Hello! I’m quite new to SIP, but while I may be a noob to it I don’t think I’m THAT much of a noob as far as basic networks goes. Here’s the fun thing I’m going through right now:
If I call from my test extension 1000 (conveniently named “God”) to my smartphone at 1003, the call drops are 32 seconds. Apparently I get a repeated flood of 401s in the debug. If I call from 1003 to 1000, the call goes on forever. The codec negotiation and quality are great, and I love how Linphone gives me opus and speex so readily.
Some notes: the smartphone is an LG V20 Android smartphone, and the test extension right now is on a MacBook Pro running High Sierra; both are running Linphone and are updated. I tried calling an iOS device last night and it worked just beautifully, no problem. The server is an Ubuntu VirtualBox system with an adapter in bridging mode.
1000 to 1003, Fail
1003 to 1000, Success
NAT isn’t being used in either case at this point, and yet this is happening. I’m so lost x.x. To make this even more interesting, 1003 can’t call 1005, it also fails in a similar manner. 1005 is a One Plus One, so it’s an Android smartphone. The same problem occurs when 1005 calls 1003, it still drops at 32 seconds.
I was initially only referencing to my Asterisk box via its private IP address, but then I figured “what the heck, might as well try to do it all in one go.” The problem is still the same =. Zoiper coincidentally doesn’t have as many issues but I don’t want to pay 8 dollars to use a codec that’s otherwise free per phone.
I’m using a Sophos UTM 9 firewall, I’ve enabled SIP mode, I’ve added the DNAT rule, I’ve enabled all SIP relevant communications to go through. Some firewall specific rules:
1000 is on a laptop with full access and all ports opened and on the same WAN as the Asterisk server
1003 is on a smartphone with all ports opened but on a different WAN (AND THIS ONE WORKS!)
1005 is a standard smartphone with ports closed on the same WAN as 1003