VOS3000 SIP Routing Gateway Contact: Essential INVITE Header Guide
📞 When VOS3000 sends a SIP INVITE to a routing gateway, which header determines the callee number? Is it the To header, the Request-Line, or the Contact header? The answer depends on a critical — yet often overlooked — parameter: SS_SIP_ROUTING_GATEWAY_INVITE_USE_CONTACT, which governs the VOS3000 SIP routing gateway contact behavior for outbound INVITE messages. 🎯
🔄 By default, this parameter is set to Off, meaning VOS3000 uses the standard SIP convention where the callee number is taken from the To header. But when enabled (On), VOS3000 extracts the callee number from the request-line of the INVITE and preserves the original number in the To field. This subtle change has a significant impact on how calls are routed through your VoIP softswitch — especially when interfacing with gateways that rely on the Contact header or request-line for number identification. 🔧
📡 This guide covers everything you need to know about the VOS3000 SIP routing gateway contact setting — from the core parameter SS_SIP_ROUTING_GATEWAY_INVITE_USE_CONTACT to the related per-gateway SIP settings (Reply address, Request address, Peer number information) and Mapping Gateway callee/caller field selection. All data is sourced exclusively from the official VOS3000 V2.1.9.07 Manual, Section 4.3.5.2 (Table 4-3). For expert assistance, contact us on WhatsApp at +8801911119966. 💡
Table of Contents
🔐 What Is VOS3000 SIP Routing Gateway Contact?
📞 The VOS3000 SIP routing gateway contact parameter controls how VOS3000 constructs the SIP INVITE message when sending calls to a routing gateway. Specifically, it determines whether the callee number should be extracted from the request-line and whether the original number should be preserved in the To field. 📋
📌 According to the official VOS3000 V2.1.9.07 Manual, Table 4-3:
| Attribute | Value |
|---|---|
| 📌 Parameter Name | SS_SIP_ROUTING_GATEWAY_INVITE_USE_CONTACT |
| 🔢 Default Value | Off |
| 📝 Description | Use number from request-line as callee and keep original number in To field when send invite to callee |
| 🔀 Options | On / Off |
| 📍 Navigation | Operation management → Softswitch management → Additional settings → SIP parameter |
💡 Key insight: When this parameter is set to Off (default), VOS3000 follows the standard SIP RFC 3261 convention — the callee number in the INVITE is determined by the To header, and the request-line matches. When set to On, VOS3000 extracts the callee number from the request-line of the incoming SIP message and uses that for routing, while keeping the original number in the To field of the outbound INVITE. This is essential when upstream gateways manipulate the request-line during transit. 📡
🎯 Why VOS3000 SIP Routing Gateway Contact Matters
⚠️ Understanding and correctly configuring this parameter is critical for several reasons:
- 📞 Number routing accuracy: If the callee number source does not match what the downstream gateway expects, calls may be routed to the wrong destination or rejected entirely
- 🔄 To field preservation: Enabling this setting preserves the original dialed number in the To field, which is essential for billing, CDR accuracy, and troubleshooting
- 🔗 Gateway compatibility: Some SIP gateways and carrier equipment extract the callee number from the request-line rather than the To header — this parameter ensures compatibility
- 📊 Call flow integrity: Mismatched headers can cause call failures, one-way audio, or incorrect number display on the receiving end
- 🛡️ Interoperability: Different vendors implement SIP differently; this parameter gives you the flexibility to adapt VOS3000 to various gateway behaviors
⚙️ How SS_SIP_ROUTING_GATEWAY_INVITE_USE_CONTACT Works
🔄 To understand this parameter, you need to see how the SIP INVITE message changes based on the setting. Here is a text-based comparison of the two modes: 📡
