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<div><span class="gmail_quote">On 10/19/05, <b class="gmail_sendername">Yair Hakak</b> <<a href="mailto:yhakak@gmail.com">yhakak@gmail.com</a>> wrote:</span>
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<div>i do it this way because i want all the dialplan logic and CDR having to do with PSTN in asterisk, not SER. </div>
<div>so, calls from the outside are adressed to <a onclick="return top.js.OpenExtLink(window,event,this)" href="mailto:sipaddress@myserver:5070" target="_blank">sipaddress@myserver:5070</a> and hit asterisk. asterisk either sends them along to 5060, or handles them internally (IVR, voicemail, etc) based on the dialplan.
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<div>clients on the inside are registered to the SER at 5060 and the SER automatically forwards them to asterisk. if they are PSTN asterisk serves as PSTN gateway, if they are internal, asterisk native bridges and drops out, but still keeps the CDR (i have full SIP addresses in my dial statements instead of asterisk SIP peers)
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<div>the reason i do this is i found that if the endpoints are scattered on the internet, SER+rtpproxy is much more stable than asterisk as a SIP server (asterisk kept dropping endpoints). This way SER serves as a completely "dumb" SIP server, and just sends everything along. there is a minimal increase in overhead (i could handle internal calls just with SER) but it's worth it to have all the dialplan logic and CDR's in one place.
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<div>also, obviously, if i use an IAX provider for outgoing, asterisk has to be in the middle.</div>
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<div>i agree though, it makes more sense to have SER on 5060 and asterisk somewhere else.</div>
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<div>hope i'm making some sense, please point out if i'm doing something really stupid.</div>
<div><br>-yair<br> </div>
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<div><span class="e" id="q_107083faa51f8711_1"><span class="gmail_quote">On 10/19/05, <b class="gmail_sendername">trixter aka Bret McDanel</b> <<a onclick="return top.js.OpenExtLink(window,event,this)" href="mailto:trixter@0xdecafbad.com" target="_blank">
trixter@0xdecafbad.com</a>> wrote:</span> </span></div>
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<div><span class="e" id="q_107083faa51f8711_3">On Wed, 2005-10-19 at 10:55 +0200, Yair Hakak wrote:<br>> hello,<br>> trace the SIP packets and see if they are actually addressed to 5062. <br>> if you post the ngrep or ethereal dump we'll see whats actually going
<br>> on. I do this with SER on 5060 and asterisk on 5070 and there are no<br>> problems - my extensions point to 5060 and my DID's point to 5070 so <br>> asterisk serves as the gateway to the PSTN.<br>><br>> -yair
<br>><br>><br>also look for dns packets and see if htey are pulling the server info.<br>Some sip clients look for specific server type dns records to see where <br>they should go.<br><br>5060 is the default, wouldnt it make more sense to have the default port
<br>be what you want the devices to goto and have that proxy to the device<br>you dont want direct connectivity to? Or am I missing something in that <br><br>><br>--<br>Trixter <a onclick="return top.js.OpenExtLink(window,event,this)" href="http://www.0xdecafbad.com/" target="_blank">
http://www.0xdecafbad.com</a> Bret McDanel<br>UK +44 870 340 4605 Germany +49 801 777 555 3402<br>US +1 360 207 0479 or +1 516 687 5200<br>FreeWorldDialup: 635378 <br><br><br></span></div>-----BEGIN PGP SIGNATURE-----
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