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Hey, I’m setting up a SIP integration using Ziwo a...
# ❓・ai-evy-support
r
@AI Evy Hey, I’m setting up a SIP integration using Ziwo as our telephony provider instead of Twilio, and I wanted to confirm a few things please Ziwo asked for Retell’s IPs to whitelist on their side, and also asked if they should share their IPs so we can whitelist them as well Just wanted to confirm: Do we need to whitelist Ziwo’s IPs on our side for this setup? Should we rely on the IPs listed in Retell's documentation, or do Retell provide specific/static IPs for SIP integrations? Is there anything else required from the telephony provider side (Ziwo) to ensure a smooth integration other than the IP? Just want to make sure we’re not missing anything before going live. Thanks a lot!
a
Thanks for tagging me @Rowan. I'll look through relevant sources and get back to you shortly.
​ To learn more, see the sources I considered: [[1] Custom Telephony Overview](https://docs.retellai.com/deploy/custom-telephony) [[2] Need Clarity on Whitelisting CIDR Block Email](https://community.retellai.com/t/need-clarity-on-whitelisting-cidr-block-email/179) [[3] Telnyx](https://docs.retellai.com/deploy/telnyx) ​
### Answer For setting up SIP integration with Ziwo, here's what you need to know: Retell's IP blocks for whitelisting: [(1) ]() -
18.98.16.120/30
(All regions) -
143.223.88.0/21
(certain United States traffic) -
161.115.160.0/19
(certain United States traffic) Ziwo should whitelist these IP blocks on their side to allow traffic from Retell's SIP server. Do you need to whitelist Ziwo's IPs on Retell's side? No. Retell doesn't have a static IP that requires whitelisting from your telephony provider's side. [(2) ]() The authentication for elastic SIP trunking works through either valid credentials (username/password) or an approved IP whitelist on the telephony provider's side. Additional requirements from Ziwo: Beyond IP whitelisting, ensure Ziwo's SIP trunk configuration includes: [(1) ]() - **Transport method**: TCP (recommended) , UDP, or TLS - **SIP server URI**:
sip:sip.retellai.com
- For specific transport methods, append to the URI: - TCP:
sip:sip.retellai.com;transport=tcp
- UDP:
sip:sip.retellai.com;transport=udp
- TLS:
sip:sip.retellai.com;transport=tls
You'll also need to set up authentication credentials (username/password) for the SIP trunk, which you'll use when importing your number into Retell. [(1) ]() [(3) ]()
Feel free to tag @AI Evy with additional questions. I also appreciate feedback, it helps me improve.
r
@User can I only whitelist this IP since it's all regions: 18.98.16.120/30?
a
Thanks for the feedback, glad I was helpful.
r
@AI Evy can I only whitelist this IP since it's all regions: 18.98.16.120/30?
a
Thanks for the follow-up, @Rowan. I'll get back to you soon.
### Answer Yes, you can whitelist only
18.98.16.120/30
since it covers all regions. [(1) ]() However, if you're calling US/Canada numbers, you should also whitelist the additional US-specific IP blocks to ensure all traffic is properly allowed: [(2) ]() -
143.223.88.0/21
(certain United States traffic) [(1) ]() -
161.115.160.0/19
(certain United States traffic) The documentation notes that "Some non US traffic might go from IPs not listed above, but will be migrated to the first IP block by 12/03/2025". For US-based calling, whitelisting all three IP blocks is recommended to avoid any potential connection issues, as traffic may originate from any of these ranges. [(2) ]() [(3) ]()
​ To learn more, see the sources I considered: [[1] Custom Telephony Overview](https://docs.retellai.com/deploy/custom-telephony) [[2] Outbound calls fail when whitelisting IP address block](https://community.retellai.com/t/outbound-calls-fail-when-whitelisting-ip-address-block/180) [[3] HELP Outbound calls don't work](https://discord.com/channels/1186424376300679209/1438261528364056667) ​
Feel free to tag @AI Evy with additional questions. I also appreciate feedback, it helps me improve.
r
@AI Evy if based in the UAE?
a
Thanks for the follow-up, @Rowan. I'll get back to you soon.
### Answer For UAE-based operations, you should whitelist
18.98.16.120/30
as it covers all regions. [(1) ]() The two additional US-specific IP blocks (
143.223.88.0/21
and
161.115.160.0/19
) are specifically for certain United States traffic. Since you're based in the UAE and not routing calls through US infrastructure, you likely only need the primary
18.98.16.120/30
block. However, the documentation notes that "some non US traffic might go from IPs not listed above, but will be migrated to the first IP block by 12/03/2025". This suggests that whitelisting
18.98.16.120/30
should be sufficient for your UAE setup, as all non-US traffic is being consolidated to this block. To be safe and ensure no connectivity issues, you could whitelist all three blocks, but for UAE-specific traffic, the primary
18.98.16.120/30
block should cover your needs.
​ To learn more, see the sources I considered: [[1] Custom Telephony Overview](https://docs.retellai.com/deploy/custom-telephony) ​
Feel free to tag @AI Evy with additional questions. I also appreciate feedback, it helps me improve.
r
@AI Evy my ai callbot is not following the prompt and I rewrotte over and over, how can I fix this?
a
Thanks for the follow-up, @Rowan. I'll get back to you soon.
Copy code
///## Tool Usage Instructions
1. Gather initial information about the customer's issue.
2. Determine the type of request:
 - If customer mentions "refund" or "money back":
 → Call function `transfer_to_support` immediately
 - If customer needs order status:
 → Call function `check_order_status` with order_id
 - If customer wants to change their order:
 → First call `check_order_status`
 → Then transition to modification_state
3. After retrieving information:
 - Always summarize what you found
 - Ask if they need additional help
 - If yes, determine next appropriate action
Consider Switching to Conversation Flow - If your agent handles complex logic or multiple tools, Conversation Flow agents provide more deterministic behavior than Single/Multi Prompt agents. They work better when you have: - Multiple decision branches (more than 3-4 conditional paths) - Tool coordination (using 5+ different functions/tools) - State management (tracking multiple variables throughout the conversation) - Reliability concerns (single prompt shows inconsistent behavior) Lock Down Critical Rules - Add strict rules at the top of your prompt to prevent hallucinations: [(2) ]()
Copy code
CRITICAL RULE:
You must ONLY offer the time exactly as written in {{availability_text}}.
You are not allowed to guess, infer, or suggest other dates.
If availability_text is missing, politely end the call.
To see examples of effective prompts, create a new agent in the Dashboard and explore the pre-built templates. [(1) ]()
### Answer If your AI agent isn't following your prompt, here are proven strategies to improve prompt adherence: [(1) ]() Use Sectional Prompts - Break your prompt into focused sections for better LLM comprehension:
Copy code
///## Identity
You are a friendly AI assistant for [Company Name].
Your role is to [specific purpose].
You have expertise in [relevant domains].
///## Style Guardrails
Be concise: Keep responses under 2 sentences unless explaining complex topics.
Be conversational: Use natural language, contractions, and acknowledge what the caller says.
Be empathetic: Show understanding for the caller's situation.
///## Response Guidelines

Return dates in spoken form: Say "January fifteenth" not "1/15".
Ask one question at a time: Avoid overwhelming the caller with multiple questions.
Confirm understanding: Paraphrase important information back to the caller.
///## Task Instructions
[Specific steps the agent should follow]
///## Objection Handling
If the caller says they're not interested: "I understand. Is there anything specific..."
If the caller is frustrated: "I hear your frustration, let me help resolve this..."
Add Explicit Tool Calling Instructions - If your agent uses tools/functions, define clear triggers:
​ To learn more, see the sources I considered: [[1] Prompt Engineering Guide](https://docs.retellai.com/build/prompt-engineering-guide) [[2] GHL + Make + Retell](https://community.retellai.com/t/ghl-make-retell/806) ​
Feel free to tag @AI Evy with additional questions. I also appreciate feedback, it helps me improve.
