6.1 Overview

The SIP Trunks section allows AIUNIFY Assistant to connect directly to compatible telephone carriers and business voice systems without requiring ordinary Twilio Programmable Voice calling.

SIP stands for:

A SIP Trunk can connect AIUNIFY Assistant to:

The AIUNIFY Assistant interface describes SIP Trunks as connections to local carriers such as STC, Mobily, and Zain, as well as other SIP-compatible providers. The application also includes provider presets for services such as Twilio SIP Trunking, Telnyx, and generic SIP providers.

Understanding the SIP Architecture

6.2 SIP Trunk Versus Phone Number

A SIP Trunk and a Phone Number are separate resources.

SIP Trunk

Contains the technical connection to the carrier:

Phone Number

Represents an actual telephone number that uses the SIP Trunk.

The relationship is:

Creating a SIP Trunk does not automatically create a Phone Number.

After configuring the Trunk, it must be assigned to a SIP Phone Number from the Phone Numbers section. The system rejects a Phone Number configured as SIP when no SIP Trunk has been selected.

6.3 SIP Provider Versus AIUNIFY Assistant

The SIP provider supplies the external telephone service.

AIUNIFY Assistant supplies:

The provider is responsible for the carrier connection, number availability, call termination, origination, permitted destinations, and applicable provider billing.

6.4 Asterisk

AIUNIFY Assistant uses Asterisk as the server-side voice engine for SIP calls.

Asterisk is responsible for functions such as:

The SIP Trunks page reports whether AIUNIFY Assistant is currently connected to Asterisk and displays the number of active SIP calls.

6.5 ARI

AIUNIFY Assistant communicates with Asterisk through the:

commonly abbreviated:

ARI allows the Assistant backend to manage active call channels, bridges, audio streams, inbound routing, outbound origination, and call termination.

A SIP Trunk may be correctly entered while SIP calling still fails when the ARI connection is offline.

6.6 PJSIP

The application uses Asterisk’s PJSIP system to represent SIP Trunks as telephone endpoints.

When SIP configuration is applied, AIUNIFY Assistant generates or refreshes Asterisk configuration information for:

Normal users should configure this through the AIUNIFY Assistant interface rather than manually editing Asterisk files.

6.7 SIP Calling Workflow

A typical outbound SIP call follows this process:

A typical inbound SIP call follows:

When to Use SIP

6.8 Appropriate SIP Use Cases

SIP can be appropriate when the organization:

6.9 When Twilio May Be Simpler

Twilio may be simpler when the organization:

SIP generally requires more networking and carrier-specific configuration than a standard Twilio number.

6.10 Technical Assistance

SIP configuration can involve:

A normal account user can enter and test a Trunk through the interface, but some failures require assistance from:

Opening SIP Trunks

6.11 Accessing the Page

After signing in:

1. Locate SIP Trunks in the Sidebar.
2. Select it once.
3. Wait for the loading indicator.
4. Review the Asterisk status.
5. Review the Configured Trunks table.

The page contains:

6.12 Loading Failure

When the page cannot retrieve the Trunks or Asterisk status, it can display:

Possible causes include:

Refresh once, confirm another protected page works, and sign in again when necessary.

Asterisk Status Bar

6.13 Asterisk Connected

A green status displays:

This means the Assistant backend reports an active ARI connection to Asterisk.

It indicates that the Voice Engine is reachable, but it does not prove that:

A Trunk test and telephone test call are still required.

6.14 Asterisk Offline

A red status displays:

This means the backend does not currently have an active ARI connection.

SIP calling will not operate normally while this condition remains.

Twilio calling can continue independently because the application treats SIP/Asterisk as a separate calling path.

6.15 Active SIP Calls

The status bar displays:

This value represents SIP calls currently tracked by the Assistant voice engine.

It can include:

Do not reload or restart Asterisk during active calls unless an emergency requires it.

6.16 Configured Trunk Count

The status bar displays the number of saved SIP Trunks.

Example:

This count does not mean all three are active, tested, or usable.

Review each Trunk’s:

Configured Trunks

6.17 Empty Trunk Account

When no Trunks exist, the page displays:

and explains that a SIP Trunk must be added to connect to a telecommunications provider.

