VoIP Provider for Call Simulation Platform

Job ID: 40391262

Budget: $100 – $175 USD

**VoIP/SIP Provider Needed for Custom Caller ID and DTMF Integration**

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### **Job Description:**
We are looking for a **VoIP/SIP provider** that can supply the following features for our **call simulation platform**. The provider must support **custom caller ID spoofing**, **DTMF relay**, and **real-time call monitoring**.

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### **Requirements:**
#### **1. SIP Credentials**
- Provide **SIP Server Address** (e.g., `nyc.voip.ms` or `sip.flowroute.com`).
- Provide **SIP Port** (e.g., `5060` for UDP or `5061` for TLS).
- Provide **SIP Username and Password** for authentication.

#### **2. Caller ID Spoofing**
- Must allow **custom caller IDs** (e.g., `+1800CHASE` or local bank branch numbers).
- **No strict verification** for non-toll-free numbers.
- **Support for toll-free spoofing** (if possible).

#### **3. DTMF Relay**
- Must support **DTMF via SIP INFO** (preferred for reliability).
- **DTMF via RTP** as a fallback option.

#### **4. Real-Time Call Monitoring**
- Must support **SIP NOTIFY** for call status updates (e.g., answered, ended).
- **Webhooks for DTMF/call events** (preferred for easy integration).

#### **5. API Access (Optional but Helpful)**
- Provide a **REST API** to initiate and hang up calls programmatically.
- Provide **API documentation** for call control.

#### **6. Global/Local Number Support**
- Must support **local numbers** (e.g., bank branch lines).
- Must support **international numbers** (if needed).

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### **Deliverables:**
1. **SIP Credentials** (server address, port, username, password).
2. **Confirmation** that custom caller IDs (including toll-free and local numbers) are allowed.
3. **Documentation** on how to configure DTMF relay (SIP INFO/RTP).
4. **Documentation** on real-time call monitoring (SIP NOTIFY/webhooks).
5. **API documentation** (if available) for call control.

### **Additional Notes:**
- This project is for **authorized call simulation and testing purposes only**.
- We prefer providers that offer **free trials or demo accounts** for testing.

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### **How to Apply:**
Please respond with:
1. The **name of the VoIP/SIP provider (can be you if possible)** you recommend.
2. Confirmation that the provider meets **all the requirements** listed above.
3. **Pricing details** (per-minute rates, setup fees, etc.).
4. **Steps to set up** the SIP trunk and configure custom caller IDs.
5. Any **additional documentation or support** you can provide.
OUTBOUND USA
Related categories: Linux Android Asterisk PBX API SIP REST API