Thanks for the follow-up, @Rowan. I'll get back to you soon.
r
@AI Evy what if I want to use multi prompt?
a
​ To learn more, see the sources I considered: [[1] Single/Multi Prompt Agent Overview](https://docs.retellai.com/build/single-multi-prompt/prompt-overview) [[2] Create Retell LLM](https://docs.retellai.com/api-references/create-retell-llm) ​
### Answer If you want to use Multi-Prompt agents to improve prompt adherence, here's how they work: [(1) ]() Multi-Prompt Structure organizes conversations into a tree of states, where each state has: - **Focused Prompt**: Specific instructions for that conversation phase - **State-Specific Functions**: Only relevant tools available in each state - **Transition Logic**: Clear conditions for moving between states - **Context Preservation**: Variables and information flow between states Benefits for Prompt Following: 1. **Predictable Behavior**: Each state has a clear, focused purpose 2. **Easier Debugging**: Issues isolated to specific states 3. **Better Function Control**: Tools available only when appropriate 4. **Scalable Design**: Add new states without affecting existing ones Example Structure: [(2) ]()
Copy code
json
{
  "states": [
    {
      "name": "information_collection",
      "state_prompt": "You will follow the steps below to collect information...",
      "edges": [
        {
          "destination_state_name": "appointment_booking",
          "description": "Transition to book an appointment."
        }
      ],
      "tools": [
        {
          "type": "transfer_call",
          "name": "transfer_to_support",
          "description": "Transfer to the support team.",
          "transfer_destination": {
            "type": "predefined",
            "number": "16175551212",
            "ignore_e164_validation": false
          },
          "transfer_option": {
            "type": "cold_transfer",
            "show_transferee_as_caller": false
          }
        }
      ]
    },
    {
      "name": "appointment_booking",
      "state_prompt": "You will follow the steps below to book an appointment...",
      "tools": [
        {
          "type": "book_appointment_cal",
          "name": "book_appointment",
          "description": "Book an annual check up.",
          "cal_api_key": "cal_live_xxxxxxxxxxxx",
          "event_type_id": 60444,
          "timezone": "America/Los_Angeles"
        }
      ]
    }
  ],
  "starting_state": "information_collection"
}
When to Use Multi-Prompt: [(1) ]() - Agents with multiple decision branches (more than 3-4 conditional paths) - Using 5+ different functions/tools - Need to track multiple variables throughout the conversation - Single prompt shows inconsistent behavior Multi-prompt agents reduce hallucination by keeping prompts shorter and limiting tool choices per state, helping the LLM better focus and follow rules.
Feel free to tag @AI Evy with additional questions. I also appreciate feedback, it helps me improve.
r
hey about the sip trunk how can I do this and reply to it? what the telephony layer company said: 1. The range of IPs shared is very broad, over 10k+ IPs which isn't good for security reasons. Please share specifci IPs and this is the IP I shared 18.98.16.120/30 143.223.88.0/21 161.115.160.0/19 2. For SIP trunk credentials, please generate credentials from their side and share with us we will configure on our end
@AI Evy hey about the sip trunk how can I do this and reply to it? what the telephony layer company said: 1. The range of IPs shared is very broad, over 10k+ IPs which isn't good for security reasons. Please share specifci IPs and this is the IP I shared 18.98.16.120/30 143.223.88.0/21 161.115.160.0/19 2. For SIP trunk credentials, please generate credentials from their side and share with us we will configure on our end
a
Thanks for the follow-up, @Rowan. I'll get back to you soon.
### Answer ### Response to Your Telephony Provider Regarding IP Ranges: The IP blocks you shared are the official Retell IP ranges for SIP traffic. [(1) ]() These are the only IP ranges Retell provides: -
18.98.16.120/30
(All regions) - This is a /30 subnet = 4 IP addresses -
143.223.88.0/21
(certain United States traffic) - This is a /21 subnet = 2,048 IP addresses -
161.115.160.0/19
(certain United States traffic) - This is a /19 subnet = 8,192 IP addresses You can respond to your provider: "These are the official IP blocks provided by Retell AI for SIP signaling. The smallest block (
18.98.16.120/30
) contains only 4 IPs and covers all regions. The larger blocks are specifically for US traffic routing. Retell does not provide more specific/narrower IP ranges than these official blocks. If your security policy requires narrower ranges, you may want to use credential-based authentication instead of IP whitelisting." Regarding SIP Trunk Credentials: Your telephony provider has this backwards. For elastic SIP trunking with Retell, you (the customer) create the credentials on your telephony provider's side, not Retell's side. [(2) ]() [(1) ]() You can respond: "For elastic SIP trunking integration, the authentication flow works as follows: I create SIP trunk credentials (username/password) within your platform, then I provide those credentials to Retell when importing my phone number. Retell will use these credentials to authenticate outbound calls to your SIP trunk. You do not generate credentials for Retell - I generate them on your platform and share them with Retell." Setup Steps: 1. Create credentials (username/password) in your telephony provider's SIP trunk configuration [(2) ](ht
tps://docs.retellai.com/deploy/twilio) 2. Note the termination SIP URI from your provider [(1) ]() 3. Import your number into Retell, providing the termination URI and the credentials you created
​ To learn more, see the sources I considered: [[1] Custom Telephony Overview](https://docs.retellai.com/deploy/custom-telephony) [[2] Twilio](https://docs.retellai.com/deploy/twilio) ​
Feel free to tag @AI Evy with additional questions. I also appreciate feedback, it helps me improve.
r
wait so we can use sip trunk via credintials without the IP Whitelisting?
@AI Evy wait so we can use sip trunk via credintials without the IP Whitelisting?
a
Thanks for the follow-up, @Rowan. I'll get back to you soon.
### Answer Yes, you can use credential-based authentication OR IP whitelisting - you don't necessarily need both. [(1) ]() For elastic SIP trunking with Retell, the documentation states: "For your elastic SIP trunk to accept our outbound request, you need to whitelist IP address or create a auth with username and password." Two authentication options: 1. **Credential-based authentication**: Create username/password credentials in your telephony provider's SIP trunk settings, then provide those credentials when importing your number into Retell 2. **IP whitelisting**: Whitelist Retell's CIDR blocks (
18.98.16.120/30
,
143.223.88.0/21
,
161.115.160.0/19
) in your telephony provider's settings Which should you choose? Since your provider has security concerns about the IP ranges, credential-based authentication is the better option for you. This approach doesn't require IP whitelisting and provides secure authentication through username/password credentials you create on your telephony provider's platform. However, some community discussions suggest using both methods together for added reliability, though the official documentation presents them as alternatives. [(2) ]() [(3) ]() Setup with credentials only: 1. Create SIP trunk credentials (username/password) in your telephony provider's dashboard 2. Note your termination SIP URI from your provider 3. Import your number into Retell, providing the termination URI and the credentials you created [(1) ]() [(4) ]()
​ To learn more, see the sources I considered: [[1] Twilio](https://docs.retellai.com/deploy/twilio) [[2] Discord message from @dumpfile on 2025-05-23](https://discord.com/channels/1186424376300679209/1186462578310910112/1375548017675075795) [[3] Discord message from @naseem_53863 on 2025-05-23](https://discord.com/channels/1186424376300679209/1186462578310910112/1375547580527677492) [[4] How to Choose the Right Telephony System in 2024 | Retell AI](https://www.retellai.com/blog/how-to-choose-the-right-telephony-system-in-2024) ​
Feel free to tag @AI Evy with additional questions. I also appreciate feedback, it helps me improve.
r
hey so can we use IP whitelisting only instead of credentials?
@AI Evy hey so can we use IP whitelisting only instead of credentials?
a
Thanks for the follow-up, @Rowan. I'll get back to you soon.
### Answer Yes, you can use IP whitelisting only without credentials. [(1) ]() The documentation states: "For your elastic SIP trunk to accept our outbound request, you need to whitelist IP address or create a auth with username and password." Using IP whitelisting only: Whitelist Retell's CIDR blocks in your telephony provider's SIP trunk access control settings: [(1) ]() [(2) ]() -
18.98.16.120/30
(All regions) -
143.223.88.0/21
(certain United States traffic) -
161.115.160.0/19
(certain United States traffic) When you import your number into Retell, you'll provide the termination SIP URI but can leave the username/password fields blank since authentication will be handled via IP whitelisting. [(1) ]() However, given your provider's security concerns about the broad IP ranges, credential-based authentication might be more acceptable to them. The documentation presents these as alternative authentication methods - you can choose the one that works best for your security requirements.
​ To learn more, see the sources I considered: [[1] Twilio](https://docs.retellai.com/deploy/twilio) [[2] Custom Telephony Overview](https://docs.retellai.com/deploy/custom-telephony) ​
Feel free to tag @AI Evy with additional questions. I also appreciate feedback, it helps me improve.
r
I'm using ZIWO not twilio
@AI Evy I'm using ZIWO not twilio
a
Thanks for the follow-up, @Rowan. I'll get back to you soon.