Select:

to begin.

6.18 Trunk Information

A saved Trunk can contain:

The database supports the statuses:

and test results:

6.19 Actions

Each Trunk provides operational actions such as:

The page also provides a separate Reload Asterisk action that applies SIP configuration to the Voice Engine.

Collecting Provider Information

6.20 Required Provider Details

Before creating a Trunk, obtain the official SIP details from the carrier.

These can include:

Do not guess provider values.

6.21 Carrier Documentation

Use current documentation supplied directly by the carrier.

SIP hostnames, access rules, and regional endpoints can differ based on:

A host used by another organization may not work for your account.

6.22 Authentication Models

SIP providers commonly use one of two approaches.

Credential authentication

Requires:

IP authentication

The provider authorizes the public IP address of the AIUNIFY/Asterisk server.

With IP authentication, Username and Password may be blank.

Some providers support both methods.

6.23 Public Server IP

When the carrier uses an IP Access Control List, provide the public outbound IP address of the Asterisk server.

Do not provide:

The carrier must authorize the server that sends the SIP traffic.

Adding a SIP Trunk

6.24 Opening the Trunk Dialog

To create a connection:

1. Open SIP Trunks.
2. Select Add SIP Trunk.
3. Select a provider preset or manual configuration.
4. Enter the provider information.
5. Review every value.
6. Save the Trunk.
7. Test connectivity.
8. Reload Asterisk.
9. Assign it to a Phone Number.

6.25 Provider Presets

The application contains preset information for several SIP providers.

A preset can automatically suggest values such as:

The preset values may contain placeholders.

Example:

A placeholder must be replaced with the actual value supplied for the organization’s account.

Do not save placeholder text as production configuration.

6.26 Generic Provider

Use the Generic or manual option when:

Trunk Name

6.27 Purpose of the Name

The Trunk Name is an internal identifier.

Recommended examples:

Avoid:

6.28 Unique Operational Names

Use a naming convention that distinguishes:

Example:

This reduces the risk of assigning the wrong Trunk to a Phone Number.

Provider Name and Region

6.29 Provider Name

Provider Name identifies the carrier in the interface.

Examples:

The provider name is descriptive. The actual network connection depends on Host, Port, Transport, and authentication.

6.30 Region

Region identifies the geographic area associated with the provider endpoint.

Examples:

Region does not automatically restrict Campaign Leads or calling hours.

The business must still manage:

SIP Host

6.31 Host or Registrar

The Host is the network address of the SIP provider.

It may be:

Examples of format:

Do not include:

6.32 Host Accuracy

A single typing error can prevent:

Copy the value from the provider documentation and remove accidental spaces.

6.33 Regional Hosts

Some providers offer different hosts for:

Choose the endpoint approved for the account and nearest to the Asterisk server where recommended by the provider.

SIP Port

6.34 Default Port

The default Trunk port is:

This is commonly used for UDP or TCP SIP signaling.

6.35 TLS Port

Secure TLS SIP commonly uses:

The exact port must match the carrier configuration.

Do not select 5061 merely because TLS was chosen unless the provider confirms it.

6.36 Valid Port

The Port field should contain only the provider-approved numeric port.

Examples:

A reachable port does not prove authentication or audio routing is correct.

Transport

6.37 Supported Transport Methods

The Trunk model supports:

6.38 UDP

UDP is commonly used for traditional SIP signaling.

Characteristics include:

A UDP test timeout is not conclusive proof that the Trunk cannot place calls.

6.39 TCP

TCP uses a connection-oriented transport.

It can be selected when the provider requires:

The Assistant connectivity test attempts to open a TCP connection to the configured Host and Port.

6.40 TLS

TLS provides encrypted SIP signaling when supported and correctly configured.

Selecting TLS does not automatically guarantee a secure working connection. The provider may also require:

The basic interface test treats TCP and TLS as network port reachability tests. It should not be interpreted as a complete TLS certificate or SIP registration test.

6.41 Matching Provider Transport

The transport selected in AIUNIFY Assistant must match the provider’s expected connection.

