Asterisk

Sometime in August 2025

Lets get Asterisk setup. With the shutdown of FireRTC and BobRTC, this is the only way to safely scambait. I chose Telnyx as my SIP trunk to reach the PSTN.

We need to build Asterisk 16.9.0 from source.

1. tar -xvf <path to asterisk> and cd into dir
2. ./configure Fix any dependancy issues by installing using your distro’s package manager.
3. make -j<num of cpu threads>
4. sudo make install
5. sudo make samples
6. sudo make config

Using this reference, lets setup SIP trunking with Telnyx.

In /etc/asterisk/pjsip_wizard.conf:

trunk_defaults
type = wizard

telnyx
endpoint/transport=0.0.0.0-udp
endpoint/allow = !all,ulaw,alaw,G729,G722
endpoint/rewrite_contact=yes
endpoint/dtmf_mode=rfc4733
endpoint/context = from-pstn
endpoint/force_rport = yes
aor/qualify_frequency = 60
sends_auth = yes
sends_registrations = yes
remote_hosts = sip.telnyx.com:5060
outbound_auth/username = username
outbound_auth/password = password
registration/expiration = 600

Where outbound_auth/username and pwd is the one you made when setting up the trunk. Make sure to select credential authentication.

In /etc/asterisk/pjsip_wizard.conf:

user_defaults
type = wizard
accepts_registrations = yes
sends_registrations = no
accepts_auth = yes
sends_auth = no
endpoint/context = from-internal
endpoint/allow = !all,ulaw,alaw,G729,G722
endpoint/dtmf_mode = rfc4733
endpoint/rewrite_contact = yes
endpoint/force_rport = yes
aor/max_contacts = 1
aor/remove_existing = yes
aor/minimum_expiration = 30

1001
endpoint/callerid = Test User <1001>
inbound_auth/username = 1001
inbound_auth/password = strong@pass123$

Configure your inbound user/pass to a strong credential. This will be used by your softphone to connect.

Ok we are ditching Telnyx because they don’t allow Clip No Screening Support. This basically means they will enforce caller id outbound to registered numbers only. This is a good list of good SIP trunks. I decided to go with Sinch. One downside is they require you to “Request access” to use elastic SIP trunking. Hopefully they allow it, I just want to do testing at this point lol.

2025-09-04

I decided to go with an unnamed SIP trunk. Lets say that a lot of SIP trunks don’t respect the From header or the PAI header. They simply ignore and rewrite the From header to the SIP DID. This won’t work for my purposes. I eventually found a trunk that actually respects these headers. I don’t want to name them here because they might shut it down and of course scammers can use my knowledge to perform robocalling. Just know that these SIP trunks still exist, you just need to actively look for one.