I'm looking for help with debugging a yate VOIP/SIP/RTP issue. I have a number setup to redirect to a sound file after a timeout (stand in for an voicemail IVR). When called from an internal number registered with yate it works. When called from the outside number (yate acting as a SIP client) it doesn't. Yate says it routed the call to the announcement but I don't see that reflected in the SIP/RTP traffic, the call keeps ringing until I hang up.
There's a new closed community VoIP network called TeleHamster that started in my area. The idea is cool—to use old school telephones to talk to friends and neighbors. "No screens, no ads, no AI, no cookies." The phone numbers are 4 digits long 😎
One of my favorite pastimes as a teen was calling BBSes, so I've installed Mystic on my old Thinkpad laptop and Mgetty monitors a Startech hardware-based USB modem, passing actual modem calls to the board.
So, my initial review of the Yaelink SIP-T42S before I have the in hand:
- The documentation exists, but can be a pain in the ass to track down exactly what you need.
- Looks like I've got everything already setup to serve the boot configurations and should work with my FreePBX installation. I'll probably get these setup as backups in case I lose a couple of Digium D40 phones during the competition.
- Firmware is still available for the phone, but it is EoL.
- English isn't the greatest in the documentation, but it gets the job done.
- Cheap as hell as I bought 15 of these for $34 which is absolutely insane.
Got all of the phones online and working, replaced two bad phones, added a third phone, and got the smol Netgear PoE switch working. The Cisco C1000 switch's configuration is good to go and all of the IP addresses/networks and phone numbers have been documented. Pretty sure everything is a go!