📋 SIP INVITE Message Structure — SS_SIP_ROUTING_GATEWAY_INVITE_USE_CONTACT: ━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━ ⬇️ Mode: OFF (Default) ━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━ INVITE sip:[email protected] SIP/2.0 ← Request-Line Via: SIP/2.0/UDP 10.0.0.1:5060 From: "Caller" <sip:[email protected]>;tag=abc To: <sip:[email protected]> ← Callee = To header Contact: <sip:[email protected]:5060> Content-Type: application/sdp 📌 Result: Callee number = 8801234567 (from To header) Request-Line and To header contain the SAME number ━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━ ⬇️ Mode: ON ━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━ INVITE sip:[email protected] SIP/2.0 ← Request-Line (used for routing) Via: SIP/2.0/UDP 10.0.0.1:5060 From: "Caller" <sip:[email protected]>;tag=abc To: <sip:[email protected]> ← Original number PRESERVED Contact: <sip:[email protected]:5060> Content-Type: application/sdp 📌 Result: Callee number = 8801234567 (from Request-Line) To field keeps the ORIGINAL number (before any manipulation)
📊 Critical distinction: When the parameter is On, the request-line contains the number that VOS3000 uses for actual routing (the callee number), while the To header retains the original dialed number before any prefix manipulation or routing transformation. This is extremely useful when VOS3000 applies prefix conversion rules — the gateway receives the routing number in the request-line but the original number remains in the To field for reference. 🎯
📋 Per-Gateway SIP Settings: Reply Address and Request Address
🔗 Beyond the global SS_SIP_ROUTING_GATEWAY_INVITE_USE_CONTACT parameter, VOS3000 provides per-gateway SIP settings that control how reply and request signals are addressed. These settings are configured at the individual gateway level under Routing Gateway → Additional settings → Protocol → SIP. 🛠️
📨 Reply Address: Where to Send Reply Signal
📬 The Reply address setting determines where VOS3000 sends the reply signal after receiving a SIP request from the gateway. This is critical for ensuring SIP responses (such as 200 OK, 180 Ringing) reach the correct destination. 📡
| Option | Description | Recommendation |
|---|---|---|
| 🟢 Socket | Send reply to the source IP and port from which the SIP request was received | ✅ Recommended — most reliable for NAT traversal |
| 🔵 Via port | Send reply to the port specified in the Via header | ⚠️ Use when gateway requires Via-based routing |
| 🟡 Via | Send reply to the address and port in the Via header | ⚠️ Standard SIP behavior per RFC 3261 |
💡 Best practice: The Socket option is recommended for the Reply address because it ensures SIP responses are sent back to the actual source of the request, which is critical for NAT traversal scenarios. For a deeper understanding of SIP signal flow, see our VOS3000 SIP call flow guide. 📖
📤 Request Address: Where to Send Request Signal
📨 The Request address setting determines where VOS3000 sends the request signal after a call is established. This setting directly relates to the VOS3000 SIP routing gateway contact concept because it controls whether VOS3000 uses the Contact header or socket information for subsequent requests. 🔧
| Option | Description | Recommendation |
|---|---|---|
| 🟢 Socket | Send request to the source IP and port from which the SIP request was received | ✅ Recommended — most reliable |
| 🔵 Contact Port | Send request to the port specified in the Contact header | ⚠️ Use when gateway advertises a specific Contact port |
| 🟡 Contact | Send request to the full address in the Contact header | ⚠️ Standard SIP per RFC 3261 |