### Answer The authentication options I mentioned apply to any telephony provider that supports elastic SIP trunking, including Ziwo. [(1) ]() The Custom Telephony Overview documentation states that for elastic SIP trunking integration, "you will need to set up a SIP trunking and configure your number to point to it, and then import that number to Retell." The authentication methods (IP whitelisting OR credentials) are standard for SIP trunk integrations regardless of provider. For Ziwo specifically: You can use IP whitelisting only by having Ziwo whitelist Retell's CIDR blocks: -
18.98.16.120/30
(All regions) -
143.223.88.0/21
(certain United States traffic) -
161.115.160.0/19
(certain United States traffic) When importing your number into Retell, you'll provide Ziwo's termination SIP URI but can leave username/password blank since authentication is handled via IP whitelisting. The setup process is the same as documented for other providers like Twilio, Telnyx, and Vonage - these guides serve as references for any telephony provider supporting SIP trunks. **
​ To learn more, see the sources I considered: [[1] Custom Telephony Overview](https://docs.retellai.com/deploy/custom-telephony) ​
Feel free to tag @AI Evy with additional questions. I also appreciate feedback, it helps me improve.
r
@AI Evy I can't find narrower ip address?
a
Thanks for the follow-up, @Rowan. I'll get back to you soon.
### Answer Unfortunately, the IP blocks you've shared are the only official IP ranges Retell provides for SIP traffic. [(1) ]() Retell does not offer more specific or narrower IP addresses than these three CIDR blocks: -
18.98.16.120/30
(All regions) - 4 IP addresses -
143.223.88.0/21
(certain United States traffic) -
161.115.160.0/19
(certain United States traffic) Your options with Ziwo: Since Ziwo has security concerns about the broad IP ranges and Retell doesn't provide narrower ranges, credential-based authentication is your best solution. This eliminates the need for IP whitelisting entirely. Setup with credentials: 1. Create SIP trunk credentials (username/password) in Ziwo's dashboard 2. Get Ziwo's termination SIP URI 3. Import your number into Retell, providing the termination URI and credentials you created This approach addresses Ziwo's security requirements while maintaining full SIP trunk functionality with Retell.
​ To learn more, see the sources I considered: [[1] Custom Telephony Overview](https://docs.retellai.com/deploy/custom-telephony) ​
Feel free to tag @AI Evy with additional questions. I also appreciate feedback, it helps me improve.
r
@AI Evy since it's 4 ip means it's narrow?
a
Thanks for the follow-up, @Rowan. I'll get back to you soon.
### Answer Yes,
18.98.16.120/30
is narrow - it contains only 4 IP addresses. [(1) ]() This is the smallest and most specific CIDR block Retell provides. For Ziwo, you have two options: 1. **Use only the narrow block**: Since you're UAE-based, whitelist only
18.98.16.120/30
(4 IPs, covers all regions) . The documentation notes that "some non US traffic might go from IPs not listed above, but will be migrated to the first IP block by 12/03/2025". 2. **Use credential-based authentication instead**: This completely avoids the IP whitelisting concern and may be more acceptable to Ziwo's security team. Recommendation: Start with just
18.98.16.120/30
for IP whitelisting. If Ziwo still considers 4 IPs too broad for their security policy, switch to credential-based authentication.
​ To learn more, see the sources I considered: [[1] Custom Telephony Overview](https://docs.retellai.com/deploy/custom-telephony) ​
Feel free to tag @AI Evy with additional questions. I also appreciate feedback, it helps me improve.
r
@AI Evy what do you mean by all regions?
a
Thanks for the follow-up, @Rowan. I'll get back to you soon.
### Answer "All regions" means this IP block handles SIP traffic globally for all geographic regions where Retell operates. [(1) ]() The
18.98.16.120/30
block is used for SIP signaling regardless of where your calls originate or terminate - whether you're in the UAE, United States, Europe, or anywhere else. The other two CIDR blocks (
143.223.88.0/21
and
161.115.160.0/19
) are specifically labeled as "certain United States traffic", meaning they only handle some US-based call routing. [(1) ]() [(2) ]() [(3) ]() Since you're using Ziwo from the UAE, the
18.98.16.120/30
block (4 IPs) is the primary range you need to whitelist - it covers your region and all other regions globally. [(1) ]()
​ To learn more, see the sources I considered: [[1] Custom Telephony Overview](https://docs.retellai.com/deploy/custom-telephony) [[2] Genesys](https://docs.retellai.com/deploy/genesys) [[3] Avaya](https://docs.retellai.com/deploy/avaya) ​
Feel free to tag @AI Evy with additional questions. I also appreciate feedback, it helps me improve.
r
okay what if I want to do both whitelistig and credientials?
@AI Evy okay what if I want to do both whitelistig and credientials?
a
Thanks for the follow-up, @Rowan. I'll get back to you soon.
### Answer Yes, you can use both IP whitelisting AND credentials together for added security and reliability with your Ziwo SIP trunk. [(1) ]() [(2) ]() Setup with both authentication methods: 1. In Ziwo's SIP trunk settings: - Whitelist Retell's IP block:
18.98.16.120/30
(the 4-IP block covering all regions) [(3) ]() - Create SIP trunk credentials (username/password) in Ziwo's dashboard [(1) ]() - Note Ziwo's termination SIP URI 2. When importing your number into Retell: - Provide Ziwo's termination SIP URI [(1) ]() [(3) ]() - Include the username and password you created in Ziwo [(1) ]() This dual-layer approach provides both network-level security (IP whitelisting) and application-level authentication (credentials) . While the documentation presents these as alternative options ("or") , some community members have found using both methods together improves reliability. [(2) ]() Benefits of using both: - Enhanced security with two authentication layers - Potential fallback if one method encounters issues - May satisfy Ziwo's security requirements while maintaining robust authentication
​ To learn more, see the sources I considered: [[1] Twilio](https://docs.retellai.com/deploy/twilio) [[2] Discord message from @naseem_53863 on 2025-05-23](https://discord.com/channels/1186424376300679209/1186462578310910112/1375547211064279252) [[3] Custom Telephony Overview](https://docs.retellai.com/deploy/custom-telephony) ​
Feel free to tag @AI Evy with additional questions. I also appreciate feedback, it helps me improve.
Thanks for the follow-up, @Rowan. I'll get back to you soon.
r
@AI Evy if it contains only 4 how many does the other ips contain?
a
### Answer The other two CIDR blocks contain significantly more IP addresses: [(1) ]() [(2) ]() [(3) ]() -
143.223.88.0/21
contains 2,048 IP addresses (certain United States traffic) -
161.115.160.0/19
contains 8,192 IP addresses (certain United States traffic) In comparison,
18.98.16.120/30
contains only 4 IP addresses and covers all regions. Since you're in the UAE and Ziwo has security concerns about broad IP ranges, you should whitelist only the
18.98.16.120/30
block (4 IPs) as it covers your region globally. The larger US-specific blocks are unnecessary for your use case. [(1) ]()
​ To learn more, see the sources I considered: [[1] Custom Telephony Overview](https://docs.retellai.com/deploy/custom-telephony) [[2] Genesys](https://docs.retellai.com/deploy/genesys) [[3] Avaya](https://docs.retellai.com/deploy/avaya) ​
Feel free to tag @AI Evy with additional questions. I also appreciate feedback, it helps me improve.
r
@AI Evy if we proceed with whitelisting only we will still need the termniation URI from ziwo?
a
Thanks for the follow-up, @Rowan. I'll get back to you soon.
### Answer Yes, you absolutely need Ziwo's termination SIP URI even if you're using IP whitelisting only. [(1) ]() [(2) ]() Required information when importing your number to Retell: When you import your Ziwo number into Retell, you must provide the termination SIP URI regardless of which authentication method you choose. The termination SIP URI tells Retell where to route outbound calls. [(1) ]() Setup with IP whitelisting only: 1. Have Ziwo whitelist Retell's IP block:
18.98.16.120/30
in their SIP trunk access control settings [(1) ]() [(2) ]() 2. Get Ziwo's termination SIP URI from their platform 3. Import your number into Retell, providing Ziwo's termination SIP URI 4. Leave the username/password fields blank since authentication is handled via IP whitelisting [(2) ]() The termination SIP URI is essential for the integration - it's the destination address Retell uses to send outbound calls through Ziwo's network. [(1) ]() [(2) ]() [(3) ]()
​ To learn more, see the sources I considered: [[1] Custom Telephony Overview](https://docs.retellai.com/deploy/custom-telephony) [[2] Twilio](https://docs.retellai.com/deploy/twilio) [[3] How to Choose the Right Telephony System in 2024 | Retell AI](https://www.retellai.com/blog/how-to-choose-the-right-telephony-system-in-2024) ​
Feel free to tag @AI Evy with additional questions. I also appreciate feedback, it helps me improve.
r
@AI Evy that's ZIWO's reply what should I do now? Hello Rowan, Regarding the credentials, since ZIWO is sending calls to Retell, Retell is the receiving/authenticating side. This means the credentials should be generated on your Retell dashboard and shared with us to configure on our end. Could you please share the SIP Username, Password, and Termination URI from your Retell dashboard? Thank you, ZIWO Support
a
Thanks for the follow-up, @Rowan. I'll get back to you soon.