Example:

Using UDP with a TLS-only endpoint will fail.

SIP Username

6.42 Authentication Username

The Username is the SIP authentication identity supplied by the provider.

It may differ from:

Use the exact SIP username supplied for the Trunk.

6.43 Blank Username

Leave Username blank only when:

Do not leave it blank merely because the provider’s customer portal uses another login.

SIP Password

6.44 Credential Password

The SIP Password authenticates the Trunk to the carrier.

It is not necessarily the same as the provider portal password.

Use the Trunk-specific credential supplied or created through the provider.

6.45 Encrypted Storage

AIUNIFY Assistant stores the SIP Password in encrypted form.

API responses do not return the encrypted value or reveal the decrypted Password. Instead, the interface receives an indication that a Password has been set.

6.46 Editing Without Changing the Password

When editing a Trunk, leaving the Password field blank preserves the currently stored Password.

This allows a user to change:

without re-entering the secret.

The frontend removes an empty Password from the update request unless the user deliberately chooses to clear the credentials.

6.47 Replacing the Password

To replace it:

1. Open Edit.
2. Enter the new SIP Password.
3. Save the Trunk.
4. Reload Asterisk.
5. Test the Trunk.
6. Place a controlled test call.

6.48 Clearing Credentials

When the interface provides the clear-credentials option, using it can clear both:

Use this only when converting the Trunk to IP authentication or intentionally removing access.

Clearing credentials accidentally can cause:

Authentication Realm

6.49 Realm Definition

The Authentication Realm identifies the SIP authentication domain.

It may be:

Example:

6.50 Realm Versus Host

Host and Realm can be the same, but they are not always identical.

Example:

Use the provider’s exact values.

6.51 Blank Realm

Leave it blank only when the provider documentation does not require a separate Realm.

An incorrect Realm can cause authentication failures even when Username and Password are correct.

Default Caller ID

6.52 Caller ID Purpose

Default Caller ID is the number the SIP Trunk can use as the originating identity for outbound calls.

Recommended format:

The value should represent a number:

6.53 Trunk Caller ID Versus Phone Number

The Trunk’s Default Caller ID provides carrier-level configuration.

The actual Phone Number assigned to an AI Agent also identifies the outbound number.

Keep these values consistent.

Example:

Conflicting values can cause:

6.54 Caller-ID Authorization

Do not enter a number that the provider has not approved.

The carrier may reject or rewrite unauthorized caller IDs.

The Friendly Name inside AIUNIFY Assistant does not control the caller-name text shown on the recipient’s telephone.

Codecs

6.55 Codec Definition

A codec determines how telephone audio is encoded and transmitted.

The Trunk’s Codec field accepts a comma-separated preference list.

The default is:

6.56 PCMU

PCMU is also commonly known as:

It is widely used in North American telephone environments.

6.57 PCMA

PCMA is also known as:

It is widely used internationally.

6.58 Other Codecs

A provider may support codecs such as:

Only enter codecs supported by:

Adding an unsupported codec does not install or license it.

6.59 Codec Order

The comma-separated order can represent preference.

Example:

means PCMU is preferred before PCMA where negotiation allows.

6.60 Audio Problems

Codec disagreement can cause:

Begin with the provider’s recommended G.711 codecs unless another requirement exists.

Saving a Trunk

6.61 Review Before Saving

Before selecting Save, confirm:

6.62 Creation Result

A successful operation displays:

The backend associates the Trunk with the signed-in user and attempts to regenerate and reload the Asterisk configuration after creation.

6.63 Save Error

A failure can display the backend message or:

Possible causes include:

6.64 Ownership

Each Trunk belongs to the account that created it.

Normal users can list, edit, test, or delete only their own SIP Trunks.

A Phone Number cannot use another user’s Trunk. The backend checks ownership when the number is saved.

Reloading Asterisk

6.65 Reload Asterisk Button

The SIP Trunks page contains:

This action regenerates Asterisk SIP configuration from the active Trunks and attempts to reload Asterisk.

The backend also attempts automatic regeneration after a Trunk is created, edited, or deleted. The manual button provides a direct way to apply or retry the configuration.

