> For the complete documentation index, see [llms.txt](https://docs.didforsale.com/llms.txt). Markdown versions of documentation pages are available by appending `.md` to page URLs; this page is available as [Markdown](https://docs.didforsale.com/voice-and-sms-apis/voice-and-sms/didml-verbs/dial/sip.md).

# \<sip>

**Role:** \<sip> keyword can be used to send the call directly to you existing SIP Server, SIP URI or SIP End Point. Make sure you accept calls from all our SIP Gateways. Otherwise your SIP server might reject the call.  Call forwarding to SIP gateways does not incur any extra charge.

**Use:**&#x20;

* *Existing SIP Gateway*: Already have a SIP Gateway or SIP PBX, and agents registered on that gateways. This is is great way to dynamically forward the calls to your SIP Servers.
* Load Balancing/Failover: Dynamically send call to multiple SIP gateways depending on the load and traffic.

SIP Keyword can be used alone or it can be nested inside dial command.

### Attributes

SIP does not have any attributes at this time.

#### Here is an example of how \<sip> is used within code. Call will be sent to <dialed_number@sip.example.com>

```
<?xml version="1.0" encoding="UTF-8"?> 
<Response>
     <dial>
         <sip>sip.example.com</Say>
     </dial>
 </Response>
```

In another example you can send the call to a specific uri. In the example below, we will forward the call to conference bridge at sip.example.com.

```
<?xml version="1.0" encoding="UTF-8"?> 
<Response>
     <dial>
         <sip>sip:conference@sip.example.com</Say>
     </dial>
 </Response>
```

### Troubleshooting

* Make sure attributes inside the nodes are in quotes " ".


---

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