### Answer Ziwo's understanding is incorrect. For elastic SIP trunking with Retell, *you create credentials on Ziwo's side (not Retell's side) * and then provide those credentials to Retell when importing your number. [(1) ]() [(2) ]() Here's what you need to tell Ziwo: For inbound calls (Ziwo → Retell) : - Ziwo should configure their SIP trunk to send calls to Retell's SIP server:
sip:sip.retellai.com
- No credentials are needed for inbound - Ziwo just routes calls to this URI [(1) ]() For outbound calls (Retell → Ziwo) : - Ziwo needs to create SIP trunk credentials (username/password) in their dashboard [(1) ]() [(2) ]() - Ziwo provides you with their termination SIP URI (e.g., something like
yourdevice.sip.ziwo.com
) [(3) ]() - You then import your number into Retell and provide Ziwo's termination URI along with the credentials Ziwo gave you [(1) ]() [(4) ]() What Ziwo needs from you: - Retell's SIP server address for origination (inbound) :
sip:sip.retellai.com
[(1) ]() [(2) ]() [(5) ]() - Retell's IP block to whitelist (if using IP authentication) :
18.98.16.120/30
[(1) ]() What you need from Ziwo: - Their termination SIP URI for outbound calls [(1) ]() [(2) ]() - SIP trunk cred
entials (username/password) they create for you The authentication flow is: Retell authenticates to Ziwo's trunk using credentials Ziwo creates, not the other way around. [(3) ]()
Feel free to tag @AI Evy with additional questions. I also appreciate feedback, it helps me improve.
​ To learn more, see the sources I considered: [[1] Twilio](https://docs.retellai.com/deploy/twilio) [[2] Telnyx](https://docs.retellai.com/deploy/telnyx) [[3] Discord message from @fyodor2572 on 2025-02-10](https://discord.com/channels/1186424376300679209/1186462578310910112/1338632610464927774) [[4] Import Phone Number](https://docs.retellai.com/api-references/import-phone-number) [[5] How to Choose the Right Telephony System in 2024 | Retell AI](https://www.retellai.com/blog/how-to-choose-the-right-telephony-system-in-2024) ​
r
@AI Evy can I even create credentials on retell side?
a
Thanks for the follow-up, @Rowan. I'll get back to you soon.
### Answer No, you cannot create SIP trunk credentials on Retell's side. Retell does not generate credentials for your telephony provider to use. [(1) ]() [(2) ]() How authentication works with Retell: When you import a number into Retell, you provide credentials that your telephony provider (Ziwo) creates. Retell uses these credentials to authenticate outbound calls to Ziwo's SIP trunk. [(1) ]() [(3) ]() What you provide when importing: When importing your number via the API or dashboard, you supply: [(3) ]() -
termination_uri
- Ziwo's SIP URI for outbound calls -
sip_trunk_auth_username
- Username created by Ziwo -
sip_trunk_auth_password
- Password created by Ziwo The correct flow: 1. Ziwo creates SIP trunk credentials (username/password) in their system 2. Ziwo provides you with their termination URI and credentials 3. You import your number into Retell using Ziwo's information 4. Retell authenticates to Ziwo using those credentials for outbound calls [(1) ]() [(2) ]() For inbound calls, Ziwo simply routes calls to Retell's SIP server (
sip:sip.retellai.com
) without needing credentials from Retell.
​ To learn more, see the sources I considered: [[1] Twilio](https://docs.retellai.com/deploy/twilio) [[2] Custom Telephony Overview](https://docs.retellai.com/deploy/custom-telephony) [[3] Import Phone Number](https://docs.retellai.com/api-references/import-phone-number) ​
Feel free to tag @AI Evy with additional questions. I also appreciate feedback, it helps me improve.
r
what if it would be for inbound calls only?
@AI Evy what if it would be for inbound calls only?
a
Thanks for the follow-up, @Rowan. I'll get back to you soon.
### Answer For inbound-only calls, the setup is simpler and you don't need credentials at all. [(1) ]() What Ziwo needs to configure for inbound calls: 1. Set Retell's SIP server as the origination URI:
sip:sip.retellai.com
2. Whitelist Retell's IP block:
18.98.16.120/30
[(2) ]() [(1) ]() What you need to do in Retell: When importing your number, you still need to provide Ziwo's termination URI (even for inbound-only) , but you can leave the username and password fields blank since you won't be making outbound calls. The key difference for inbound-only: - No credentials needed from Ziwo [(1) ]() - No termination authentication required [(2) ]() - Ziwo simply routes incoming calls to
sip:sip.retellai.com
[(1) ]() - Retell identifies which agent to use based on the phone number that received the call This is the simplest setup - Ziwo just needs to point their SIP trunk's origination to Retell's SIP server and whitelist the IP block.
​ To learn more, see the sources I considered: [[1] Custom Telephony Overview](https://docs.retellai.com/deploy/custom-telephony) [[2] Twilio](https://docs.retellai.com/deploy/twilio) ​
Feel free to tag @AI Evy with additional questions. I also appreciate feedback, it helps me improve.
r
@AI Evy is this wrong? Authentication: Retell mentioned that using both IP whitelisting and credentials is ideal, but we can proceed with just whitelisting if needed, especially with the smaller IP block
a
Thanks for the follow-up, @Rowan. I'll get back to you soon.
### Answer That statement is partially misleading. The official Retell documentation presents IP whitelisting and credentials as alternative authentication methods, not as a combined "ideal" approach. [(1) ]() [(2) ]() What the documentation actually says: For elastic SIP trunking, your telephony provider needs to accept outbound requests from Retell. You can achieve this by: [(1) ]() 1. IP whitelisting - Whitelist Retell's CIDR block
18.98.16.120/30
[(1) ]() [(2) ]() 2. Credential-based authentication - Create username and password on your provider's side The accurate statement should be: "Retell supports either IP whitelisting OR credential-based authentication. For your setup with Ziwo, you can choose just IP whitelisting (using the
18.98.16.120/30
block) if that meets your security requirements." While some community members have reported using both methods together for added reliability, the official documentation does not describe combining them as "ideal" - it presents them as options. For inbound-only calls specifically, you only need Ziwo to route calls to
sip:sip.retellai.com
with IP whitelisting - no credentials required at all.
​ To learn more, see the sources I considered: [[1] Twilio](https://docs.retellai.com/deploy/twilio) [[2] Custom Telephony Overview](https://docs.retellai.com/deploy/custom-telephony) ​
Feel free to tag @AI Evy with additional questions. I also appreciate feedback, it helps me improve.
r
can we do outbound without credentials?
@AI Evy can we do outbound without credentials?
a
Thanks for the follow-up, @Rowan. I'll get back to you soon.
### Answer Yes, you can do outbound calls without credentials by using IP whitelisting instead. [(1) ]() [(2) ]() For outbound calls with IP whitelisting only: Ziwo needs to whitelist Retell's CIDR block
18.98.16.120/30
in their SIP trunk access control settings. This allows Retell to authenticate based on the source IP address rather than username/password credentials. [(1) ]() What you need to provide when importing your number: When importing your Ziwo number into Retell, you must provide: [(1) ]() [(3) ]() -
termination_uri
- Ziwo's termination SIP URI for routing outbound calls [(1) ]() [(2) ]() - Leave
sip_trunk_auth_username
and
sip_trunk_auth_password
blank (authentication handled via IP) [(1) ]() [(3) ]() The authentication choice: The documentation states: "For your elastic SIP trunk to accept our outbound request, you need to whitelist IP address or create a auth with username and password." [(1) ]() This means you can choose either IP whitelisting OR credentials - both methods work for outbound calls. [(1) ]() [(2) ]() Since you're using the narrow
18.98.16.120/30
block (only 4 IPs) , IP whitelisting alone should be acceptable to Ziwo's security requirements for outbound calling. [(2) ]()
​ To learn more, see the sources I considered: [[1] Twilio](https://docs.retellai.com/deploy/twilio) [[2] Custom Telephony Overview](https://docs.retellai.com/deploy/custom-telephony) [[3] Import Phone Number](https://docs.retellai.com/api-references/import-phone-number) ​
Feel free to tag @AI Evy with additional questions. I also appreciate feedback, it helps me improve.
r
@AI Evy is this wrong then: From my understanding, Retell does not generate SIP credentials, and the typical flow is: • Inbound (provider → Retell): calls are routed to sip:sip.retellai.com with IP whitelisting, without credentials • Outbound (Retell → provider): the provider generates SIP trunk credentials, which are then used in Retell
a
Thanks for the follow-up, @Rowan. I'll get back to you soon.
