span class="">know that</span> <span class="">the</span> <span class="">call is routed to</span> <span class="">the</span> <span class="">SIP</span>/PSTN <span class="">gateway</span><span class="">.</span><br>
<span class="">It could be</span> <span class="">a</span> <span class="">SIP</span> <span class="">trunk</span> <span class="">to</span> <span class="">a</span> <span class="">SIP</span> <span class="">provider</span> <span class="">Internet</span><span class="">,</span> <span class="">the user does not</span> <span class="">have</span> <span class="">to know.</span></span>..<br>
<br></div>Best regards,<br>Mickael<br><br></div><div class="gmail_extra"><br><br><div class="gmail_quote">2013/6/13 Matthew J. Roth <span dir="ltr"><<a href="mailto:mroth@imminc.com" target="_blank">mroth@imminc.com</a>></span><br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Mickael MONSIEUR wrote:<br>
><br>
> I have a standard Asterisk configuration:<br>
><br>
> SIP friends (phones) <-----> Asterisk <-----> SIP gateway to PSTN converter<br>
> <a href="tel:80.236.215.61" value="+18023621561">80.236.215.61</a> 109.69.217.6 internal IP ( <a href="http://10.4.0.10/255.255.255.0" target="_blank">10.4.0.10/255.255.255.0</a> )<br>
><br>
> When analyzing traffic on a SIP friend/phone I see this:<br>
><br>
> INVITE sip:xxxx@<a href="tel:80.236.215.61" value="+18023621561">80.236.215.61</a>:64946;ob SIP/2.0<br>
> Via: SIP/2.0/UDP 109.69.217.6:5060;branch=z9hG4bK52d50250;rport<br>
> Max-Forwards: 70<br>
> From: < <a href="mailto:sip%3Axxxx@109.69.217.6">sip:xxxx@109.69.217.6</a> >;tag=as15b47581<br>
> To: "test" < <a href="mailto:sip%3Axxxx@109.69.217.6">sip:xxxx@109.69.217.6</a> >;tag=kp1VwHD80rA9MVdBjTF4jyFIaCkrJcjh<br>
> Contact: < <a href="mailto:sip%3Axxxxx@109.69.217.6">sip:xxxxx@109.69.217.6</a> ><br>
> Call-ID: MSMhw2bsheHWAQgHlae3O7yKQ2P9EcsM<br>
> CSeq: 102 INVITE<br>
> User-Agent: Asterisk<br>
> Require: timer<br>
> Session-Expires: 1800;refresher=uas<br>
> Min-SE: 90<br>
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO<br>
> Supported: replaces, timer<br>
> Content-Type: application/sdp<br>
> Content-Length: 217<br>
><br>
> v=0<br>
> o=root 664087974 664087976 IN IP4 10.4.0.10<br>
> s=Asterisk<br>
> c=IN IP4 10.4.0.10<br>
> t=0 0<br>
> m=audio 8652 RTP/AVP 8 101<br>
> a=rtpmap:8 PCMA/8000<br>
> a=rtpmap:101 telephone-event/8000<br>
> a=fmtp:101 0-16<br>
> a=ptime:20<br>
> a=sendrecv<br>
><br>
> My equipement IP 10.4.0.10 is visible to the user, why?<br>
<br>
<br>
Mickael,<br>
<br>
What version of Asterisk are you running?<br>
<br>
Is the Asterisk server outside and the SIP gateway to PSTN converter inside of a<br>
NAT?<br>
<br>
What are the NAT SUPPORT and MEDIA HANDLING settings in sip.conf?<br>
<br>
Regards,<br>
<br>
Matthew Roth<br>
InterMedia Marketing Solutions<br>
Software Engineer and Systems Developer<br>
<br>
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</blockquote></div><br></div>