I understand where you’re coming from Chris, but it’s extremely frustrating / confusing / worrisome that a firmware update to a router can destroy local communications like this… and in such a specific way. Even if I replace all of the phones malfunctioning because of this, what’s to stop the same thing from happening again in the near future???
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Over the past week, I have continued troubleshooting this and have uncovered some interesting tidbits.
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Tried alternate sip drivers on the PBX - both the Chan_SIP & PJSIP drivers behave the same way. (Which I kind of expected with the audio arriving at the phone clean, but wanted to test with as strange of an issue this has been.)
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Tried a Cisco 7961 phone. I have a few of these phones that were sent to me instead of the 7960 model and was unable to get them to work when installing the phone server. Figured that with as much experience I’ve had with VOIP since, that I’ll be able to make them work. Got the SIP firmware on it and see the server, but was unable to get the phone to register… asterisk kept complaining about an incorrect password. I gave up, the illusion of my being semi-competent when it comes to VOIP shattered.
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Totally on a fluke, I found that the frequency of the gap occurring seems to be related to the destination you’re calling. (!!!) I’ve been testing with the toll free number of one of our vendors because that’s what the department head gave me as an example. I mistakenly discarded my notes from that day and he was in a meeting, so I started testing with a local store that has an automated recording pick up calls. I called this store 45 times with each of the chan_sip and the pjsip drivers without it happening once!
(Also tried a different switch port during this time, wondering if that might have something to do with it, but was the same.) I then tested extensively using a local pharmacy and it was fine. Thinking it might be a local vs long distance thing, I looked up the number for a pharmacy in a different area - it was also fine. By this time (takes a bit of time to dial a phone 160 times, even hanging up after 5-7 seconds 
), the manager was out of his meeting and I was able to get the vendor phone number. Tested using that number with the same phone… five calls later, the gap happened. 
I got another toll free number with an automated answer and the very first call had the gap. 
Went to the manager & asked if it’s only toll-free numbers doing it and he said that was not the case. I’m completely perplexed by this, especially with what I found next. -
I remember installing the chan_sccp module on our system years ago, but could never make it work. I dug into this again and was able to figure out that the ‘Allow Networks’ and ‘Local Networks’ settings apparently weren’t being applied and rejecting the phone’s attempts to register no matter what I put in these fields in the SCCP Server Config. Found a bug report online that seemed similar, but the source I had on the server was significantly after the fix date, so I dunno what’s going on. Not being willing to give up, I found where this was happening and put in a quick / dirty hack to allow it, recompiled, reinstalled, and tested. The phone registered and I began testing. There’s no gap in the audio.
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So something in the change Meraki made to their firmware caused only outgoing calls to specific numbers to have a short gap in the audio at the front of the call only for specific brand/model phone and only if the phone is running a particular firmware. 
Swapped out a SIP phone with one running SCCP yesterday to test. Honestly don’t know if the chan_sccp module is solid enough to use in production, but it’s really the only option I can see short of replacing 70+ phones with a different make/model phone. (Which are going to have their own quirks / problems, as evidenced by my using the 2 s305’s we had left from a now-closed off-site location reading the caller ID with the ring for every incoming call, which is goofy to say the least. I’ve not found a way to make it shut up and just ring; since nobody’s responded to the thread I opened about it, so it’s probably built into the phone without a way to turn it off.) Would like to stay with SIP if possible as that’s what the PBX is based on without add-on modules, but not if it’s going to have goofy stuff like this happen.
Can anyone see anything I missed or should be trying??? Have to say that I’m really not looking forward to flashing nearly 70 phones and rebuilding each of the extensions… 