### Answer Your understanding is mostly correct, but needs one clarification about outbound authentication options. [(1) ]() [(2) ]() What's correct: - Retell does not generate SIP credentials - this is accurate - Inbound flow: calls route to
sip:sip.retellai.com
- correct What needs clarification: For outbound calls, you have two authentication options, not just credentials: 1. **Credential-based**: Provider generates SIP trunk credentials (username/password) , which you provide to Retell when importing the number [(1) ]() [(3) ]() 2. **IP whitelisting**: Provider whitelists Retell's CIDR block
18.98.16.120/30
instead of using credentials [(1) ]() [(2) ]() The accurate statement should be: - Inbound (provider → Retell) : calls route to
sip:sip.retellai.com
with IP whitelisting, no credentials needed - Outbound (Retell → provider) : the provider either (a) generates SIP trunk credentials for Retell to use, OR (b) whitelists Retell's IP block
18.98.16.120/30
- both methods work For elastic SIP trunking, your telephony provider needs to accept Retell's outbound requests by either whitelisting the IP address or creating authentication with username and password. [(1) ]() When importing your number, you provide the termination URI and optionally the credentials if using that authentication method. [(1) ]() [(3) ]()
​ To learn more, see the sources I considered: [[1] Twilio](https://docs.retellai.com/deploy/twilio) [[2] Custom Telephony Overview](https://docs.retellai.com/deploy/custom-telephony) [[3] Import Phone Number](https://docs.retellai.com/api-references/import-phone-number) ​
Feel free to tag @AI Evy with additional questions. I also appreciate feedback, it helps me improve.
r
@AI Evy wait so can you use credentials for inbound?
a
Thanks for the follow-up, @Rowan. I'll get back to you soon.
### Answer No, you cannot use credentials for inbound calls. For inbound calls with elastic SIP trunking, your telephony provider (Ziwo) simply sets up origination to point to Retell's SIP server address:
sip:sip.retellai.com
. [(1) ]() [(2) ]() For inbound calls: - Ziwo configures their SIP trunk origination URI to
sip:sip.retellai.com
[(1) ]() - No authentication credentials are needed from either side [(1) ]() [(2) ]() - Ziwo may optionally whitelist Retell's IP block
18.98.16.120/30
, but this is for network-level security, not credential-based authentication Why credentials aren't used for inbound: The authentication model for elastic SIP trunking is one-directional. Credentials only apply to outbound calls where Retell needs to authenticate to Ziwo's termination endpoint. [(1) ]() For inbound, Ziwo is initiating the connection to Retell's publicly accessible SIP server, so no authentication from Retell is required. [(2) ]() The complete authentication picture: - **Inbound**: Ziwo →
sip:sip.retellai.com
(no credentials needed) [(1) ]() [(2) ]() - **Outbound**: Retell → Ziwo's termination URI (either IP whitelisting OR credentials required) [(1) ]()
​ To learn more, see the sources I considered: [[1] Twilio](https://docs.retellai.com/deploy/twilio) [[2] Custom Telephony Overview](https://docs.retellai.com/deploy/custom-telephony) ​
Feel free to tag @AI Evy with additional questions. I also appreciate feedback, it helps me improve.
r
@AI Evy retell doesn't provide termination URI right?
a
Thanks for the follow-up, @Rowan. I'll get back to you soon.
### Answer Correct - Retell does not provide a termination URI. The termination URI comes from *your telephony provider (Ziwo) * and you provide it to Retell when importing your number. [(1) ]() [(2) ]() What Retell provides: - SIP server address for inbound calls (origination) :
sip:sip.retellai.com
- IP blocks to whitelist:
18.98.16.120/30
[(2) ]() What your telephony provider (Ziwo) provides: - Termination SIP URI for outbound calls (e.g., something like
yourdevice.sip.ziwo.com
) [(1) ]() [(3) ]() - Optionally, SIP trunk credentials (username/password) if not using IP whitelisting [(1) ]() When you import your Ziwo number into Retell, you must supply Ziwo's termination URI in the
termination_uri
field. [(4) ]() This tells Retell where to route outbound calls through Ziwo's network. [(1) ]() [(2) ]()
​ To learn more, see the sources I considered: [[1] Twilio](https://docs.retellai.com/deploy/twilio) [[2] Custom Telephony Overview](https://docs.retellai.com/deploy/custom-telephony) [[3] Telnyx](https://docs.retellai.com/deploy/telnyx) [[4] Import Phone Number](https://docs.retellai.com/api-references/import-phone-number) ​
Feel free to tag @AI Evy with additional questions. I also appreciate feedback, it helps me improve.
r
@AI Evy so if we don't want to use sip credentials what is the best way to do that? for inbound only? and for both?
a
Thanks for the follow-up, @Rowan. I'll get back to you soon.
### Answer --- For inbound-only (no credentials) : [(1) ]() Ziwo needs to configure their SIP trunk origination to point to Retell's SIP server:
sip:sip.retellai.com
. When importing your number into Retell, you still need to provide Ziwo's termination URI (even though you won't use it for outbound) , but leave the username and password fields blank . For both inbound and outbound (no credentials) : Use IP whitelisting instead of credentials. Here's the setup: 1. Ziwo configures inbound (origination) : - Set origination URI to
sip:sip.retellai.com
2. Ziwo configures outbound (termination) : - Whitelist Retell's CIDR block:
18.98.16.120/30
- Provide you with their termination SIP URI 3. You import the number into Retell: [(1) ]() [(2) ]() - Provide Ziwo's termination URI in the
termination_uri
field [(2) ]() - Leave
sip_trunk_auth_username
and
sip_trunk_auth_password
blank The documentation states: "For your elastic SIP trunk to accept our outbound request, you need to whitelist IP address or create a auth with username and password." [(1) ]() This means IP whitelisting alone is sufficient for outbound calls without needing credentials .
​ To learn more, see the sources I considered: [[1] Twilio](https://docs.retellai.com/deploy/twilio) [[2] Import Phone Number](https://docs.retellai.com/api-references/import-phone-number) ​
Feel free to tag @AI Evy with additional questions. I also appreciate feedback, it helps me improve.
r
@AI Evy we will need termination uri from them anyway right?
a
Thanks for the follow-up, @Rowan. I'll get back to you soon.
### Answer Yes, you need Ziwo's termination URI regardless of whether you're doing inbound-only or both inbound and outbound calls. [(1) ]() [(2) ]() [(3) ]() Why you need the termination URI: When you import a phone number into Retell, the
termination_uri
field is required in the API call. [(3) ]() This is Ziwo's SIP endpoint that Retell uses to route outbound calls through their network. [(1) ]() [(2) ]() For inbound-only: Even if you only plan to receive inbound calls, you still need to provide Ziwo's termination URI when importing the number into Retell. You just won't actually use it for making outbound calls. [(1) ]() For both inbound and outbound: The termination URI is essential - it tells Retell where to send outbound calls. For example, with Twilio it typically ends with
.pstn.twilio.com
. Ziwo should provide you with their equivalent termination SIP URI. What to ask Ziwo for: Request their termination SIP URI (the endpoint Retell will use to route outbound calls through Ziwo's network) . [(1) ]() [(2) ]()
​ To learn more, see the sources I considered: [[1] Twilio](https://docs.retellai.com/deploy/twilio) [[2] Custom Telephony Overview](https://docs.retellai.com/deploy/custom-telephony) [[3] Import Phone Number](https://docs.retellai.com/api-references/import-phone-number) ​
Feel free to tag @AI Evy with additional questions. I also appreciate feedback, it helps me improve.
r
@AI Evy is this correct Yes, you can use both IP whitelisting AND credentials together for added security and reliability with your Ziwo SIP trunk. (1) (2)
a
Thanks for the follow-up, @Rowan. I'll get back to you soon.
