I need a service about caller id spofing
Budget: $30 – $250 USD
I run a small Asterisk lab and I need to run controlled test calls where the caller ID can be set to any value I choose. The goal is strictly testing and development, so everything must remain inside a legal, closed-loop environment.
Here is what I need from you:
• Configure or show me how to configure my existing Asterisk 18 instance so I can present arbitrary caller IDs on outbound calls.
• Supply any dial-plan snippets, AGI scripts, or CLI commands required, along with clear commentary so I understand why each line is there.
• Recommend a SIP trunk or gateway configuration that reliably passes the spoofed CLI without rewriting it (I am open to using my current trunk or switching to one you suggest).
• Walk me through a short live demo—screen-share or recorded session is fine—so I can see a test call leave the PBX and arrive with the desired caller ID.
• Hand over a concise checklist I can reuse when I spin up new instances of the lab.
I’m already comfortable inside the Asterisk CLI and with basic SIP debugging, so please focus on the caller-ID manipulation specifics rather than generic PBX setup. If you prefer FreeSWITCH or a hosted solution, mention why it would make the job easier, but the delivered instructions must work on raw Asterisk.
Looking forward to a clean, reproducible solution I can drop into future test environments.
Here is what I need from you:
• Configure or show me how to configure my existing Asterisk 18 instance so I can present arbitrary caller IDs on outbound calls.
• Supply any dial-plan snippets, AGI scripts, or CLI commands required, along with clear commentary so I understand why each line is there.
• Recommend a SIP trunk or gateway configuration that reliably passes the spoofed CLI without rewriting it (I am open to using my current trunk or switching to one you suggest).
• Walk me through a short live demo—screen-share or recorded session is fine—so I can see a test call leave the PBX and arrive with the desired caller ID.
• Hand over a concise checklist I can reuse when I spin up new instances of the lab.
I’m already comfortable inside the Asterisk CLI and with basic SIP debugging, so please focus on the caller-ID manipulation specifics rather than generic PBX setup. If you prefer FreeSWITCH or a hosted solution, mention why it would make the job easier, but the delivered instructions must work on raw Asterisk.
Looking forward to a clean, reproducible solution I can drop into future test environments.
Related categories:
Linux
Asterisk PBX
VoIP
Telecommunications Engineering
Debian
A/V Engineering
API Development
SIP