> ## Documentation Index
> Fetch the complete documentation index at: https://docs.dograh.com/llms.txt
> Use this file to discover all available pages before exploring further.

# Dograh Managed SIP

> Connect your own SIP carrier or PBX to the SIP domain Dograh provisions for your organization

## What you get, and what you bring

Every Dograh organization is provisioned with a telephony configuration named
**Dograh Cloudonix SIP**. It gives you a SIP domain that is already connected
to the Dograh voice pipeline — you do not need a Cloudonix account, and there
is nothing to pay for or set up on the Cloudonix side.

What it does **not** include is phone service. Dograh does not sell or resell
phone numbers, and no calls flow until you connect a carrier:

| Dograh provides                                  | You provide                                    |
| ------------------------------------------------ | ---------------------------------------------- |
| A SIP domain and its inbound SIP endpoint        | A SIP trunk from your carrier, or your own PBX |
| A fixed outbound origin IP to allow on your side | Phone numbers (DIDs) on that trunk             |
| The voice agent runtime on both call directions  | Any per-minute charges your carrier bills      |

<Note>
  Until you finish the steps below, **Dograh Cloudonix SIP** cannot place a
  call. The configuration page shows a setup checklist that tracks exactly
  which of these steps are still outstanding.
</Note>

<Tip>
  **Don't have a SIP trunk or PBX?** This page is not the shortest path. Open an
  account with a telephony provider instead — see
  [Telephony Integration](/integrations/telephony/overview) for the list — and
  you can be calling in a few minutes with no SIP configuration at all.
</Tip>

If you would rather bring your own Cloudonix account instead of using the
provisioned domain, follow [Cloudonix Integration](/integrations/telephony/cloudonix)
and add a separate configuration. Self-hosted deployments can also connect their
own Asterisk via [Asterisk ARI](/integrations/telephony/asterisk-ari).

## Prerequisites

* A SIP trunk from a carrier (or a PBX such as Asterisk or FreeSWITCH) that can
  accept calls from a fixed IP address and send calls to a hostname
* At least one phone number (DID) on that trunk
* Permission to change the trunk's IP allowlist and routing on the carrier side

## Step 1: Read your SIP endpoints from Dograh

1. Go to **Telephony configurations** and open **Dograh Cloudonix SIP**
2. Expand **SIP connectivity**
3. Choose the **region** closest to your carrier — `Global`, `India`, or `UAE`.
   Every value on the panel is region-specific, so set this first
4. Note the two values you will need on the carrier side:
   * **Inbound hostname** and **ports** — where your carrier sends calls *to* Dograh
   * **Outbound origin IP address** — the address Dograh's calls arrive *from*

## Step 2: Point your carrier at Dograh (inbound)

On your carrier or PBX, route your DIDs to the inbound hostname and port shown
in **SIP connectivity**. UDP and TCP share one port; TLS uses a separate one.

This is the direction that makes your agent answer calls.

## Step 3: Create the outbound trunk (outbound)

Dograh needs to know where to send calls when your agent dials out.

1. On your carrier, allow the **origin IP address** from Step 1 to send SIP
   traffic to your trunk. Calls from Dograh are unauthenticated and identified
   by source IP, so this step is required
2. Back in Dograh, under **Outbound trunks**, choose **Add trunk** and fill in:
   * **Name** — any name you choose. Letters, digits and hyphens only
   * **SIP domain** — the hostname of your carrier or PBX, for example
     `sip.example.com`. Dograh puts this in both the SIP `To` header and the
     Request-URI
   * **Region** — the edge Dograh dials your carrier from. The dialog shows
     the origin IP that region uses, which is the one to allowlist above
3. Leave **Enabled** on and choose **Save trunk**

You can add more than one trunk — a second carrier, or a failover. Each keeps
its own region, and each lists the numbers that dial out over it.

<Note>
  Without an enabled outbound trunk, outbound calls have nowhere to go. This is
  the most common reason a test call from the agent editor fails.
</Note>

## Step 4: Add your phone numbers

Cloudonix requires a caller ID on **every** outbound call and rejects the call
without one, so this step is required even if you only care about inbound.

1. Choose **Add number** on the trunk whose carrier gave you the number — the
   trunk is then filled in for you. (**Add phone number** further down works
   too; there you pick the **Outbound trunk** yourself, or leave it unset for a
   number that belongs to no trunk.)
2. Enter a number your carrier delivers to you, in
   [E.164](https://en.wikipedia.org/wiki/E.164) format — `+`, country code,
   then the number, with no spaces or dashes, for example `+14155550123`
3. Mark the number you want outbound calls to present as the **default caller ID**

Add every DID you intend to use. A number your carrier does not own will be
rejected by the carrier even though Dograh accepts it — which is why the trunk
assignment matters as soon as you have more than one carrier. With a single
trunk, Dograh uses it for every number and the field can be left alone.

## Step 5: Route inbound calls to an agent

Only needed if you want to receive calls.

1. In the **Phone numbers** section, edit the number that should be answered
2. Set its **Inbound workflow** to the agent that should pick up
3. Save

Dograh pushes the inbound webhook URL to the Voice Application on the
provisioned domain automatically. You still need the DID bound to that
application — see
[Step 2 of the Cloudonix inbound setup](/integrations/telephony/cloudonix#step-2-create-the-voice-application-and-link-dnids).

## Step 6: Place a test call

1. Open an agent, and make sure it is saved with no validation errors
2. Choose **Phone Call**
3. Select **Dograh Cloudonix SIP**, pick your caller ID, enter a destination
   number, and start the call

## Troubleshooting

<AccordionGroup>
  <Accordion title="'Not ready for outbound calls' when starting a call">
    The setup checklist on the configuration page names the missing step. The
    two blocking ones are an enabled trunk (Step 3) and at least one phone
    number (Step 4).
  </Accordion>

  <Accordion title="Outbound calls fail immediately at the carrier">
    * Confirm the origin IP from **SIP connectivity** is allowed on your trunk
    * Confirm the **SIP domain** on the trunk resolves and accepts SIP on the
      port your carrier expects
    * Confirm the caller ID you are dialing from is a number your carrier
      actually owns — many carriers reject unowned caller IDs
  </Accordion>

  <Accordion title="Inbound calls never reach the agent">
    * Confirm your carrier is sending to the **inbound hostname and port** for
      the region selected on the panel, not another region's
    * Confirm the called number exists in the **Phone numbers** list and has an
      **Inbound workflow** assigned
    * Confirm the DID is bound to the Voice Application on the domain
  </Accordion>

  <Accordion title="You changed regions after setup">
    The inbound hostname, ports, and outbound origin IP all change with the
    region. Each trunk carries its own **Region**, so edit the trunk to move it,
    then update your carrier with the new values.
  </Accordion>
</AccordionGroup>