6.66 When to Reload

Use Reload Asterisk after:

6.67 Reload During Calls

Avoid reloading during active SIP conversations.

Before selecting it, review:

Wait until the count reaches zero when practical.

6.68 Reload Success

A successful result displays the server’s confirmation message.

Afterward:

1. Wait briefly.
2. Confirm Asterisk remains Connected.
3. Test the Trunk.
4. Place a controlled telephone test call.

6.69 Reload Warning

The action can return a warning such as:

This can mean configuration files were generated but Asterisk did not confirm a complete reload.

A server administrator may need to check:

6.70 Reload Failed

Possible causes include:

Normal users should record the exact message and contact Support.

Testing Trunk Connectivity

6.71 Test Action

The Trunk Test action checks basic connectivity between the AIUNIFY server and the configured SIP Host.

It records:

The test is not a complete telephone call and does not validate every calling feature.

6.72 TCP and TLS Test

For a Trunk using TCP or TLS, the backend attempts a TCP connection to:

A successful result can resemble:

This confirms that the network port accepted a connection.

It does not necessarily confirm:

6.73 TCP Failure

A result such as:

can indicate:

6.74 UDP Test

For UDP, the application sends a basic SIP OPTIONS request.

When the provider responds, the test can display:

6.75 UDP Timeout

The application can display:

This warning is important.

Some providers:

Therefore, a UDP timeout should be followed by a controlled telephone test rather than treated as conclusive failure.

6.76 Test Success Is Not Call Success

A reachable Host can still reject telephone calls because of:

Creating the SIP Phone Number

6.77 Next Required Step

After the Trunk is configured:

1. Open Phone Numbers.
2. Select Add Number.
3. Choose SIP Trunk as Provider.
4. Enter the actual telephone number.
5. Select the saved SIP Trunk.
6. Assign an inbound Agent or fallback.
7. Save the number.
8. Test calling.

6.78 SIP Number Requirement

A SIP Phone Number must have a SIP Trunk.

The backend responds:

when the Provider is SIP but no Trunk is assigned.

6.79 Number Format

Enter the provider-controlled number in international format.

Example:

The inbound SIP routing system performs flexible matching with and without certain number-format variations, but consistent international formatting is still recommended.

6.80 Selecting the Trunk

Choose the Trunk that actually carries the number.

Do not select a Trunk merely because it has a similar provider name.

Confirm:

Origination URI and Inbound Delivery

6.81 Provider Origination

For inbound SIP calls, the carrier must send calls to the public SIP address of the AIUNIFY/Asterisk server.

The provider may call this:

The exact URI and port must be obtained from the AIUNIFY platform administrator or deployment configuration.

6.82 SIP Signaling Port

The application deployment can use a configured public SIP port that may differ from the provider’s registrar port.

These are separate concepts:

Do not assume both are 5060.

6.83 RTP Audio Ports

SIP establishes the call, but the telephone audio normally travels through RTP ports.

A call can ring and connect while having no audio when RTP traffic is blocked.

The server firewall must allow the RTP range configured for AIUNIFY Assistant.

This is generally a server-administrator responsibility.

6.84 Provider IP Access

Some providers require AIUNIFY to approve their signaling addresses, while others require the carrier to approve the AIUNIFY server IP.

Confirm both directions:

Assigning an Inbound Agent

6.85 Inbound Agent

On the SIP Phone Number record, select the AI Agent that should answer calls received on that number.

The Agent should have:

6.86 Inbound Number Matching

When Asterisk receives an inbound SIP call, AIUNIFY Assistant searches for a SIP Phone Number matching the called number.

The matching process considers number variations, including forms with or without a leading plus sign.

When no matching SIP Phone Number exists, the call is treated as unavailable and ends.

6.87 No Inbound Agent

When the SIP Phone Number has no assigned Agent, the system checks for a Fallback Number.

When no Agent and no fallback are available, the call ends.

SIP Fallback Number

6.88 Purpose

The SIP Phone Number form provides an optional Fallback Number.

It is intended for inbound calls when no AI Agent is assigned.