### Answer This statement is partially misleading based on the official documentation. [(1) ]() [(2) ]() The official Retell documentation presents IP whitelisting and credentials as alternative authentication methods ("or") , not as a combined approach. [(1) ]() The documentation states: "For your elastic SIP trunk to accept our outbound request, you need to whitelist IP address or create a auth with username and password." What the documentation actually says: For outbound authentication with elastic SIP trunking, you can choose: [(1) ]() [(2) ]() - **IP whitelisting**: Whitelist Retell's CIDR block
18.98.16.120/30
- **Credential-based authentication**: Create username and password [(1) ]() The documentation does not describe using both methods together as a recommended or "added security" approach. [(1) ]() [(2) ]() While it's technically possible that some telephony providers might accept both configurations simultaneously, the official Retell documentation presents these as either/or options, not as a combined security strategy. For your Ziwo setup, you should choose one method: - Use IP whitelisting alone (whitelist
18.98.16.120/30
) - OR use credentials alone (username/password from Ziwo) [(1) ]()
​ To learn more, see the sources I considered: [[1] Twilio](https://docs.retellai.com/deploy/twilio) [[2] Custom Telephony Overview](https://docs.retellai.com/deploy/custom-telephony) ​
Feel free to tag @AI Evy with additional questions. I also appreciate feedback, it helps me improve.
r
@AI Evy but this was your statement?
a
Thanks for the follow-up, @Rowan. I'll get back to you soon.
### Answer I apologize for the confusion in my previous responses. You're correct to call this out. After reviewing the official documentation more carefully, the sources consistently present IP whitelisting and credentials as alternative authentication methods using "or", not as a combined approach. [(1) ]() [(2) ]() The Twilio documentation states: "For your elastic SIP trunk to accept our outbound request, you need to whitelist IP address or create a auth with username and password." [(1) ]() The Custom Telephony Overview similarly describes these as two separate options, not a combined security strategy. [(2) ]() The accurate statement should be: For outbound authentication with Ziwo's elastic SIP trunking, choose one method: [(1) ]() [(2) ]() - **IP whitelisting**: Whitelist Retell's CIDR block
18.98.16.120/30
- **Credential-based authentication**: Use username and password from Ziwo [(1) ]() While some telephony providers might technically accept both configurations, the official Retell documentation does not recommend or describe using both together as an "added security" approach. [(1) ]() [(2) ]() I apologize for the inconsistency in my earlier responses. Thank you for catching this.
​ To learn more, see the sources I considered: [[1] Twilio](https://docs.retellai.com/deploy/twilio) [[2] Custom Telephony Overview](https://docs.retellai.com/deploy/custom-telephony) ​
Feel free to tag @AI Evy with additional questions. I also appreciate feedback, it helps me improve.
r
@AI Evy are all of these correct: SIP credentials are not generated on Retell’s side; they are provider-generated only. Retell only allows entering provider-supplied SIP credentials and does not generate them. Retell does not provide a termination URI; it must be provided by the telephony provider (ZIWO). Inbound calls (ZIWO → Retell) are routed to sip:sip.retellai.com. Inbound authentication is handled only via IP whitelisting (no SIP credentials used). Outbound calls (Retell → ZIWO) require provider-side authentication (either SIP credentials or IP whitelisting). Retell supports IP whitelisting as a sufficient authentication method for outbound calls. The current POC scope may be inbound only (uncertainty about whether outbound is included or not). Retell dashboard does not include an option to generate SIP credentials or configure them internally.
a
Thanks for the follow-up, @Rowan. I'll get back to you soon.
### Answer Yes, all of your statements are correct based on the official documentation. Let me confirm each one: 1. SIP credentials are not generated on Retell's side; they are provider-generated only. ✅ Correct. Retell does not generate SIP credentials. [(1) ]() 2. Retell only allows entering provider-supplied SIP credentials and does not generate them. ✅ Correct. When importing a number, you provide the
sip_trunk_auth_username
and
sip_trunk_auth_password
that come from your telephony provider. [(2) ]() 3. Retell does not provide a termination URI; it must be provided by the telephony provider (ZIWO) . ✅ Correct. The
termination_uri
field is required when importing a number and must come from your provider (e.g., for Twilio it ends with
.pstn.twilio.com
) . [(1) ]() [(2) ]() 4. Inbound calls (ZIWO → Retell) are routed to sip:sip.retellai.com. ✅ Correct. For origination (inbound) , you specify Retell's SIP server address as
sip:sip.retellai.com
. [(1) ]() [(3) ]() [(4) ]() 5. Inbound authentication is handled only via IP whitelisting (no SIP credentials used) . ✅ Correct. For inbound calls, no credentials are needed from either side. [(1) ]() 6. Outbound calls (Retell → ZIWO) require provider-side authentication (either SIP credentials or IP whitelisting) . ✅ Correct. The documentation states: "For your elastic SIP trunk to accept our outbound request, you need to whitelist IP address or create a auth with username and password." **7. Retell supports IP whitelisting as a su
fficient authentication method for outbound calls.** ✅ Correct. You can whitelist Retell's CIDR block
18.98.16.120/30
instead of using credentials. 8. The current POC scope may be inbound only (uncertainty about whether outbound is included or not) . ✅ This is a reasonable statement about your specific project scope - not something the documentation would confirm or deny. 9. Retell dashboard does not include an option to generate SIP credentials or configure them internally. ✅ Correct. The dashboard allows you to enter provider-supplied credentials when importing a number, but does not generate them. [(1) ]() [(2) ]() All nine statements accurately reflect how Retell's elastic SIP trunking integration works.
​ To learn more, see the sources I considered: [[1] Twilio](https://docs.retellai.com/deploy/twilio) [[2] Import Phone Number](https://docs.retellai.com/api-references/import-phone-number) [[3] Telnyx](https://docs.retellai.com/deploy/telnyx) [[4] Vonage](https://docs.retellai.com/deploy/vonage) ​
Feel free to tag @AI Evy with additional questions. I also appreciate feedback, it helps me improve.
r
@AI Evy can I set a custom telephony layer for inbound calls only without getting the termination URI?
a
Thanks for the follow-up, @Rowan. I'll get back to you soon.
### Answer No, you cannot import a phone number into Retell without providing the termination URI. [(1) ]() When you import a phone number using the Import Phone Number API, the
termination_uri
field is required. The API documentation shows:
Copy code
json
{
  "phone_number": "+14157774444",
  "termination_uri": "someuri.pstn.twilio.com"
}
The
termination_uri
is described as "The termination uri to uniquely identify your elastic SIP trunk. This is used for outbound calls." Why the termination URI is required: Even if you only plan to handle inbound calls, Retell's system architecture requires the termination URI when importing a number through elastic SIP trunking. [(2) ]() [(1) ]() This is a required field in the import process, regardless of whether you intend to use outbound functionality. [(1) ]() Alternative approach if you truly need inbound-only: If you have a more complex telephony setup that cannot use elastic SIP trunking, you could use Method 2: Dial to SIP URI instead. [(2) ]() With this method, you would call the Register Phone Call API to obtain a
call_id
, then dial to
sip:{call_id}@sip.retellai.com
. [(2) ]() [(3) ]() However, this requires custom code integration with your telephony provider and is not the recommended approach. [(2) ]()
​ To learn more, see the sources I considered: [[1] Import Phone Number](https://docs.retellai.com/api-references/import-phone-number) [[2] Custom Telephony Overview](https://docs.retellai.com/deploy/custom-telephony) [[3] Register Phone Call](https://docs.retellai.com/api-references/register-phone-call) ​
Feel free to tag @AI Evy with additional questions. I also appreciate feedback, it helps me improve.
r
@AI Evy How to setup sip on retell I have termination uri, DID, and retell IP is already whitelisted what do I need to do now?
a
Thanks for the follow-up, @Rowan. I'll get back to you soon.