🔧 How this relates to the Contact header: The Request address setting determines how VOS3000 uses the Contact header for subsequent in-dialog requests (such as re-INVITE or BYE). When set to Contact, VOS3000 follows the SIP standard and sends requests to the URI in the Contact header. When set to Socket, VOS3000 uses the source socket address instead — which can be more reliable in NAT scenarios. This is especially important when combined with SS_SIP_ROUTING_GATEWAY_INVITE_USE_CONTACT. 📡
👤 Peer Number Information: Caller Selection Mode
📋 The Peer number information setting in the Routing Gateway SIP configuration determines how VOS3000 selects the caller number from incoming SIP signals. This works in conjunction with the Contact and request-line settings to ensure proper number identification for both caller and callee. 🎯
📞 For complete gateway configuration details, see our VOS3000 gateway configuration routing mapping guide. 🔧
🗺️ Mapping Gateway SIP Settings: Callee and Caller Field Selection
🔄 While the Routing Gateway settings control how VOS3000 sends INVITE messages, the Mapping Gateway settings control how VOS3000 receives and interprets incoming SIP signals. These are configured at Mapping Gateway → Additional settings → Protocol → SIP. 📡
📞 Callee Number Field Selection
🎯 The Mapping Gateway provides a setting to determine which SIP field VOS3000 uses to extract the callee number from incoming INVITE messages. This is the inbound counterpart to SS_SIP_ROUTING_GATEWAY_INVITE_USE_CONTACT: 📋
| Callee Source Field | Description | When to Use |
|---|---|---|
| 📩 To | Extract callee number from the SIP To header | ✅ Default — standard SIP behavior, use when gateways follow RFC 3261 |
| 📨 Request-Line | Extract callee number from the SIP Request-Line | ⚠️ Use when upstream gateway modifies the request-line with the routing number |
💡 Critical relationship: The Mapping Gateway callee setting (To vs Request-Line) is the receiving side of the same concept that SS_SIP_ROUTING_GATEWAY_INVITE_USE_CONTACT handles on the sending side. If an upstream system sends INVITE messages where the request-line contains the routing number (different from the To header), you must configure the Mapping Gateway to extract the callee from the Request-Line. Similarly, if your downstream gateway expects the callee in the request-line, enable SS_SIP_ROUTING_GATEWAY_INVITE_USE_CONTACT. 🔄
📞 Caller Number Field Selection
👤 The Mapping Gateway also provides settings for extracting the caller number from incoming SIP signals: 📋
| Caller Source Field | Description | When to Use |
|---|---|---|
| 📩 From | Extract caller number from the SIP From header | ✅ Default — standard SIP behavior |
| 🆔 Remote-Party-ID | Extract caller number from the Remote-Party-ID header | ⚠️ Use when upstream gateway sends caller ID in RPID header |
| 🖥️ Display | Extract caller number from the Display name portion of the From header | ⚠️ Use when caller ID is in the display name, not the URI |
📞 Practical tip: When connecting to carrier gateways that manipulate caller ID through P-Asserted-Identity or Remote-Party-ID headers, configure the Mapping Gateway caller field accordingly. For more on caller ID management, see our VOS3000 callee rewrite rule and prefix conversion guide. 🔧
🔄 VOS3000 SIP Routing Gateway Contact: Complete Signal Flow
📊 To fully understand how SS_SIP_ROUTING_GATEWAY_INVITE_USE_CONTACT interacts with per-gateway settings, here is a complete signal flow diagram: 📡
🔄 VOS3000 SIP Routing Gateway Contact — Complete Signal Flow:
┌──────────────┐ ┌──────────────┐ ┌──────────────┐
│ Calling │ SIP │ VOS3000 │ SIP │ Routing │
│ Endpoint │ INVITE │ Softswitch │ INVITE │ Gateway │
└──────┬───────┘ └──────┬───────┘ └──────┬───────┘
│ │ │
│─── INVITE ────────────►│ │
│ To: 8801234567 │ │
│ From: 8801999888 │ │
│ │ │
│ 📋 Mapping Gateway: │ │
│ Callee from: To/ │ │
│ Request-Line │ │
│ Caller from: From/ │ │
│ RPID/Display │ │
│ │ │
│ │ 📤 SS_SIP_ROUTING_ │
│ │ GATEWAY_INVITE_USE_ │
│ │ CONTACT = ? │
│ │ │
│ │─── INVITE ─────────────►│
│ │ ┌──────────────────┐ │
│ │ │ OFF (default): │ │
│ │ │ Request-Line: │ │
