---
title: "VOIP Desk Phone Integration (Twilio SIP)"
description: "Learn how to connect your physical VOIP phone to Aesthetix CRM via Twilio SIP."
url: https://help.aesthetixcrm.com/en/articles/1223-voip-desk-phone-integration-twilio-sip
collection: "Making & Receiving Calls"
updated: 2026-10-20
language: en
---

# VOIP Desk Phone Integration (Twilio SIP)

Learn how to connect your physical VOIP phone to Aesthetix CRM via Twilio SIP.

## **What is the VoIP Deskphone Integration?**

This feature empowers users to **connect their physical VoIP deskphones directly to Aesthetix CRM.** It provides a stable link between your CRM and your physical phone system, ensuring that all call activities, from making outbound calls to receiving inbound calls, are managed inside Aesthetix CRM. By leveraging this integration, you can enhance call-handling capabilities and maintain a consistent connection to your customer data.

## **Permissions To Set Up VoIP Deskphones**

-   **Admins** – can provision devices only inside their own location  
    
-   **Other users** – read-only; they’ll see a tooltip prompting them to “Ask an admin to configure this.”
    

## **Technical Requirements & Recommended Deskphones**

**Network & protocol prerequisites**

-   SIP over UDP, TCP, or TLS
    
-   Open outbound ports **5060/5061** for SIP signalling
    
-   Open UDP **10000-20000** for RTP audio
    
-   PoE switch or external power adaptor
    

**What to look for when buying hardware**

-   “SIP” and “PoE” on the spec-sheet
    
-   At least two programmable line keys
    
-   Avoid carrier-locked or proprietary-provisioning devices
    

**Popular models**

-   Yealink T54W / T58W
    
-   Poly VVX 450
    
-   Grandstream GXP 2170
    
-   Snom D785
    
-   Cisco 7841
    

## **Connect VoIP Deskphones to Aesthetix CRM**

**Step 1:** Navigate to the Wizard

Log in and go to **Settings → Phone Numbers → Advanced Settings → VoIP Deskphone (SIP) → Get Started**.

**Step 2:** SIP Server Configuration

-   Confirm or edit your SIP domain. A suggested domain such as `<LocationName>.sip.ashburn.twilio.com` appears.
    
-   You can change it **once** before saving.
    

![](https://staticfiles.us.gleap.ai/ghelparticle/1WMjzI1TqRF0qVUd5e7qfhL8l0leB79wUmpVAbomMucf6XwJGNEKJI3EnSgSFvIzgBxMjFisJBl.png)

**Step 3:** SIP User

Choose an extension (number or text) plus a strong password and save them somewhere safe.

For numeric extensions, we recommend 3 to 5 digits that don't start with 0, 1, or 9. Matching the last digits of the user's direct-dial number makes extensions easy to remember. The saved password can't be viewed again later, so if it's lost, reset it from **Manage Devices** (covered below).

![](https://staticfiles.us.gleap.ai/ghelparticle/XmoOreEY6J80OTAIv1CgFQ4WmjA2CuXsQ2sgwcJKBlhmzAlWlKCoN6MgTegxsSAgN0k6LpWSj0v.png)

**Step 4:** Assign User

Each VoIP deskphone needs to be connected to a user in your account. Select one user from the Assign to User dropdown.

Make sure the user's individual phone settings are configured to use deskphone. Navigate to Settings > My Staff > Choose staff > Edit > Call & Voicemail Settings, please enable "Deskphone" for the options you need. Enable it for "Forward Calls to" to receive calls addressed to you on the Deskphone. If you're receiving calls as part of "Ring multiple", and you want to receive them on the deskphone, enable Deskphone under "Ring all".

If an IVR (phone menu) routes calls to this user, also select Deskphone as the user's **Default Channel for IVR** so those routed calls ring the deskphone.

![](https://staticfiles.us.gleap.ai/ghelparticle/QanaWk6EChPx4EyoiCdYjMhJ6cv6isBOTnyQX2xR5pG38I4W7Y9AzxS01udk1T9ivGD7r9DsiAt.png)

![](https://staticfiles.us.gleap.ai/ghelparticle/XnHSnDQJAACuN1q7giS7mottB8UBnd09COg2KkLlwu5x61VfxZedkvwXHLFfjiFItxQTxkkuDrY.png)

**Step 5:** Configure the Physical Phone

-   Enter the SIP Domain, Username, and Password into the phone’s **Registrar / Server / Proxy** fields (varies by phone model and manufacturer).
    
-   Note: You should utilize your IT team or MSP to assist you with this process as you have to set this up in your office and re-configure your existing VOIP phones. We recommend you start with your least used desk phone.
    