### Answer Since you have your termination URI, DID (phone number) , and Retell's IP already whitelisted on Ziwo's side, you're ready to import the number into Retell. [(1) ]() [(2) ]() Next steps to complete your SIP setup: 1. Import your number into Retell You can import the number either through the Retell dashboard or programmatically via the Import Number API. [(1) ]() [(3) ]() Via Dashboard: - Go to Phone Numbers > Import Number in your Retell dashboard [(1) ]() - Fill in the following fields: [(1) ]() [(3) ]() - **Phone Number**: Your DID in E.164 format (e.g.,
+14157774444
) [(3) ]() - **Termination SIP URI**: Ziwo's termination URI that you received [(1) ]() [(2) ]() - **SIP Username / Password**: Leave blank since you're using IP whitelisting for authentication [(1) ]() Via API:
Copy code
javascript
import Retell from 'retell-sdk';

const client = new Retell({
  apiKey: process.env['RETELL_API_KEY'],
}) ;

const phoneNumberResponse = await client.phoneNumber.import({
  phone_number: '+14157774444',
  termination_uri: 'someuri.pstn.twilio.com',
}) ;

console.log(phoneNumberResponse.inbound_agent_id) ;
​ To learn more, see the sources I considered: [[1] Twilio](https://docs.retellai.com/deploy/twilio) [[2] Custom Telephony Overview](https://docs.retellai.com/deploy/custom-telephony) [[3] Import Phone Number](https://docs.retellai.com/api-references/import-phone-number) ​
[(3) ]() *2. Configure Ziwo's origination (for inbound calls) * Ensure Ziwo has configured their SIP trunk origination to point to Retell's SIP server address:
sip:sip.retellai.com
. [(1) ]() [(2) ]() 3. Assign a Retell agent to the number After importing, bind the number to a specific Retell AI agent in your dashboard so inbound calls are handled by the appropriate agent. [(1) ]() 4. Test the integration - **Inbound**: Place a call to your DID and verify the assigned AI agent responds - **Outbound**: Use the Retell dashboard or Create Phone Call API to initiate outbound calls Once imported, your number will appear in the Retell dashboard and you can make and receive calls just like a Retell-purchased number.
Feel free to tag @AI Evy with additional questions. I also appreciate feedback, it helps me improve.
r
@AI Evy what to do if I want to use the exact same number for multiple ai agents?
a
Thanks for the follow-up, @Rowan. I'll get back to you soon.
### Answer To use the same number for multiple AI agents, you can configure inbound agents with weights when importing or updating your phone number. [(1) ]() How it works: When you set up multiple agents for a single number, Retell randomly selects one agent for each inbound call based on the weights you assign. The weights must add up to 1, representing the probability of each agent being selected. Setup via API: When importing your number, include the
inbound_agents
field:
Copy code
javascript
import Retell from 'retell-sdk';

const client = new Retell({
  apiKey: process.env['RETELL_API_KEY'],
}) ;

const phoneNumberResponse = await client.phoneNumber.import({
  phone_number: '+14157774444',
  termination_uri: 'someuri.pstn.twilio.com',
}) ;

console.log(phoneNumberResponse.inbound_agent_id) ;
The response structure shows how multiple agents are configured:
Copy code
json
{
  "inbound_agents": [
    {
      "agent_id": "oBeDLoLOeuAbiuaMFXRtDOLriTJ5tSxD",
      "weight": 0.5,
      "agent_version": 1
    }
  ]
}
Key requirements: - If
inbound_agents
is set and non-empty, one agent will be picked randomly for each inbound call with probability proportional to the weight - Total weights must add up to 1 - If not set or empty, the system falls back to
inbound_agent_id
Alternative approach using webhooks: You can also use an
inbound_webhook_url
to dynamically override the agent ID for each specific call, allowing you to implement custom routing logic based on caller information or other conditions. [(1) ]() [(2) ]()
​ To learn more, see the sources I considered: [[1] Import Phone Number](https://docs.retellai.com/api-references/import-phone-number) [[2] multiple agent for same inbound number](https://discord.com/channels/1186424376300679209/1422923667821826058) ​
Feel free to tag @AI Evy with additional questions. I also appreciate feedback, it helps me improve.
r
@AI Evy can I test the ingeration without calling the DID?
a
Thanks for the follow-up, @Rowan. I'll get back to you soon.
### Answer --- Yes, you can test your SIP integration without calling the DID using Retell's testing features. [(1) ]() Testing options available: 1. LLM Playground The LLM Playground is an interactive console in the dashboard for prototyping and debugging agent prompts without making live calls. 2. Simulation Testing Retell offers simulation testing to evaluate AI agents in a controlled, virtual environment that mimics real-world scenarios without the risks of live deployment. [(2) ]() You can create user prompts to guide how users would interact with your agent and evaluate the results using defined metrics. Simulation testing is now available for both single and multi-prompt agents, allowing you to write a user prompt to quickly simulate a full conversation instead of manually typing inputs. [(3) ]() 3. Batch Testing Batch testing allows you to test AI agents with larger sets of data or scenarios simultaneously. [(2) ]() Retell has added Batch Simulation Testing so you can run multiple test cases at once with ease. [(3) ]() 4. Test Call from Dashboard You can send a test call directly from the platform and enter dynamic variables when testing—making it easier to simulate real scenarios during testing. The test call will automatically pull in the dynamic variables you've set in the agent configuration. These testing methods allow you to identify and fix issues early, automate testing processes, and reduce costs associated with manual testing before going live with your Ziwo SIP integration. [(2) ]()
​ To learn more, see the sources I considered: [[1] Introduction](https://docs.retellai.com/general/introduction) [[2] How to Deploy Conversational AI | Retell AI](https://www.retellai.com/blog/how-to-deploy-conversational-ai) [[3] Platform Changelogs | Retell AI](https://www.retellai.com/changelog) ​
Feel free to tag @AI Evy with additional questions. I also appreciate feedback, it helps me improve.
Thanks for the follow-up, @Rowan. I'll get back to you soon.
r
@AI Evy can I test the sip ingeration without calling the DID (the number)? to make sure the integration is correct? I don't want to test the agent
a
### Answer Based on the available documentation, there isn't a specific method to test just the SIP integration connectivity without placing an actual call to verify the end-to-end connection between Ziwo and Retell. [(1) ]() What you can verify without calling: After importing your number into Retell, you can confirm the configuration is saved correctly by checking that the number appears in your Retell dashboard with the correct termination URI and agent assignment. [(2) ]() [(3) ]() To test the actual SIP connectivity: The documentation indicates that testing requires making actual calls: [(2) ]() [(4) ]() [(5) ]() - **Inbound test**: Call the DID from an external phone to verify the call traverses through your provider to Retell [(4) ]() [(5) ]() - **Outbound test**: Place an outbound call via the Retell dashboard or Create Phone Call API using the imported number [(2) ]() [(4) ]() [(5) ]() Why an actual call is needed: SIP trunk connectivity verification requires testing the actual signaling path (SIP INVITE messages) and media path (RTP packets) between Ziwo's infrastructure and Retell's SIP server at
sip.retellai.com
. [(1) ]() [(4) ]() [(5) ]() This can only be confirmed when a real call is placed. Recommendation: Make a quick test call to the DID to veri
fy the SIP integration is working correctly. You can keep it brief—just confirm the call connects to your Retell agent and you hear bidirectional audio, then hang up. [(2) ]() [(4) ]() [(5) ]()
​ To learn more, see the sources I considered: [[1] Custom Telephony Overview](https://docs.retellai.com/deploy/custom-telephony) [[2] Twilio](https://docs.retellai.com/deploy/twilio) [[3] Import Phone Number](https://docs.retellai.com/api-references/import-phone-number) [[4] Avaya](https://docs.retellai.com/deploy/avaya) [[5] Genesys](https://docs.retellai.com/deploy/genesys) ​
Feel free to tag @AI Evy with additional questions. I also appreciate feedback, it helps me improve.
r
@AI Evy 1. I would need RTP IPs if this is different from the SIP endpoint shared 2. Does Retell use the same IPs for RTP as SIP, or do they have a separate RTP range? If seperate I would need the details please
a
Thanks for the follow-up, @Rowan. I'll get back to you soon.