│ │ │ 8801234567 │ │
│ │ │ To: 8801234567 │ │
│ │ └──────────────────┘ │
│ │ ┌──────────────────┐ │
│ │ │ ON: │ │
│ │ │ Request-Line: │ │
│ │ │ 8801234567 │ │
│ │ │ To: ORIGINAL │ │
│ │ │ (preserved) │ │
│ │ └──────────────────┘ │
│ │ │
│ │ 📨 Reply address: │
│ │ Socket / Via port / Via│
│ │ 📤 Request address: │
│ │ Socket / Contact Port │
│ │ / Contact │
│ │ │
│◄── 200 OK ────────────│◄── 200 OK ─────────────│
│ │ │
└────────────────────────┴─────────────────────────┘
🎯 Key takeaway: The Mapping Gateway settings control what VOS3000 reads from incoming INVITE messages, while SS_SIP_ROUTING_GATEWAY_INVITE_USE_CONTACT and the per-gateway Reply/Request address settings control what VOS3000 writes into outgoing INVITE messages. For a comprehensive understanding of SIP session management, see our VOS3000 SIP session guide. 📖
⚙️ Step-by-Step VOS3000 SIP Routing Gateway Contact Configuration
🔧 Follow these steps to configure the VOS3000 SIP routing gateway contact and related settings on your system:
Step 1: Configure Global SS_SIP_ROUTING_GATEWAY_INVITE_USE_CONTACT 📋
- 🔐 Log in to VOS3000 Client with administrator credentials
- 📌 Navigate: Operation management → Softswitch management → Additional settings → SIP parameter
- 🔍 Locate SS_SIP_ROUTING_GATEWAY_INVITE_USE_CONTACT in the parameter list
- ✏️ Set the value:
- 🟢 Off (default) — Standard SIP behavior; callee from To header
- 🔵 On — Use request-line number as callee; preserve original in To field
- 💾 Save and apply the changes
Step 2: Configure Routing Gateway SIP Settings 📡
- 📌 Navigate: Operation management → Softswitch management → Routing gateway
- 🔍 Select the target routing gateway
- 🔧 Go to Additional settings → Protocol → SIP
- ⚙️ Configure the following settings:
- 📬 Reply address: Socket (recommended) / Via port / Via
- 📤 Request address: Socket (recommended) / Contact Port / Contact
- 👤 Peer number information: Set caller number selection mode
- 💾 Save gateway settings
Step 3: Configure Mapping Gateway SIP Settings 🗺️
- 📌 Navigate: Operation management → Softswitch management → Mapping gateway
- 🔍 Select the target mapping gateway
- 🔧 Go to Additional settings → Protocol → SIP
- ⚙️ Configure the following settings:
- 📞 Callee: To / Request-Line — determines which field VOS3000 reads for the callee number
- 👤 Caller: From / Remote-Party-ID / Display — determines which field VOS3000 reads for the caller number
- 💾 Save mapping gateway settings
Step 4: Verify with SIP Debug 🔍
📝 After configuration, verify the Contact header behavior by monitoring the SIP INVITE flow. Use the SIP debug tools to confirm that the request-line and To header are populated correctly. For comprehensive debugging techniques, see our VOS3000 SIP debug guide. 🔧
📊 VOS3000 SIP Routing Gateway Contact: When to Enable vs. Disable
🎯 The decision to enable or disable SS_SIP_ROUTING_GATEWAY_INVITE_USE_CONTACT depends on your specific gateway interoperability requirements. Here is a detailed comparison: 💡
| Aspect | Off (Default) | On |
|---|---|---|
| 📌 Callee Number Source | To header | Request-Line |
| 🔄 To Field Content | Same as request-line callee | Original number (preserved) |
| 📞 Request-Line | Matches To header | Contains routing number |
| 🛡️ RFC 3261 Compliance | ✅ Full compliance | ⚠️ Modified behavior |
| 🔧 Best For | Standard SIP gateways, carriers that follow RFC 3261 | Gateways that read callee from request-line, prefix conversion scenarios |
| 📊 Billing Impact | CDR shows final routing number | CDR can preserve original dialed number |
| 🔗 Compatibility | Broad — works with most gateways | Specific — needed for certain gateway types |
💡 Decision rule: Keep the default Off unless you encounter a specific gateway that routes based on the request-line callee number instead of the To header. Most modern SIP gateways follow RFC 3261 and use the To header. However, some legacy systems or carrier equipment may depend on the request-line — in those cases, enable the parameter to On. For help identifying which mode your gateway requires, contact us on WhatsApp at +8801911119966. 📞
