Add isolated SIP call tool, optional debug capture and split interface schemas
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# Copy to a private file, chmod 600, then fill your provider-approved values.
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# Documentation-only address: this template cannot test a real line as-is.
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# Plain KEY=VALUE only: no quotes, shell commands, variables or duplicate keys.
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SIP_ID=test-line
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SIP_SERVER=192.0.2.10
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SIP_PORT=5060
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SIP_CALLER_ID=YOUR_CALLER_ID
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# Empty is valid, but this field must be present. Raw CLI number is unchanged.
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SIP_PREFIX=
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# Only these supported values are accepted; no REGISTER or Digest credentials.
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SIP_TRANSPORT=udp
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SIP_AUTH=ip
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SIP_REGISTER=no
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SIP_CODEC=alaw
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# Existing native Asterisk installation and the user's private ari-secret.
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ASTERISK_CONFIG_DIR=/home/rogee/.config/go-sip-asterisk
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ASTERISK_BIN=/usr/local/sbin/asterisk
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ASTERISK_LIBRARY_DIR=/usr/local/lib
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OUTPUT_DIR=/home/rogee/.local/state/sip-call
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# Ring: 1-120 seconds. Connected hold: 1-30 seconds. No redial.
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RING_SECONDS=20
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HOLD_SECONDS=3
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# Used only with --debug/-D; no tcpdump/tshark needed otherwise.
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DEBUG_INTERFACE=any
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