Example:

6.89 Fallback Routing

The system attempts to route the fallback call through the same SIP Trunk.

The fallback destination must therefore be accepted by that carrier.

The inbound SIP manager checks the Phone Number’s fallback value and continues the call through the SIP dial plan when a Trunk is available.

6.90 Fallback Limitations

Fallback can fail because:

Test fallback separately before relying on it for customer calls.

6.91 Agent Versus Fallback

The reviewed SIP flow uses fallback specifically when no inbound Agent is mapped.

Do not assume fallback automatically activates when:

Human transfer behavior should be separately configured and tested through the Agent’s tools where supported.

Inbound SIP Leads

6.92 Matching an Existing Lead

For an inbound call, the caller’s number is normalized into digits and compared with Lead records owned by the same AIUNIFY user.

When a matching Lead exists, the call is associated with that Lead.

6.93 Creating a New Lead

When no Lead matches, AIUNIFY Assistant creates a new record similar to:

The Lead receives:

Tags.

The system can also trigger the configured leadCreated webhook event.

6.94 Updating the New Lead

After the call:

1. Open Leads.
2. Search the caller’s digits.
3. Replace the generated name where known.
4. Add Company and Tags.
5. Confirm the country code.
6. Preserve the number for future inbound matching.

Outbound SIP Calls

6.95 Assigning the SIP Number to an Agent

To place outbound calls through the Trunk:

1. Create the SIP Phone Number.
2. Open AI Agents.
3. Edit the intended Agent.
4. Select the SIP Phone Number as its outbound number.
5. Save the Agent.
6. Run a controlled test call.
7. Use that Agent in a Campaign.

6.96 Campaign Provider Selection

The Campaign uses the provider associated with the Agent’s outbound Phone Number.

When that Phone Number is SIP, the Campaign uses the SIP/Asterisk calling path instead of Twilio.

6.97 SIP Campaign Requirements

Before a SIP Campaign starts, the system checks that:

A SIP Campaign can return:

when ARI is offline.

6.98 Concurrent Call Limit

The SIP voice engine enforces a maximum number of concurrent SIP calls.

When the limit is reached, additional calls can be rejected or delayed.

The default shown in the backend is server-configurable.

Normal users should not assume that every Lead in a Campaign is called simultaneously.

SIP Telephone Test Calls

6.99 Connectivity Test Versus Telephone Test

There are two different SIP tests.

Trunk connectivity test

Checks network reachability to the SIP Host.

Phone Number Test Call

Attempts to originate an actual call through Asterisk and the SIP Trunk.

The second test is more meaningful for validating production readiness.

6.100 Test Call Requirements

A SIP Phone Number test call requires:

The application uses an Agent voice to play the test phrase.

When no Agent exists, the system can return:

When ElevenLabs is missing:

6.101 Test Destination

Enter the destination in full international format.

Example:

Use an approved internal telephone during initial testing.

6.102 Test Call Timeout

SIP calls can take longer to originate than basic provider checks.

The backend allows an extended test-call request period.

A timeout can display:

A timeout may occur before the carrier’s final result is received.

Check:

before repeating the request.

Recording SIP Calls

6.103 SIP Recording

When account recording is enabled, the SIP voice engine attempts to record calls through Asterisk.

The recording can later be associated with the Call Log.

6.104 Missing SIP Recording

Possible causes include:

6.105 Recording Compliance

SIP calls are subject to the same recording and privacy obligations as Twilio calls.

Review:

Auto Hangup and SIP

6.106 Call Duration Limits

Account-level Auto Hangup settings apply to SIP calling.

Separate timeout values can affect:

6.107 One-Minute Warning

For applicable inbound SIP calls, the voice engine can provide a warning approximately one minute before the configured maximum duration.

6.108 Immediate Termination

A zero-minute timeout can cause immediate termination.

Review account Settings before diagnosing a Trunk as defective.

Editing a SIP Trunk

6.109 Opening Edit

To edit:

1. Open SIP Trunks.
2. Find the correct Trunk.
3. Open its Actions menu.
4. Select Edit.
5. Review existing information.
6. Update only the approved values.
7. Save.
8. Reload Asterisk.
9. Test again.