📋 VOS3000 SIP Routing Gateway Contact: Complete Gateway Settings Reference
📊 Here is the complete reference for all per-gateway SIP settings related to the VOS3000 SIP routing gateway contact configuration: 📖
| Setting | Location | Options | Recommended |
|---|---|---|---|
| 📬 Reply address | Routing Gateway → Protocol → SIP | Socket / Via port / Via | ✅ Socket |
| 📤 Request address | Routing Gateway → Protocol → SIP | Socket / Contact Port / Contact | ✅ Socket |
| 👤 Peer number info | Routing Gateway → Protocol → SIP | Caller selection mode | Depends on gateway |
| 📞 Callee field | Mapping Gateway → Protocol → SIP | To / Request-Line | To (default) |
| 👤 Caller field | Mapping Gateway → Protocol → SIP | From / Remote-Party-ID / Display | From (default) |
🔧 For complete documentation on all SIP parameters, see our VOS3000 parameter description reference. 📖
📊 Deployment Best Practices by Gateway Type
🎯 Different gateway types require different configurations for the VOS3000 SIP routing gateway contact settings. Here are recommended configurations based on common deployment scenarios: 💡
| Gateway Type | SS_SIP_ROUTING_GATEWAY_INVITE_USE_CONTACT | Callee Field (Mapping) | Reply / Request Address |
|---|---|---|---|
| 🏢 Standard SIP carrier | Off (default) | To | Socket / Socket |
| 🔄 Legacy gateway (request-line routing) | On | Request-Line | Socket / Socket |
| 📞 PSTN gateway (prefix conversion) | On | Request-Line | Socket / Contact |
| 📡 NAT-traversed gateway | Off (default) | To | Socket / Socket |
| 🌐 Wholesale carrier (multiple prefixes) | On | Request-Line | Socket / Contact |
💡 Key pattern: Enable SS_SIP_ROUTING_GATEWAY_INVITE_USE_CONTACT (On) when your gateway performs prefix conversion and you need the original number preserved in the To field while the routing number goes in the request-line. For more on prefix conversion, see our VOS3000 routing optimization guide. 🔧
🛡️ Common VOS3000 SIP Routing Gateway Contact Problems and Solutions
⚠️ Misconfigured Contact header settings can cause a range of call routing issues. Here are the most common problems and their solutions:
❌ Problem 1: Calls Routed to Wrong Number After Prefix Conversion
🔍 Symptom: VOS3000 applies a prefix conversion rule, but the downstream gateway still routes the call using the original number instead of the converted number.
💡 Cause: SS_SIP_ROUTING_GATEWAY_INVITE_USE_CONTACT is set to Off, so both the request-line and To header contain the original number. The gateway reads the To header and ignores the prefix conversion.
✅ Solutions:
- 🔧 Enable SS_SIP_ROUTING_GATEWAY_INVITE_USE_CONTACT to On
- 📞 This puts the converted (routing) number in the request-line while preserving the original in the To field
- 📊 Verify with SIP debug that the request-line contains the correct routing number
❌ Problem 2: Gateway Rejects INVITE — 404 Not Found
🔍 Symptom: Downstream gateway returns 404 Not Found for calls that should be routable, even though the callee number exists in the gateway’s routing table.
💡 Cause: The gateway extracts the callee number from a different header than what VOS3000 is populating. For example, the gateway reads the request-line but VOS3000 is only populating the To header (parameter set to Off).
✅ Solutions:
- 🔧 Confirm which SIP field the downstream gateway uses for callee identification
- 📋 If the gateway reads the request-line, enable SS_SIP_ROUTING_GATEWAY_INVITE_USE_CONTACT
- 📡 Check the gateway documentation or contact the carrier for their header requirements
❌ Problem 3: CDR Shows Incorrect Original Number
🔍 Symptom: Call Detail Records show the routing number (with prefix) instead of the original dialed number, making billing reconciliation difficult.
