<html><head><meta http-equiv="content-type" content="text/html; charset=utf-8"></head><body dir="auto"><div>Ain't it enough just the fact that they are rewriting the sip stack with pjsip? Lol. <br><br>Sent from my iPhone</div><div><br>On Apr 24, 2013, at 5:13 AM, Jeff Bernhardt <<a href="mailto:jeff@askcornerstone.net">jeff@askcornerstone.net</a>> wrote:<br><br></div><blockquote type="cite"><div>
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<p class="MsoNormal"><span style="font-size:10.0pt">Thanks for taking the time to answer. I know it gets busy around here with all sorts of stuff that frankly is over my head! It’s kind of nice that way, though… keeps some of the mystery and excitement alive
for what’s possible.<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:10.0pt"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:10.0pt">Yeah, I didn’t mean it like “Asterisk can do this so what the hell is wrong with Freeswitch?” Was just wondering why, so thanks for the clear explanation.
<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:10.0pt"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:10.0pt">I actually didn’t know Asterisk had so much goofiness. Can you (or anyone else) give any examples of its goofiness? We’re relatively light PBX users in general (just the basics for clients with no more than
150 phones, some with only 5 phones!), so we might not have come across any of them.
<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:10.0pt"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:10.0pt">Jeff Bernhardt<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:10.0pt">Systems Administrator<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:10.0pt">Cornerstone Consulting<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:10.0pt">808.440.2900<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:10.0pt"><o:p> </o:p></span></p>
<p class="MsoNormal"><b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif"">From:</span></b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif""> <a href="mailto:freeswitch-users-bounces@lists.freeswitch.org">freeswitch-users-bounces@lists.freeswitch.org</a> [<a href="mailto:freeswitch-users-bounces@lists.freeswitch.org">mailto:freeswitch-users-bounces@lists.freeswitch.org</a>]
<b>On Behalf Of </b>Michael Collins<br>
<b>Sent:</b> Tuesday, April 23, 2013 7:46 PM<br>
<b>To:</b> FreeSWITCH Users Help<br>
<b>Subject:</b> Re: [Freeswitch-users] External Softphone vs. Internal Question<o:p></o:p></span></p>
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<p class="MsoNormal" style="margin-bottom:12.0pt">Hi Jeff,<o:p></o:p></p>
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<p class="MsoNormal" style="margin-bottom:12.0pt">The short answer is that you are not forced to create a separate profile for internal vs. external phones. However, FreeSWITCH gives you this freedom whereas Asterisk does not. You *could* try to cram everything
into port 5060, but there's no compelling reason to do so. A lot of VoIPers are accustomed to using 5060 and only 5060, come what may. FreeSWITCHers generally view that as a limitation, not a feature.
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By having multiple SIP profiles - quite literally multiple SIP UAs - you have more freedom and flexibility to handle goofy scenarios like dealing with broken NAT devices. You can put all your broken stuff on a different profile and not have to worry that setting
a particular option to fix one device will break another device. <o:p></o:p></p>
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<p class="MsoNormal" style="margin-bottom:12.0pt">Oh, and keep in mind that "just because Asterisk can do it" doesn't mean that Asterisk does it correctly. There are a lot of devices out there that "work" but only because they all choose to be synchronized
in their goofiness. Reams have been written about how FS does not pander to broken devices so I won't belabor the point here. Just know this: FS is relatively strict in adhering to specs and standards, so if something works with Asterisk (or whatever VoIP
software) but not with FS then most likely it's a matter of figuring out how to tell FS to emulate the brokenness for the sake of interoperability.<o:p></o:p></p>
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<p class="MsoNormal">Hope this helps. Let us know how your setup is coming along. Be sure to use
<a href="http://pastebin.freeswitch.org">pastebin.freeswitch.org</a> to share any configurations or logs with us.<br>
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Thanks,<o:p></o:p></p>
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<p class="MsoNormal">-MC<o:p></o:p></p>
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<p class="MsoNormal" style="margin-bottom:12.0pt"><o:p> </o:p></p>
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<p class="MsoNormal">On Sat, Apr 20, 2013 at 2:50 AM, Jeff Bernhardt <<a href="mailto:jeff@askcornerstone.net" target="_blank">jeff@askcornerstone.net</a>> wrote:<o:p></o:p></p>
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<p class="MsoNormal"><span style="font-size:10.0pt">Hi. I have the following basic setup questions:
<o:p></o:p></span></p>
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<p class="MsoNormal"><span style="font-size:10.0pt"><o:p> </o:p></span></p>