6.110 Password Protection During Edit

The current Password is not displayed.

Leave the Password blank to preserve it.

Enter a value only when replacing it.

Use the clear-credentials control only when intentionally removing authentication.

6.111 Status

A Trunk can be set to:

through supported update behavior.

Active Trunks are included when Asterisk configuration is generated.

An inactive Trunk should not be treated as available for production calling.

6.112 Changing Host or Transport

Changes to Host, Port, or Transport can completely redirect the carrier connection.

Before saving:

6.113 Editing During Campaigns

Do not change a Trunk used by a running Campaign.

Stop or complete the Campaign first, wait for active calls to end, then make the change.

Deleting a SIP Trunk

6.114 Delete Protection

AIUNIFY Assistant prevents deletion when one or more Phone Numbers still use the Trunk.

The system responds:

6.115 Safe Deletion Procedure

1. Open Phone Numbers.
2. Identify every number assigned to the Trunk.
3. Stop affected Campaigns.
4. Reassign Agents where necessary.
5. Remove or change the Phone Numbers.
6. Confirm no active SIP calls remain.
7. Return to SIP Trunks.
8. Delete the Trunk.
9. Reload Asterisk.
10. Confirm Asterisk remains connected.

6.116 Provider Account Remains

Deleting the AIUNIFY record does not necessarily:

Complete those actions with the carrier where required.

SIP Error Messages

6.117 Authentication Required — 407

A SIP 407 response generally means the provider requires authentication.

AIUNIFY Assistant translates this into guidance similar to:

Check:

6.118 Endpoint Not Found

A message indicating that the SIP endpoint was not found can mean:

Reload Asterisk and contact Support when the endpoint remains unavailable.

6.119 Provider Unreachable — 503

A 503 response can mean:

The application advises checking the SIP Host or Registrar and provider availability.

6.120 SIP Timeout

A timeout means the provider did not respond within the expected period.

Check:

6.121 Asterisk Not Available

A SIP call can return:

This is a Voice Engine problem rather than a Lead or Campaign problem.

6.122 Concurrent Limit Reached

When the server has reached the active SIP call limit, the user can receive a message similar to:

Wait for existing calls to end.

Troubleshooting Asterisk

6.123 Asterisk Offline After Page Load

Try:

1. Refreshing the SIP Trunks page once.
2. Confirming the API is available.
3. Confirming no maintenance is underway.
4. Selecting Reload Asterisk only when authorized.
5. Contacting Support.

A server administrator may need to restart the Asterisk service or repair ARI configuration.

6.124 Asterisk Connected but Calls Fail

Check:

Asterisk Connected means the Assistant can communicate with its Voice Engine. It does not prove that the carrier accepted the call.

6.125 Asterisk Becomes Offline During Calls

Possible causes include:

Stop new Campaign activity and contact Support.

6.126 Reload Does Not Restore Connection

A configuration reload is not the same as restarting a stopped Asterisk service.

A server administrator may need to review:

Troubleshooting Trunks

6.127 Trunk Does Not Save

Check:

6.128 Test Reports TCP Failure

Check:

6.129 UDP Test Times Out

Do not immediately delete the Trunk.

Confirm:

1. Provider accepts UDP.
2. Correct port is used.
3. Server IP is approved.
4. Registration requirements are met.
5. Actual telephone test call works.

6.130 Test Succeeds but Call Fails

The test checks reachability, not full SIP authorization.

Review:

6.131 Trunk Works Outbound but Not Inbound

Check:

6.132 Trunk Works Inbound but Not Outbound

Check:

Audio Troubleshooting

6.133 Call Connects with No Audio

Possible causes include:

This generally requires server or network review.

6.134 One-Way Audio

One-way audio often indicates:

Record whether:

That distinction helps Support diagnose the media path.

6.135 Distorted Voice

Check:

Begin testing with PCMU or PCMA when supported by the carrier.

6.136 Delayed Conversation

Long pauses can result from:

Test each provider and region before production deployment.

Number and Routing Problems

6.137 Wrong Number Receives Calls

Check:

6.138 Inbound Number Not Found

The called-number format may not match a SIP Phone Number record.

