7 Asterisk Development Mistakes That Only Show Up After You Go Live

I've been building and fixing Asterisk-based systems for close to a decade now. PBX platforms, multi-tenant hosted solutions, IVR systems, call center dialers - the works. And the pattern I keep seeing is that most Asterisk projects don't fail during development. They fail after launch. The dev environment works perfectly. Calls connect, the dialplan routes correctly, voicemail picks up, CDRs get written. Everyone's happy. Then you go live, connect real SIP trunks, put actual traffic on it, and things start falling apart in ways nobody anticipated. Here are the mistakes I've seen repeatedly - not the beginner stuff, but the production-level problems that cost teams weeks of debugging and sometimes a full re-architecture. 1. Still using chan_sip when you should've migrated to PJSIP already I still run into Asterisk deployments using chan_sip in 2026. It technically works, sure. But chan_sip has been deprecated for years, it doesn't get security patches anymore, and it's missing features