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One-Way Audio, 503s, and the SIP Problems That Eat Your Shift
How-ToDevOps

One-Way Audio, 503s, and the SIP Problems That Eat Your Shift

via Dev.to DevOpsJason Shouldice

You're staring at the Asterisk CLI. Calls are connecting but nobody can hear anything. Or the agent hears the customer fine, but the customer hears dead air. Or calls fail immediately with a 503 and your trunk shows "UNREACHABLE." Every one of these problems came from a real VICIdial system I fixed. Here's what to check and in what order, because the order matters — you don't want to spend two hours debugging codec mismatches when the problem is a blocked UDP port. Before You Touch Anything: Get Your Tools Ready # This is the single most useful SIP debugging command sngrep -d eth0 port 5060 # Check current trunk status asterisk -rx "sip show peers" # Active channels with codec info asterisk -rx "core show channels verbose" sngrep shows you the full SIP ladder diagram — INVITE, 100 Trying, 180 Ringing, 200 OK, ACK, BYE — in a visual flow. If you don't have it installed: yum install sngrep or apt install sngrep . This tool will save you more hours than any config change. Problem 1: One-W

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