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Looks like the last commit to 4.4 is about 2.5 years old :/ I did a quick diff of the php from 4.4 to 4.5 and saw this commit: 66786e17911635f1eb647c184241a7775af2a0ff That MAY be it, but there's 2 years worth of commits and the fix could be any of those. You could also try maybe just...

Try setting "media_bug_answer_req" to true at the top of your dialplan. https://freeswitch.org/confluence/plugins/servlet/mobile?contentId=16353855#content/view/16353855

Recently, I am running into an issue with some severe audio delay with some of my deployments in AWS. It gets to the point where audio is delayed up to 10 seconds in both directions. The calls start off fine and slowly get worse. The only way I've been able to fix this is by disabling the rtp...

The IP address advertised in the Via header is usually configured on the socket. Based on the Opensips docs it should look something like this: "listen = udp:<private_ip>:5060 as <public_ip>:5060" I'm not 100 percent sure on that syntax, in Kamailio it's: "listen = udp:<private_ip>:5060...

Oh I assumed 45.147.* was the fs server b/c of the user-agent header. But yeah in this case it's pretty clear that sip stack isn't getting the response from the invite which is causing all those retransmission. Based on the 3g thing it's most likely network related. You could confirm this by...

The DB schema is pretty simple, most fields are just text fields. I once wrote a similar feature that created the inbound_route when a customers port order entered FOC status in a separate number porting app. If you have a web app you can just setup a DB connection to the fusion database and...

Alex's guide was super helpful when I was setting up Kamailio as a registrar in front of several FS boxes. You can find it here: http://www.evaristesys.com/blog/server-side-nat-traversal-with-kamailio-the-definitive-guide/ Not sure if Opensips has an equivalent to the Kamailio nathelper module...

I am also using RDS without issues for several years in a fairly large deployment with multiple freeswitch servers. I'd definitely recommend this. As far as filesystem goes I went a different route, I moved everything off the filesystem and into s3(recodings, voicemail files, etc). I use...

It looks like freeswitch isn't getting the sip responses. Those packets with the extra ">>>" are retransmission and are sent when requests don't receive the reply the expect. Where are you running the packet capture?

Been in IT for about 5 years now, started doing network/server support for a small MSP. Now I help manage several stand alone PBX's (fusionPBX and freepbx mostly). Looking to contribute to the community and learn as much as I can.

You said put a router between the main network and the client? I'm assuming this is a nat router? I suspect you need to enable options ping(check under extension config). This periodically sends a sip options packet to the client and "tricks" the firewall into not closing the port.