-   If the phone has a **Keep Alive** setting (often under Account > Advanced), enable it and set the type to **Options**. This helps inbound calls reach the deskphone reliably.
    
-   Before testing, confirm the phone shows the account as **Registered** (or a similar success indicator).
    

**Step 6:** Run the Built-In Test Calls (Optional)

-   **Navigate** to VOIP Deskphone (SIP) > Test Calls > Simulate Calls.
    
-   **Outbound test** – Dial the displayed number; you should hear “This is a test call” three times if successful.
    
-   **Inbound test** – Click _Inbound Test Call_; the deskphone should ring.
    

![](https://staticfiles.us.gleap.ai/ghelparticle/kiLH7AfJFW3UjAVj6tUoRq8bOF7qjslt1m4f2t6J3q2qQMi4PTgG7iAVrg95Jl8RC8YdrY8j663.png)

> **Info:** Need Help? Message us in customer support, and we can help you troubleshoot.

## Manage Deskphone Devices

Admins can reset a device's password or remove a device from **Settings → Phone Numbers → Advanced Settings → VoIP Deskphone (SIP) → Manage Devices**.

**Reset a device password:** Click the **pencil icon** next to the SIP user, enter a new password, and save. Then enter the same new password on the physical phone. Until the phone is updated, it will fail to register.

**Delete a device:** Click the **trash can icon** next to the SIP user. This immediately removes the device's credentials, and the phone can no longer register with that SIP user.

## Calling and Transferring Between Deskphones

**Call another deskphone by extension:** Dial the other deskphone's extension directly. For example, if a teammate's extension is 201, dial 201. No outside number is needed, but both deskphones must be set up and registered.

**Transfer a call to another deskphone:** During a call, start the transfer on your handset, dial the teammate's extension, and complete the transfer on the phone.

**Note:** Deskphones support **blind transfers** only (the call is sent straight to the other extension). Warm transfers, where you speak with your teammate before handing off the call, are not currently supported from a deskphone.

## Troubleshooting

<table style="min-width: 75px;"><colgroup><col style="min-width: 25px;"><col style="min-width: 25px;"><col style="min-width: 25px;"></colgroup><tbody><tr><th colspan="1" rowspan="1"><p><strong><strong>Symptom</strong></strong></p></th><th colspan="1" rowspan="1"><p><strong><strong>Likely Cause</strong></strong></p></th><th colspan="1" rowspan="1"><p><strong><strong>Fix</strong></strong></p></th></tr><tr><td colspan="1" rowspan="1"><p><strong>401 / 403 Unauthorized</strong></p></td><td colspan="1" rowspan="1"><p>Wrong username or password (case-sensitive)</p></td><td colspan="1" rowspan="1"><p>Re-enter credentials</p></td></tr><tr><td colspan="1" rowspan="1"><p><strong>No audio</strong></p></td><td colspan="1" rowspan="1"><p>Firewall blocking RTP (UDP 10000-20000)</p></td><td colspan="1" rowspan="1"><p>Open ports / disable SIP ALG</p></td></tr><tr><td colspan="1" rowspan="1"><p><strong>Phone rings but won’t answer</strong></p></td><td colspan="1" rowspan="1"><p>NAT / SIP ALG interfering with RTP</p></td><td colspan="1" rowspan="1"><p>Disable SIP ALG on router</p></td></tr><tr><td colspan="1" rowspan="1"><p><strong>Can’t save SIP domain</strong></p></td><td colspan="1" rowspan="1"><p>Name already taken</p></td><td colspan="1" rowspan="1"><p>Edit the domain or accept the suggested increment</p></td></tr><tr><td colspan="1" rowspan="1"><p><strong>Phone doesn't register</strong></p></td><td colspan="1" rowspan="1"><p>SIP server, credentials, or network</p></td><td colspan="1" rowspan="1"><p>Check the SIP Domain, username, password, outbound ports 5060/5061, and SIP ALG</p></td></tr><tr><td colspan="1" rowspan="1"><p><strong>Outbound works, inbound doesn't ring</strong></p></td><td colspan="1" rowspan="1"><p>User routing, Keep Alive, or SIP ALG</p></td><td colspan="1" rowspan="1"><p>See the inbound steps below</p></td></tr><tr><td colspan="1" rowspan="1"><p><strong>Test ring works, but real inbound calls don't</strong></p></td><td colspan="1" rowspan="1"><p>Phone number call routing</p></td><td colspan="1" rowspan="1"><p>Check the number's routing, assigned users, IVR or Voice AI, Ring all, and the user's Deskphone settings</p></td></tr><tr><td colspan="1" rowspan="1"><p><strong>Phone stopped registering after a password reset</strong></p></td><td colspan="1" rowspan="1"><p>Old password still on the phone</p></td><td colspan="1" rowspan="1"><p>Enter the new SIP password on the handset</p></td></tr><tr><td colspan="1" rowspan="1"><p><strong>Still stuck?</strong></p></td><td colspan="1" rowspan="1"><p>–</p></td><td colspan="1" rowspan="1"><p>Click Contact Support with your SIP Domain &amp; handset model</p></td></tr></tbody></table>

### Inbound Calls Work but Outbound Calls Fail

1.  **Confirm the phone is registered.** If it isn't, recheck the SIP Domain, username, and password before anything else.
    