Confirm:

6.139 Caller Reaches Fallback Instead of Agent

The SIP Phone Number may have:

Edit the number and select the intended Agent.

6.140 Fallback Does Not Ring

Check:

Campaign Troubleshooting

6.141 Campaign Says Voice Engine Is Offline

Open SIP Trunks and confirm Asterisk status.

Do not recreate the Campaign. Correct the Voice Engine connection first.

6.142 Campaign Starts but Calls Fail

Review Call Logs for:

6.143 Campaign Uses Twilio Instead of SIP

Check the Campaign’s Agent.

The Agent’s outbound Phone Number may still be a Twilio number.

Assign the intended SIP Phone Number and save the Agent.

6.144 Campaign Uses the Wrong SIP Trunk

The selected Agent may be assigned to a SIP Phone Number linked to another Trunk.

Review:

Security

6.145 Protect SIP Credentials

Never place SIP credentials in:

6.146 Provider Portal Security

Protect the carrier account with:

6.147 IP Restrictions

Where supported, restrict the Trunk to the AIUNIFY server IP rather than allowing calls from every address.

Coordinate IP changes before moving or rebuilding the server.

6.148 Toll Fraud

A compromised SIP Trunk can be used for unauthorized calls.

Watch for:

Stop the Trunk and contact the carrier when suspicious activity appears.

6.149 Credential Rotation

When rotating credentials:

1. Stop active Campaigns.
2. Wait for calls to end.
3. Change the credential with the provider.
4. Update AIUNIFY Assistant.
5. Reload Asterisk.
6. Test connectivity.
7. Place a telephone test.
8. Resume Campaigns.

Compliance

6.150 Calling Rules

Using SIP instead of Twilio does not change the organization’s responsibility for:

6.151 Carrier Requirements

Follow the provider’s:

Recommended Setup Workflow

6.152 Complete SIP Setup Order

Use this sequence:

6.153 Controlled Outbound Test

1. Use an internal Lead.
2. Confirm Asterisk Connected.
3. Confirm the Trunk test result.
4. Confirm the SIP Phone Number.
5. Confirm ElevenLabs.
6. Confirm an Agent exists.
7. Start a test call.
8. Verify caller ID.
9. Verify two-way audio.
10. Review the Call Log.

6.154 Controlled Inbound Test

1. Confirm provider Origination URI.
2. Call the SIP number from an outside telephone.
3. Confirm the correct Agent answers.
4. Confirm the opening message.
5. Test speech recognition.
6. Test Knowledge Base answers.
7. End the call normally.
8. Find the inbound Lead.
9. Review the SIP and inbound Tags.
10. Review recording and transcript.

6.155 Fallback Test

1. Temporarily remove the inbound Agent.
2. Confirm the Fallback Number.
3. Call the SIP number.
4. Confirm the fallback destination rings.
5. Restore the Agent after testing.
6. Retest normal inbound AI handling.

Do not perform this test during production hours without approval.

Checklists

6.156 Provider Information Checklist

Confirm:

6.157 Trunk Entry Checklist

Confirm:

6.158 Asterisk Checklist

Confirm:

6.159 Phone Number Checklist

Confirm:

6.160 Outbound Checklist

Confirm:

6.161 Inbound Checklist

Confirm:

6.162 Security Checklist

Confirm:

6.163 Pre-Deletion Checklist

Confirm:

Chapter Completion

At the end of this chapter, the user should understand the difference between SIP Trunks and SIP Phone Numbers; how Asterisk, ARI, PJSIP, carriers, Phone Numbers, AI Agents, and Campaigns work together; how to create a Trunk using a provider preset or manual configuration; how to enter Host, Port, Transport, credentials, Realm, Caller ID, and Codecs; how encrypted SIP Passwords are handled; how to reload Asterisk; how TCP, TLS, and UDP connectivity tests differ; how to create inbound and outbound SIP routing; how fallback forwarding works; how inbound SIP Leads are created; how to test real telephone calls; how to recognize common SIP errors; and how to protect SIP infrastructure from unauthorized use.