💡 Cause: SS_SIP_ROUTING_GATEWAY_INVITE_USE_CONTACT is Off, so the To header always contains the same number as the request-line — no original number is preserved.
✅ Solutions:
- 🔄 Enable SS_SIP_ROUTING_GATEWAY_INVITE_USE_CONTACT to On — this preserves the original number in the To field
- 📊 CDR systems can then read the original number from the To field for billing accuracy
- 📞 For billing system configuration, see our VOS3000 billing system guide
❌ Problem 4: One-Way Audio After Enabling Contact Header Routing
🔍 Symptom: After enabling SS_SIP_ROUTING_GATEWAY_INVITE_USE_CONTACT, calls connect but audio only flows in one direction.
💡 Cause: The Request address setting is configured to use Contact or Contact Port, and the Contact header in the gateway’s 200 OK response points to an incorrect or unreachable address.
✅ Solutions:
- 🔧 Change the Request address to Socket — this ensures subsequent requests go to the actual source IP:port
- 📡 Verify the Contact header in the gateway’s responses using SIP debug
- 🛠️ If the gateway is behind NAT, Socket-based routing is more reliable than Contact-based routing
- 🔍 For detailed troubleshooting steps, see our VOS3000 troubleshooting guide
💡 VOS3000 SIP Routing Gateway Contact Configuration Checklist
✅ Use this checklist when deploying or tuning your VOS3000 SIP routing gateway contact settings:
| Check | Action | Status |
|---|---|---|
| 📌 1 | Determine if downstream gateway reads callee from To or Request-Line | ☐ |
| 📌 2 | Set SS_SIP_ROUTING_GATEWAY_INVITE_USE_CONTACT to On if gateway uses Request-Line | ☐ |
| 📌 3 | Configure Routing Gateway Reply address (Socket recommended) | ☐ |
| 📌 4 | Configure Routing Gateway Request address (Socket recommended for NAT) | ☐ |
| 📌 5 | Configure Mapping Gateway Callee field (To or Request-Line) | ☐ |
| 📌 6 | Configure Mapping Gateway Caller field (From, Remote-Party-ID, or Display) | ☐ |
| 📌 7 | Test with SIP debug — verify INVITE header fields match expected values | ☐ |
| 📌 8 | Verify CDR records show correct callee number for billing | ☐ |
📞 Need help configuring your gateway Contact header settings? Contact our VOS3000 experts on WhatsApp at +8801911119966 for personalized assistance. 🔧
❓ Frequently Asked Questions
❓ What is SS_SIP_ROUTING_GATEWAY_INVITE_USE_CONTACT in VOS3000?
📋 SS_SIP_ROUTING_GATEWAY_INVITE_USE_CONTACT is a VOS3000 SIP parameter that controls how the callee number is placed in outbound INVITE messages to routing gateways. When set to Off (default), VOS3000 follows standard SIP behavior where the To header and request-line contain the same callee number. When set to On, VOS3000 uses the number from the request-line as the callee for routing and keeps the original number in the To field. This is essential for gateways that read the callee from the request-line rather than the To header. 🔧
❓ When should I enable VOS3000 SIP routing gateway contact?
🔄 Enable SS_SIP_ROUTING_GATEWAY_INVITE_USE_CONTACT (On) when your downstream routing gateway extracts the callee number from the SIP request-line instead of the To header. This is common with legacy PSTN gateways, gateways that perform number manipulation, and carrier equipment that routes based on the INVITE request-line. You should also enable it when you apply prefix conversion rules and need the original dialed number preserved in the To field for billing and CDR accuracy. 📡
❓ What is the difference between the Routing Gateway Contact setting and the Mapping Gateway Callee field?
📊 These settings control opposite directions of SIP signaling. SS_SIP_ROUTING_GATEWAY_INVITE_USE_CONTACT (Routing Gateway) controls what VOS3000 writes into outbound INVITE messages — it determines whether the callee comes from the request-line and whether the To field preserves the original number. The Mapping Gateway Callee field (To / Request-Line) controls what VOS3000 reads from inbound INVITE messages — it determines which SIP field VOS3000 uses to extract the callee number from incoming calls. They work together to ensure proper number handling in both directions. 🔄
❓ What does the Reply address Socket setting do in VOS3000?
