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I'm probably living on borrowed time, but my HiFormance Chicago server has been active throughout this discussion and continues to run quite robustly. I really like this server for its super-low RTTs to nearly everywhere I route VoIP traffic. (Keeping my fingers crossed, but running daily...

Nothing. Indeed, I'm using that approach now. (See my write-up at https://cboh.org/voip/obi/OBi_As_ITSP.html for instructions on how to do it.) There are two significant impediments to this approach that NAF's implementation doesn't have: First, there is a bug in the OBi firmware that...

I've been in contact with James offline and mentioned that I'd seen this myself, although I don't recall ever resolving it. Of interest is the fact that it showed up for the both of us while we were attempting to get GVSIP to work in an ARM environment. Perhaps it is a bug that just happens to...

Setting up SSL under Apache is really a function of Apache: Do you have mod_ssl loaded? Have you specified the correct path to your credentials? (For CentOS 6, this would be done in /etc/httpd/conf.d/ssl.conf.) BTW, for future reference, since it is likely only you that will be using these...

I'm no expert on the crypto stuff, nor am I familiar with Asterisk on the Raspberry Pi (which I believe you are running), but here's a couple things I'd look at in your situation: 1. Under CentOS, I see Ward sets "CFLAGS='-DENABLE_SRTP_AES_256 -DENABLE_SRTP_AES_GCM'" during the configuration...

Sorry, I'm not using Find Me/Follow Me

OK. With the help of James McMuffin, I think I have a handle on what's going on with the "no shared cipher" messages. For some reason NAF created his own TLS transport, in pjsip_custom.conf, which he calls transport_tls. This works for GVsip, but it is not complete. It is lacking at least a...

With respect to your issue 1, I believe the GV1XXXXXXXXXX (were there really only 9 Xes?) is a hold-over from Bill Simon's GVGW. If so, it can safely be deleted. "OBITALK Device" is what was created by the GVsip code. What does FreePBX > Settings > Voicemail Admin show? With respect to...

At some point TLS and SRTP started working here for me, although I don't know exactly what caused them to do so, but I have some suspicions. Let's try to narrow it down. For those having problems: 1. Are you using chan_sip or pj_sip? (I only have experience with PJsip.) 2. Are you seeing...

I can say with certainty that call recordings work. I can't do the same with voicemail because I pass the call to another server for that function, but I would be very surprised to find out voicemail does not work. I'm also pleased to report that TLS and SRTP seems to be working with the...

As luck would have it, I do have IPv6 on my test server. I do not have any "network unreachable" errors in my logs and a quick "lsof" of asterisk shows two open IPv6 ports: lsof -c asterisk | grep -i ipv6 asterisk 30586 asterisk 12u IPv6 3188795947 0t0 TCP *:8089 (LISTEN)...

I noticed when compiling the above that it generated two warnings: [CC] sdp_srtp.c -> sdp_srtp.o sdp_srtp.c: In function ‘ast_sdp_crypto_process’: sdp_srtp.c:263: warning: ‘crypto’ may be used uninitialized in this function sdp_srtp.c:275: warning: ‘tag_from_sdp’ may be used uninitialized in...

Did you try both http://... and https://... ?

You can have more than one swap device, including one partition and one file, but personally I view any total swap expanse in excess of whatever your base memory allocation is to be of little value: By the time you've shoved that much of your virtual address space onto swap you've got a...

Ward (and anyone else with an opinion): Given the choice between 13-13 on CentOS 6 and 13-13 on Ubuntu 18.04, which would you choose and why?

My experience with low cost VPS offerings in general is that the good ones tend to grow until they get big enough to attract the attention of the bigger fish, at which point they get bought out, almost always to the detriment of their customer base. HiFormance seems to be such an outlier --...

Yeah, I realize it makes absolutely no difference to Asterisk or Google, but it does make it clearer which GVsip trunk is which when navigating the Connectivity > Trunks page. That's the only reason I suggested it.

I've proffered the following fix be before but so far it has not been picked up and made official: To resolve this, edit pjsip_custom.conf and add the following line after each occurrence of "contact=sip: obihai.sip.google.com": qualify_frequency=120 (This should occur in each gvsipX context...

Unless I have misunderstood something, I have some bad news to report. Yesterday, I posted about some SSL errors I was seeing after adding the NAF updates to an Asterisk server running on an ARM-based PogoPlug. At the time, I believed it was due to a cockpit error of some sort. However, after...

There seems to be a fair amount of interest in GVsip for the PogoPlugs and/or the ARMel architecture. Personally, I'm using a PogoPlug E2. I did basically what Ward suggests: Since openssl 1.1.0 is not packaged (that I could find), I downloaded the source and compiled it using the configure...