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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" target="_blank" style="font-size: 10pt;">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" target="_blank" style="font-size: 10pt;">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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