2.  **Verify the server and credentials exactly.** Use the SIP Domain from Aesthetix CRM as the phone's Registrar or SIP Server, and the exact SIP username and password you created.
    
3.  **Check outbound SIP ports.** Your network must allow outbound traffic on ports **5060/5061**.
    
4.  **Check the audio port range.** Allow UDP **10000-20000**. If calls connect but audio is missing or one-way, review this range and SIP ALG.
    
5.  **Disable SIP ALG** on your router or firewall if it is changing SIP traffic.
    
6.  **Run the built-in outbound test** from **VoIP Deskphone (SIP) → Test Calls**.
    
7.  **If you see 401 or 403 errors,** the username or password on the phone doesn't match what's saved in Aesthetix CRM.
    

Don't open SIP ports to the public internet or add inbound port-forwarding rules just because calls are failing. Work through the steps above first.

### Outbound Calls Work but Inbound Calls Don't Ring

1.  **Run the inbound test ring** from **VoIP Deskphone (SIP) → Test Calls**.
    
2.  **If the test ring fails, confirm the phone is still registered** and the SIP credentials are correct.
    
3.  **Confirm the device is assigned to the right user** in Setup SIP or Manage Devices.
    
4.  **Check the user's Call & Voicemail Settings.** Go to **Settings → My Staff → Edit → Call & Voicemail Settings** and enable Deskphone under "Forward Calls to."
    
5.  **Check Ring all and IVR.** If calls reach the user through Ring all or an IVR, select Deskphone as the user's default channel for each.
    
6.  **Review the phone number's call routing.** Make sure the user is part of the number's call flow and that an IVR or Voice AI isn't answering the call first.
    
7.  **Enable Keep Alive** on the handset with the type set to **Options**.
    
8.  **Disable SIP ALG** if the problem continues.
    
9.  **Retest with the test ring and a real inbound call.** If the test ring works but real calls don't reach the phone, the issue is the number's call routing, not the handset.
    

### What to Have Ready Before Contacting Support

-   SIP Domain
    
-   SIP extension or username (never send the password)
    
-   The user the device is assigned to
    
-   Phone manufacturer, model, and firmware version
    
-   Registration status shown on the phone
    
-   Results of the outbound test and the inbound test ring
    
-   Date, time, and time zone of a failed call
    
-   The calling number and the Aesthetix CRM number that was called
    
-   Any error codes shown
    
-   Screenshots of the user's call routing settings
    
-   Whether SIP ALG is enabled on your network
    

##   
  
**Frequently Asked Questions**

**Q: Which VoIP deskphones are compatible?**  
Most open-SIP models—see the recommended list above.

**Q: Can I record every call?**  
Yes. Once enabled, all calls are automatically recorded.

**Q: How do I perform a blind transfer during a call?** Use the **Blind Transfer** option on your deskphone UI. Warm transfers are not currently supported from a deskphone.

**Q: Is there an extra charge for using deskphones?**  
No. Deskphone calls share the same per-minute rates as web/mobile calls.

**Q: Can two deskphones point at the same Aesthetix CRM user?**  
No, you can only connect a single user to each VoIP deskphone.

**Q: Do you support SIP Trunking?**  
No, we do not support SIP Trunking as of now. So you cannot connect your existing PBX/Asterix to Aesthetix CRM as of now.

**Q: Why am I not able to receive the incoming call on my deskphone?**  
Make sure the user's individual phone settings are deskphone. Navigate to Settings > My Staff > Choose staff > Edit > Call & Voicemail Settings, please enable "Deskphone" for the options you need.   

  

**Q: Can I call a teammate's deskphone without dialing an outside number?**

Yes. Dial their extension directly. Both deskphones must be set up and registered.

**Q: I reset a device password and now the phone won't register. Why?**

The physical phone still has the old password. Enter the new SIP password on the handset.

**Q: Does the voicemail button on my deskphone open Aesthetix CRM voicemail?**

No. The handset's own voicemail feature is separate. Review voicemails and call activity inside Aesthetix CRM.