I wasn’t sure how to title this. Basically, I just want to clarify some inconsistencies in their documentation. When they say “Base URL” they have it listed as https://example.signalwire.com/api/laml/2010-04-01. Well, I couldn’t get that to work for me when I was trying to use it with Textit. I had to use https://example.signalwire.com/api/laml/2010-04-01/Accounts
Also, your AuthToken is your API token. Something you generate. I figured that out easily enough and it is a better implementation than Twilio‘s secret but the API base URL I really thought should have been listed with “Accounts” at the end of it.
Is free openheatmap.com better than Excel? For me it was by far. I used Excel to clean up my data but then it’s heat map options were just awful. I could barely tell there was a numerical difference between each state. I tried other colors but to no avail.
I took the data and imported it into openheatmap.com and bam! Worked like a charm. I was trying to plot some data on the 50 US States and was just having a miserable time doing it Excel. Excel isn’t free. Excel isn’t cheap. Sure, I could have tried libreoffice – which I did – but had some issues with all the data. 10,000+ rows is too much for a pivot table in a 32 bit application. Had to uninstall and install a 64 bit version.
For me, openheatmap was awesome. Putting your data into a visual representation has always been more helpful than displaying columns and rows of data. In this instance I was actually surprised at the results that it showed. The data will be used to direct marketing money to different states.
After all these years VPN technology is still frustrating. I spent hours yesterday and today trying to setup two different VPNs. One was IPSEC site to site and the other was openvpn.
The Site to Site was started yesterday. I was between to Sophos XG routers. I wasted quite a bit of time setting it up only to find out that both devices needed to be on the same firmware. This morning I updated the firmware on both routers and I was able to connect. Stability seemed fine on my side but the connection on the other end seemed a bit sluggish.
This leads me to OpenVPN. Have used OpenVPN for several years but only off and on. I have setup a number of OpenVPN servers but always on linux.
The goal for this setup was to setup a High Availability (HA) Cloud-based Asterisk system with a local Asterisk system. Apparently the key to get the master to fail over to the slave is they both need to be on the same subnet. I got them to sync but I couldn’t force the failover.
Working with this garbage called Windows Server 2008 I have failed miserably. I don’t have a linux server to use on this site. Well, I do but it’s already running the PBX in the VM and that is what has become highly unstable. Not sure why as I have the same setup working great in other locations. Hyper-V on this server is also very poor.
I also want to say that I am not even going to bother with PPTP. It’s never been reliable enough.
My passenger window would not roll down. I already knew why I just couldn’t remember what I needed to do to fix it.
Just like my IT and entertainment troubleshooting I want to leave this here so I am remember next time.
When the battery is disconnected on virtually any car it wipes the radio, clock and messes with the automatic windows.
The passenger window will work with the passenger side controls but not on the drivers side. Simply roll the passenger window down (using passenger side controls) just a little bit and then roll it back up by holding the button in the up position for a few seconds. This will reset the system and your window will work again.
I believe my wife’s Ford Fusion has the same commands to fix her windows.
Conversion failed. The transcoder crashed or failed to start up was annoying message for me. Thankfully it looks like it was an easy fix for me. Apparently all the movies I had on my Plex server I must have removed. I added one back and then it worked. However, before I did that I checked the permissions of the transcode folder and found no issues.
I was unable to find my plex server for a very long time. I got frustrated and left it. It took me SIX months to get back to it. SIX MONTHS! I was frustrated. Thankfully I have it figured out now.
My issue was one of simplicity and stupidity but the latter might be too harsh. My issues were actually two fold. Number 1, I had 2 devices that were registering with the same IP. Not sure how that happened but I had one set as a static DHCP and another as just a static on the device. Switched them both to static DHCP with the Plex server having a new IP. Rebooted and that was that. The second issue I came across – and this one took awhile to discover – was that I had my server under one plex account email and my plex clients under another account. I’m using ubuntu server to host so I have to edit an XML file. It varies depending on what OS you are using.
What’s weird for me is the name of the support article. It’s called “Why am I locked out of server settings and how do I get back in?” I wouldn’t have thought to use any of those keywords in a search. I get how it is the same issue but still weird for me.
I had been unable to call landlines for some strange reason. It was in the US Virgin Islands. My trunk provider is voip.ms. Kept getting a busy signal. I troubleshooted it with voip.ms and as always they found a way.
Very simple put, my caller ID was set to area code + number. 3407741340. This caller ID worked great for any US numbers that I tried and even my 340 cellphone. I had another system setup where I was also able to call landlines. This had me believe the problem was in my dialplans. Nope, I was wrong again. If I moved the trunk out of my asterisk system and used it as a stand alone SIP trunk I had the same issue.
The solution was very simple and something I hadn’t considered. I was missing a “1” in front of the phone number listed in the caller ID. That’s it. That is all it was..
I was trying to setup a Fanvil Phone to test it out. Ran into some issues setting it up with SignalWire. Then I remembered that I hadn’t posted a screenshot of it working on my Yealink phone.
It’s pretty straightforward. I do use TLS because it is available. You will see that I am using port 5061.
Here is a quick rundown:
Register name and User Name is both the same thing. This would be your SIP endpoint username. Do not include anything after the @ sign.
Password – You set this up when you created the endpoint
Server Host – This is the part that comes after the @ sign and usually is the name of your project followed by a string of letters and characters and ends with sip.signalwire.com
Port is 5061 for TLS.
Transport also TLS. You can use UDP but then change the port to 5060 in the above field.
Set this to whatever you want but if your hardware and software supports Opus then this is a no brainer. USE OPUS!!!! Normally I would have had G.729 second on the list but its relatively new to SignalWire support.
I’ve got it up and running on my Fanvil test phone as well now. I will add those screenshots later.
I couldn’t auto-save or just regularly save any post on WordPress 5.0. A google search turned up a few options but none of them worked for me.
Suggestions included changing permalinks then changing it back. There was a php.ini line that I tried. I think if I would have reverted to the old editor it might have worked.
While troubleshooting I checked Google Debugging at it was showing access-control-allow-headers errors. I searched for that and came up with almost nothing except one little gem of wisdom. it said that the URL I was using might be different than that of my site. Specifically, the hits were talking about http vs https. My SSL was good so I didn’t think that was the issue.
What I did notice was my admin page was lacking the www even though the admin section (and nothing in wp-settings) was www. I changed my settings to remove the www and then everything worked.
Be forwarned this will probably affect your SEO. Google showed only “www” links but now I am getting rid of the “www”.
I was unable to use star codes with obihai 1000 phones for a very long time. It wasn’t really a big deal before but was irritating that I couldn’t find an answer. Thanks to a post from voip.ms wiki i figured out that I had to change the ITSP digitmap line. One just has to add |*xx| after |xx.|
Not sure why this was even the case as it is this issue only relates to Obihai phones. In case anyone is curious I use these phones on a few different systems. They are all Asterisk based. Usually Wazo but sometimes direct to voip.ms.