*CLI> sip show peer 7052 load
…might do it. See this post from 2013: https://forums.asterisk.org/viewtopic.php?t=86342
Related, that Asterisk forum post is from the same year the Polycom SPIP 301 went End Of Life, so you might want to update the firmware or try a newer phone (even a soft phone on your computer would be helpful to test with.) Older phones tend to be a little slower…
I see you have NAT enabled with your SIP, but it looks like your devices can talk to each other without NAT. If this is the case, then you might try disabling NAT entirely, or, try configuring NAT with the following lines in your sip.conf, replacing your IP & subnets as appropriate:
Generally, if you can eliminate the NAT between your PBX and phones, then there is less that can go wrong.
For example, no NAT means you won’t need to qualify peers (this is the OPTIONS message that the PBX keeps sending.) These messages are critical for NAT because they help keep the UDP connections open on your firewalls; otherwise, inactive UDP ports will (usually) be reaped by the firewall in a few minutes or so, resulting in phones being unable to receive calls until they REGISTER again (which could be a while – see line in logs for Expires: 3600 ; also, minexpiry and maxexpiry in sip.conf)
Assuming no NAT, you might try disabling the qualify setting in sip.conf globally, for all peers:
…or change qualify to “no” for just this peer, as a test, by modifying the qualify column value for the peer’s row in the sippeers database table.
Other options include changing realtime settings in the “rt…” lines, but I’ve not used those recently.