📬 The Reply address setting in the Routing Gateway SIP configuration determines where VOS3000 sends SIP response messages (such as 200 OK, 180 Ringing, 403 Forbidden) after receiving a request from the gateway. When set to Socket (recommended), VOS3000 sends replies to the source IP address and port of the incoming SIP request. When set to Via, it uses the address in the Via header. When set to Via port, it uses the port from the Via header. The Socket option is most reliable for NAT traversal scenarios. 🛡️
❓ How does the Request address setting relate to the Contact header?
📤 The Request address setting controls where VOS3000 sends in-dialog SIP requests (like re-INVITE or BYE) after call establishment. When set to Contact, VOS3000 sends requests to the full URI in the Contact header of the gateway’s response. When set to Contact Port, it uses only the port from the Contact header. When set to Socket (recommended), it sends to the source IP:port of the received signal. This is closely related to the VOS3000 SIP routing gateway contact behavior because it determines how the Contact header is used for subsequent signaling. 🔧
❓ Can I configure SS_SIP_ROUTING_GATEWAY_INVITE_USE_CONTACT per gateway?
⚙️ The SS_SIP_ROUTING_GATEWAY_INVITE_USE_CONTACT parameter is a global SIP parameter configured at the system level under Operation management → Softswitch management → Additional settings → SIP parameter. However, the related per-gateway settings (Reply address, Request address, Peer number information for Routing Gateways; Callee and Caller field selection for Mapping Gateways) are configured at the individual gateway level. This means the base Contact header behavior is global, but the specific address routing and field selection can be customized per gateway. For system-level parameter documentation, see VOS3000 system parameters. 📖
❓ What happens to the To field when VOS3000 SIP routing gateway contact is enabled?
📞 When SS_SIP_ROUTING_GATEWAY_INVITE_USE_CONTACT is set to On, VOS3000 preserves the original callee number in the To field of the outbound INVITE. The request-line contains the number that VOS3000 uses for actual routing (which may include prefix modifications or routing transformations), while the To field retains the original dialed number before any manipulation. This dual-number approach ensures that downstream gateways can route using the request-line number while billing and CDR systems can reference the original number from the To field. 🎯
📚 Related Resources – VOS3000 SIP Routing Gateway
🔗 Explore these related VOS3000 guides for deeper understanding:
- 📞 VOS3000 SIP Call Flow — Complete SIP signal flow and message sequence
- 🔄 VOS3000 SIP Session — Session timer and call management configuration
- 🔍 VOS3000 SIP Debug Guide — Debug SIP signaling and troubleshoot header issues
- 📋 VOS3000 Parameter Description — Complete parameter reference for all SIP settings
- 📊 VOS3000 System Parameters — System-level parameter configuration guide
- 📞 VOS3000 Call Routing — Call routing configuration and optimization
- 🔧 VOS3000 Gateway Configuration Routing Mapping — Gateway setup and mapping configuration
- 📊 VOS3000 Gateway Route Prefix Billing — Prefix conversion and billing integration
- 🎯 VOS3000 Routing Optimization — Optimize routing with prefix and number manipulation
- 📖 VOS3000 Routing Guide — Comprehensive routing configuration guide
- 🔧 VOS3000 Callee Rewrite Rule Prefix Conversion — Number rewrite and prefix conversion settings
- 🛠️ VOS3000 Troubleshooting Guide — Fix common SIP signaling and routing issues
- 📦 VOS3000 Installation — Installation and initial setup guide
- 💰 VOS3000 Billing System — Billing configuration and CDR management
- 🌐 VOS3000 Official Downloads — Official software and documentation
📞 Need Professional VOS3000 Setup Support?
For professional VOS3000 installations and deployment, VOS3000 Server Rental Solution:
📱 WhatsApp: +8801911119966
🌐 Website: www.vos3000.com
🌐 Blog: multahost.com/blog
📥 Downloads: VOS3000 Downloads
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