<div dir="ltr"><div><div><div><div>Hi Jeff,<br><br></div>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. <br>
<br>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. <br>
<br></div>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.<br>
<br></div>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><br>Thanks,<br></div>
-MC<br></div><div class="gmail_extra"><br><br><div class="gmail_quote">On Sat, Apr 20, 2013 at 2:50 AM, Jeff Bernhardt <span dir="ltr"><<a href="mailto:jeff@askcornerstone.net" target="_blank">jeff@askcornerstone.net</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
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Hi. I have the following basic setup questions:
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<div>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" style="font-size:10pt" target="_blank">http://wiki.freeswitch.org/wiki/NAT_Traversal</a></div>
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<div>That didn't work, so I also set my ext-sip-ip to my public ip. After that, I could pass audio.</div>
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<div>However, if I register the phone internally instead and call for instance the IVR test line, the call drops after 30 seconds.</div>
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<div>So it's either no audio when registered externally or 30 second calls when registered internally.</div>
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<div>I found this wiki: <a href="http://wiki.freeswitch.org/wiki/General_NAT_example_scenarios" style="font-size:10pt" target="_blank">http://wiki.freeswitch.org/wiki/General_NAT_example_scenarios</a></div>
<div>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?</div>
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<div>Also, I don't know what role these play, but I also get these errors:</div>
<div>[WARNING] switch_core_media.c:1282 Asynchronous PTIME not supported, changing our end from 0 to 20</div>
<div>at seemingly random times</div>
<div><span style="font-size:10pt">...and....</span></div>
<div>[INFO] switch_nat.c:590 NAT port mapping disabled</div>
<div>when I make a call from internally or externally registered softphone to external number.</div>
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<div>Thank you.</div>
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