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SIP Technical Details


This document provides a technical, detailed explanation of SIP integration with Flip.

For a basic overview of telephony terminology, please see General FAQ.

SIP Integration Process

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Customer Requirements

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These are the requirements for a smooth SIP integration:

  1. Forward calls to Flip with the correct caller ID.

  2. In the unlikely event that Flip’s SIP domain is unavailable, forward calls to the Main Call Queue.

  3. When Flip forwards to the Main Call Queue, we must be able to mask the caller ID (so that it shows up as the user’s caller ID).

  4. Flip recommends allocating 2x your maximum simultaneous call load, enough SIP channels to make sure no calls are blocked. Ensure you have enough buffer - For our transportation customers, this corresponds to about 2 * 0.0015 * (average number of calls per week).

Specific Requirements for Customers in Transportation 

  1. The Main Call Queue that Flip accesses must be reachable via PSTN or SIP.  If Flip forwards calls to 1234567890 on your PBX, for example, then dialing the same number from your cell phone must reach the same endpoint. This may require provisioning more DIDs for use as the fallout number.

  2. Forwarding to the PSTN should be achievable through your trunk.

Flip SIP Specifications

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The Flip SIP system has the following specifications:

Flip SIP Domain

See your Portal

Flip SIP Port


RTP Ports

10,000 - 20,000

Accepted Codecs

G711 (μ-Law, A-Law), G729


Out-of-band by default

Depending on your provider or network configuration, you may need to whitelist the following IPs for your region.




North America



Call Flow

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There are two ways to send phone calls to Flip via a SIP Integration:

  1. (Recommended) Forward calls to Flip from your PBX. We have integrated with a number of PBXs (ININ, Avaya, Cisco, Fortinet, etc.), so chances are we’ve integrated with yours. This is the most cost-effective solution.

  2. Your phone carrier can forward calls to Flip and Flip can dial back to you when required.

PBX Option

PBX Integration Option

Forwarding calls from your PBX:

  1. The carrier sends the call to your PBX. 

  2. Your PBX sends the call to Flip’s PBX at the number that the caller has dialed.

  3. If Flip cannot handle the call, it sends the call back to the Main Call Queue Number on your PBX.
    The text that you input into the Outbound Routing configurations will translate directly to the to-user in the SIP INVITE sent back to your SIP domain or IP. It can be a phone extension (whether in e.164 format or not), or any other alphanumeric string.

  4. Your PBX then forwards the call to your call center. 

Carrier Option

Carrier Integration Option

If the PBX Implementation is not possible for you, Flip can receive and forward calls directly from the carrier. In this scenario, you will provision another SIP trunk with your carrier for Flip. Before trying any of these steps, please schedule a call with Flip to make sure it is absolutely necessary to go through a carrier. 

Receiving and forwarding calls from the carrier:

  1. The carrier forwards the call to Flip.

  2. If it is not possible to automate the call, Flip dials the Main Call Queue Number on the carrier’s trunk.

  3. The carrier calls the fallout number on the PSTN.

  4. The call reaches your SIP trunk or PRI line for the carrier, which forwards the call to your PBX. 

  5. Your PBX routes the call to the appropriate call queue.

Dialing to PSTN


Note: This section is specific to our transportation customers only.

Sometimes, Flip may need to dial out to PSTN. The best example of this is if a customer calls and opts to speak to their driver. To forward the call to the driver, we need to forward to the PSTN. 

This is usually easily configurable on your PBX. If we forward to one of the fallout numbers configured to go to the call queue, then the call should go to the call queue. However, if we forward the call to a non-fallout number, then your PBX should route the call to that number in the PSTN.

As an example, if someone dials their cell phone number on your trunk, then they should receive that call on their phone as it’s not one of the fallout numbers.

Dialing PSTN Option


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The following list outlines some common issues and their solutions:


The user cannot hear the IVR even though the call goes through properly.
Possible solutions:

  • Validate that the enabled RTP ports match those listed in the Flip SIP Specifications.

  • If your PBX is behind NAT, you may be signaling Flip’s PBX to forward to an incorrect address.


Flip is not accepting the call.
Possible solutions:

  • Confirm that you are sending from and to the correct address.

  • Confirm that the extension you are sending matches the Internal Telephony Number.

  • Confirm that the port number you’re sending SIP traffic to is 5060.

  • Flip’s security system may not have recognized the IP address before your system started sending RedRoute OPTIONS pings.


You hear “An application error has occurred” when you call.
Possible solutions:

  • Confirm that the extension you are sending matches the Internal Telephony Number.


Routing back to the Main Call Queue is not working.
Possible solutions:

  • Your firewall may not have whitelisted Flip’s IP.

  • Confirm that you've entered the Main Call Queue number in the exact format your PBX is configured to accept.


DTMF keypresses are not handled well.
Possible solutions:

  • Have Flip set the DTMF mode to inband.

If these solutions don't fix the problem, please open a support ticket with us through the Flip portal.


We hope you find this article clear and helpful. Our team is always available to answer your questions, and and provide all support necessary. You can direct questions at any time to support@flipcx.com.