### Answer ### Answer Based on the official documentation, Retell uses the same IP ranges for both SIP signaling and RTP media traffic. [(1) ]() IP addresses to whitelist for both SIP and RTP: For United States traffic, whitelist these CIDR blocks: [(1) ]() [(2) ]() -
18.98.16.120/30
-
143.223.88.0/21
-
161.115.160.0/19
Additional webhook/API traffic IP: [(2) ]() -
100.20.5.228
For European servers: A community member asked about European server IPs and mentioned seeing voice traffic from different IP addresses. However, the official documentation currently only lists the U.S. IP ranges above. [(1) ]() Important notes: 1. There is no separate RTP range documented - the same IPs handle both SIP signaling and RTP media. [(1) ]() [(2) ]() 2. If you're using European servers on Ziwo's side and experiencing issues, you may want to reach out to Retell support at the [Customer Support Portal]() or [support@retellai.com](support@retellai.com) to confirm whether additional European IP ranges need to be whitelisted. [(3) ]() [(4) ]() 3. Make sure Ziwo whitelists all three U.S. CIDR blocks for both inbound and outbound traffic to ensure proper SIP and RTP connectivity. [(1) ]() [(2) ]()
​ To learn more, see the sources I considered: [[1] Connect Retell voice agents to custom telephony](https://docs.retellai.com/deploy/custom-telephony) [[2] Good day!](https://discord.com/channels/1186424376300679209/1417112945564848158) [[3] Avaya](https://docs.retellai.com/deploy/avaya) [[4] Connect Genesys Cloud to Retell](https://docs.retellai.com/deploy/genesys) ​
Feel free to tag @AI Evy with additional questions. I also appreciate feedback, it helps me improve.
r
@AI Evy hi We use custom SIP trunking and are evaluating a move to speech-to-speech. Two quick questions about OpenAI's new GPT-Live models (full-duplex, API not yet public): 1. Is GPT-Live on your roadmap once the API is released? Any rough timeline? 2. Will it work over custom SIP telephony like the current Realtime option does? Thanks, Rowan
a
Thanks for the follow-up, @Rowan. I'll get back to you soon.
### Answer I don't have specific information about GPT-Live integration plans, timelines, or compatibility with custom SIP telephony in the available sources. However, I can confirm that Retell has historically been quick to integrate new OpenAI models. The changelog shows that GPT-4o was made available on Retell shortly after OpenAI's release, with a note stating "The voice interface API has not been released yet, but we plan to integrate it as soon as it becomes available." [(1) ]() For your current setup, Retell supports custom SIP trunking with various LLM options including OpenAI models, and the platform is designed to work with elastic SIP trunk configurations. For definitive answers about GPT-Live roadmap and timeline: I'd recommend checking: - [Documentation]() - [Hire certified Retell partner]() Would you like a general answer based on my knowledge about how Retell typically handles new model integrations?
​ To learn more, see the sources I considered: [[1] Platform Changelogs | Retell AI](https://www.retellai.com/changelog) ​
Feel free to tag @AI Evy with additional questions. I also appreciate feedback, it helps me improve.
r
@
@AI Evy 1. Does Retell currently support a native speech-to-speech voice engine (e.g. OpenAI's Realtime API / GPT-Realtime-2.1), as opposed to the STT → LLM → TTS pipeline we use today? 2. Is GPT-Live (OpenAI's new full-duplex model) on Retell's roadmap once its API is released? Any timeline? 3. Would either of the above work over custom SIP telephony? We connect to Retell via SIP trunking + IP whitelisting 4. If speech-to-speech is supported, can we still use a custom/cloned voice (we currently use ElevenLabs cloning), or are we limited to the model's preset voices?
a
Thanks for the follow-up, @Rowan. I'll get back to you soon.
### Answer 1. Speech-to-Speech Support Yes, Retell currently supports native speech-to-speech models. The platform offers several OpenAI Realtime API options: [(1) ]() [(2) ]() Available speech-to-speech models include: -
gpt-realtime-2
-
gpt-realtime-1.5
-
gpt-realtime
-
gpt-realtime-mini
These are configured using the
s2s_model
parameter when creating or updating a Retell LLM. Note that you can only set either
s2s_model
or
model
(the text-based LLM) , not both. 2. GPT-Live Roadmap I don't have specific information about GPT-Live integration plans or timelines in the available sources. For roadmap questions about unreleased OpenAI models, I'd recommend: - [Documentation]() - [Hire certified Retell partner]() 3. Custom SIP Telephony Compatibility Based on the architecture described in the sources, Retell's Voice AI system includes "Automatic Speech Recognition (ASR) , Large Language Model (LLM) , Dialog Management & Turn-Taking, Text-to-Speech (TTS) Synthesis, and API & System Integration." [(3) ]() The sources confirm that speech-to-speech models work within Retell's standard telephony infrastructure, which includes custom SIP trunking support. Since you've already successfully configured custom SIP with IP whitelisting, speech-to-speech models should work over your existing SIP trunk setup. 4. Custom Voice Support with Speech-to-Speech I don't have explicit documentation about whether custom/cloned voices (like ElevenLabs) can be used with speech-to-speech models. The
s2s_model
parameter documentation doesn't specify voice customization options. [(1) ](
rences/create-retell-llm) [(2) ]() For clarification on voice cloning compatibility with speech-to-speech models: - [Documentation]() - [Hire certified Retell partner]()
​ To learn more, see the sources I considered: [[1] Create Retell LLM](https://docs.retellai.com/api-references/create-retell-llm) [[2] Update Retell LLM](https://docs.retellai.com/api-references/update-retell-llm) [[3] Voice AI](https://www.retellai.com/glossary/voice-ai) ​
Feel free to tag @AI Evy with additional questions. I also appreciate feedback, it helps me improve.
r
@AI Evy how to fill my clients profiles from the database to Retell so the agents know who is this ?
a
Thanks for the follow-up, @Rowan. I'll get back to you soon.
### Answer To populate client profile information from your database so Retell agents know who they're speaking with, use dynamic variables. [(1) ]() How it works: Dynamic variables allow you to inject personalized data into your agent's responses for each specific call using the
{{variable_name}}
syntax. You can pass client information like name, account details, or any custom data from your database. For outbound calls: When using the Create Phone Call API, set your client data in the
retell_llm_dynamic_variables
field: [(1) ]() [(2) ]()
Copy code
json
{
  "user_name": "John Smith",
  "product_name": "Premium Plan",
  "account_status": "active"
}
[(1) ]() All values must be strings. For inbound calls: Supply dynamic variables through the Inbound Call Webhook, where you can query your database and return the client's information for that specific call. In your agent prompts: Reference the variables using double curly braces:
Copy code
"Hello {{user_name}}, I understand you're interested in {{product_name}}. How can I help you today?"
Where you can use dynamic variables: - Prompts (agent instructions and personality) - Begin message (opening greeting) - Tool configurations - Webhook URLs Important notes: - All values in
retell_llm_dynamic_variables
must be strings - Numbers should be stored as
"42"
and booleans as
"true"
- You can set agent-level default values as fallbacks This approach lets you pull client profiles from your database and personalize each conversation based on who's calling.
​ To learn more, see the sources I considered: [[1] Dynamic variables: personalize Retell agent calls](https://docs.retellai.com/build/dynamic-variables) [[2] Create Web Call](https://docs.retellai.com/api-references/create-web-call) ​
Feel free to tag @AI Evy with additional questions. I also appreciate feedback, it helps me improve.
r
@AI Evy how to import the data walk me through retell
a
Thanks for the follow-up, @Rowan. I'll get back to you soon.
### Answer To import client data from your database into Retell so agents can recognize callers, you'll pass the data through dynamic variables when initiating calls. Here's a step-by-step walkthrough: ## For Outbound Calls When making outbound calls via the Create Phone Call API, include client data in the
retell_llm_dynamic_variables
field: [(1) ]()
Copy code
javascript
import Retell from 'retell-sdk';

const client = new Retell({
  apiKey: process.env['RETELL_API_KEY'],
}) ;

const phoneCallResponse = await client.call.createPhoneCall({
  from_number: '+14157774444',
  to_number: '+12137774445',
  retell_llm_dynamic_variables: {
    "customer_name": "John Doe",
    "account_id": "12345",
    "account_status": "active",
    "last_purchase": "Premium Plan"
  }
}) ;
Important: All values must be strings - convert numbers to
"42"
and booleans to
"true"
. [(2) ]() ## For Inbound Calls Supply dynamic variables in the Inbound Call Webhook response. When a call comes in, Retell will hit your webhook, where you can: 1. Query your database using the caller's phone number 2. Return the client's profile data in the webhook response ## Using the Data in Your Agent Reference the variables in your agent's prompt using
{{variable_name}}
syntax:
Copy code
"Hello {{customer_name}}, I see your account status is {{account_status}}. How can I help you today?"
## Where You Can Use Dynamic Variables - Agent prompts and instructions - Begin message (opening greeting) - Tool configurations and descriptions - Webhook URLs ## Setting Default Values You can configure agent-level default dynamic variables as fallbacks when specific call data isn't provided. This approach lets you query your database and inject client-specific information into each conversation, making your agents context-aware about who they're speaking with.
​ To learn more, see the sources I considered: [[1] Create Phone Call](https://docs.retellai.com/api-references/create-phone-call) [[2] Dynamic variables: personalize Retell agent calls](https://docs.retellai.com/build/dynamic-variables) ​
Feel free to tag @AI Evy with additional questions. I also appreciate feedback, it helps me improve.