AI Outbound Calling System Completion
Budget: ₹1,500 – ₹12,500 INR
Title: Complete Two-Mode AI Calling System Using TeleCMI, Asterisk, Sarvam and ElevenLabs
Budget: Fixed price
Duration: 1 to 3 days
Type: Debugging, integration, testing and handover
We need an experienced Asterisk and voice AI engineer to complete an existing outbound AI calling system for Silent Capital.
Required architecture:
TeleCMI PSTN
to Asterisk 18.10 on a GCP Mumbai VM with static public IPv4
to Silent Voice Kernel
to Deepgram Flux for speech recognition
to Sarvam 30B for conversation
to selected TTS provider
to Asterisk
to TeleCMI
to customer
The system must support two selectable modes using the same calling pipeline:
FREE mode:
Sarvam Bulbul v2 TTS
PREMIUM mode:
ElevenLabs using Mr. Kumar's cloned voice
Switching must require only a configuration change such as:
VOICE_MODE=FREE
VOICE_MODE=PREMIUM
Do not build two separate systems.
Current setup:
- Ubuntu 22.04
- Asterisk 18.10 using res_pjsip
- GCP Mumbai VM with reserved static public IPv4
- TeleCMI or PIOPIY Indian SIP trunk
- Kamailio carrier SBC
- RTP ports 10000 to 20000
- alaw and ulaw codecs
- TeleCMI Register must remain OFF
- The same GCP static IP is entered in TeleCMI IP Whitelisting and Origin IP
- Only the SIP Trunk service should remain active in TeleCMI
- Inbound calls and audio already work
Current outbound issue:
Outbound INVITEs receive:
SIP/2.0 407 Proxy Authentication Required
Server: kamailio 5.7.7
Content-Length: 0
The 407 contains no Proxy-Authenticate header, so Asterisk cannot calculate digest authentication or retry.
Credentials, SBC hosts, SIP domains, number formats, caller ID, registration mode, balance, firewall and NAT have already been tested. Do not repeat them without evidence from SIP traces.
Scope of work:
- Diagnose and fix the outbound 407 rejection
- Identify any TeleCMI account, IP authentication, routing or Kamailio provisioning issue
- Coordinate with TeleCMI support and clearly state any carrier-side change required
- Keep only the SIP Trunk service active in TeleCMI
- Complete stable outbound SIP and RTP through Asterisk
- Connect the existing Silent Voice Kernel,
Budget: Fixed price
Duration: 1 to 3 days
Type: Debugging, integration, testing and handover
We need an experienced Asterisk and voice AI engineer to complete an existing outbound AI calling system for Silent Capital.
Required architecture:
TeleCMI PSTN
to Asterisk 18.10 on a GCP Mumbai VM with static public IPv4
to Silent Voice Kernel
to Deepgram Flux for speech recognition
to Sarvam 30B for conversation
to selected TTS provider
to Asterisk
to TeleCMI
to customer
The system must support two selectable modes using the same calling pipeline:
FREE mode:
Sarvam Bulbul v2 TTS
PREMIUM mode:
ElevenLabs using Mr. Kumar's cloned voice
Switching must require only a configuration change such as:
VOICE_MODE=FREE
VOICE_MODE=PREMIUM
Do not build two separate systems.
Current setup:
- Ubuntu 22.04
- Asterisk 18.10 using res_pjsip
- GCP Mumbai VM with reserved static public IPv4
- TeleCMI or PIOPIY Indian SIP trunk
- Kamailio carrier SBC
- RTP ports 10000 to 20000
- alaw and ulaw codecs
- TeleCMI Register must remain OFF
- The same GCP static IP is entered in TeleCMI IP Whitelisting and Origin IP
- Only the SIP Trunk service should remain active in TeleCMI
- Inbound calls and audio already work
Current outbound issue:
Outbound INVITEs receive:
SIP/2.0 407 Proxy Authentication Required
Server: kamailio 5.7.7
Content-Length: 0
The 407 contains no Proxy-Authenticate header, so Asterisk cannot calculate digest authentication or retry.
Credentials, SBC hosts, SIP domains, number formats, caller ID, registration mode, balance, firewall and NAT have already been tested. Do not repeat them without evidence from SIP traces.
Scope of work:
- Diagnose and fix the outbound 407 rejection
- Identify any TeleCMI account, IP authentication, routing or Kamailio provisioning issue
- Coordinate with TeleCMI support and clearly state any carrier-side change required
- Keep only the SIP Trunk service active in TeleCMI
- Complete stable outbound SIP and RTP through Asterisk
- Connect the existing Silent Voice Kernel,