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<p class="MsoNormal"><span style="font-size:10.0pt">When using a softphone (Bria on iPhone) from external (on a different external ip address), I could register but no audio would be passed either way for any calls. I saw that I should set ext-rtp-ip in the
internal sip profile to my external ip address (it was on auto-nat, which apparently wasn't working) in this wiki <a href="http://wiki.freeswitch.org/wiki/NAT_Traversal" target="_blank">http://wiki.freeswitch.org/wiki/NAT_Traversal</a><o:p></o:p></span></p>
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<p class="MsoNormal"><span style="font-size:10.0pt">That didn't work, so I also set my ext-sip-ip to my public ip. After that, I could pass audio.<o:p></o:p></span></p>
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<p class="MsoNormal"><span style="font-size:10.0pt"><o:p> </o:p></span></p>
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<p class="MsoNormal"><span style="font-size:10.0pt">However, if I register the phone internally instead and call for instance the IVR test line, the call drops after 30 seconds.<o:p></o:p></span></p>
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<p class="MsoNormal"><span style="font-size:10.0pt">So it's either no audio when registered externally or 30 second calls when registered internally.<o:p></o:p></span></p>
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<p class="MsoNormal"><span style="font-size:10.0pt"><o:p> </o:p></span></p>
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<p class="MsoNormal"><span style="font-size:10.0pt">I found this wiki: <a href="http://wiki.freeswitch.org/wiki/General_NAT_example_scenarios" target="_blank">http://wiki.freeswitch.org/wiki/General_NAT_example_scenarios</a><o:p></o:p></span></p>
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<p class="MsoNormal"><span style="font-size:10.0pt">I fall into either scenario 2 or 3, and for both, it says to create a dedicated profile for external registrations and put them on port 5090, which works. However, is there no other way to solve this problem
that doesn't require the use of an additional profile on port 5090 but also doesn't cut off internally registered calls after 30 seconds? On Asterisk, there's no need to open a second port to register external phones. What's different about Freeswitch?<o:p></o:p></span></p>
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<p class="MsoNormal"><span style="font-size:10.0pt"><o:p> </o:p></span></p>
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<p class="MsoNormal"><span style="font-size:10.0pt">Also, I don't know what role these play, but I also get these errors:<o:p></o:p></span></p>
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<p class="MsoNormal"><span style="font-size:10.0pt">[WARNING] switch_core_media.c:1282 Asynchronous PTIME not supported, changing our end from 0 to 20<o:p></o:p></span></p>
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<p class="MsoNormal"><span style="font-size:10.0pt">at seemingly random times<o:p></o:p></span></p>
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<p class="MsoNormal"><span style="font-size:10.0pt">...and....<o:p></o:p></span></p>
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<p class="MsoNormal"><span style="font-size:10.0pt">[INFO] switch_nat.c:590 NAT port mapping disabled<o:p></o:p></span></p>
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<p class="MsoNormal"><span style="font-size:10.0pt">when I make a call from internally or externally registered softphone to external number.<o:p></o:p></span></p>
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<p class="MsoNormal"><span style="font-size:10.0pt"><o:p> </o:p></span></p>
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<p class="MsoNormal"><span style="font-size:10.0pt">Thank you.<o:p></o:p></span></p>
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<br>
-- <br>
Michael S Collins<br>
Twitter: @mercutioviz<br>
<a href="http://www.FreeSWITCH.org" target="_blank">http://www.FreeSWITCH.org</a><br>
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</div></blockquote><blockquote type="cite"><div><span>_________________________________________________________________________</span><br><span>Professional FreeSWITCH Consulting Services:</span><br><span><a href="mailto:consulting@freeswitch.org">consulting@freeswitch.org</a></span><br><span><a href="http://www.freeswitchsolutions.com">http://www.freeswitchsolutions.com</a></span><br><span></span><br><span>FreeSWITCH-powered IP PBX: The CudaTel Communication Server</span><br><span><a href="http://www.cudatel.com">http://www.cudatel.com</a></span><br><span></span><br><span>Official FreeSWITCH Sites</span><br><span><a href="http://www.freeswitch.org">http://www.freeswitch.org</a></span><br><span><a href="http://wiki.freeswitch.org">http://wiki.freeswitch.org</a></span><br><span><a href="http://www.cluecon.com">http://www.cluecon.com</a></span><br><span></span><br><span>FreeSWITCH-users mailing list</span><br><span><a href="mailto:FreeSWITCH-users@lists.freeswitch.org">FreeSWITCH-users@lists.freeswitch.org</a></span><br><span><a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a></span><br><span>UNSUBSCRIBE:http://<a href="http://lists.freeswitch.org/mailman/options/freeswitch-users">lists.freeswitch.org/mailman/options/freeswitch-users</a></span><br><span><a href="http://www.freeswitch.org">http://www.freeswitch.org</a></span><br></div></blockquote></body></html>