Cebod Telecom
  • Introduction
  • Customer Onboarding
  • FAQs (Frequently Asked Questions)
  • Cebod Telecom Web Portal
    • Dashboard
    • Activities
      • Call records
      • Messages
      • Active calls
      • SMS records
      • Faxes
    • Phone System
      • Phone Numbers
      • IVR (Auto Receptionist)
      • Users (Extensions)
        • Dynamic Caller ID
      • Manage Devices
      • Manage Groups
      • Manage Conferences
      • Postcall Actions
      • Webhooks
      • Time Conditions
      • Call Blocking
      • Call Center
        • Active Calls
        • Agents
        • Queue
        • Agents to queue
        • Agents Status
        • Agent Statistics
    • Tools / Settings
      • Manage Phone Lines
      • Add 911 Address
      • SMS
      • International Calling
      • Outbound Rates
      • Manage Recordings
        • AI Text-to-Speech Feature
      • Global Settings
      • Voicemail Greetings
      • CNAM
    • My Account
      • Profile
      • Change Password:
      • Add Balance
      • Payment History
      • Vouchers
      • Invoices
      • Auto Refill
      • Logout
  • How To's
    • Set up BLF
    • Register a New Account
    • Setup Call Forwarding
      • Call Forwarding to an Extension
      • Call Forwarding Manually to a Cell Phone
    • Dial-By-Name Directory
    • Voicemail
      • Setting up the Voicemail Menu
      • Accessing Voicemail
      • Navigate through the Voicemail Menu
    • Provisioning
      • Cisco 8841
    • Setup Speed Dial
    • Cebod Telecom Free Applications
      • MS Windows Desktop Softphone
      • Mac Desktop Softphone
      • Apple IOS Mobile App
      • Android Mobile App
      • Desktop SMS/MMS Application
      • Click to Call Browser Extension
    • GS Wave
    • Setting Up Whisper Announcement
  • Enable Call Recording
  • Features
    • Manage Phone Features
      • Call Block
      • Call transfer
      • Call Queue
      • Call Center
      • Active Call Features
      • Ring Groups
      • Call Forward
      • Call Recording
      • Call Park
      • Paging
      • Voicemail
      • Call Monitoring
      • Hide Caller ID
    • Missed Call SMS
    • Missed Call Email Notification
    • 3-Way Calling
    • Conference Bridge
    • Setting up Time Condition
    • Setup Email FAX - Incoming
    • Sending eFax using email
  • API
    • Voice API
    • Quick Start
      • Identify Caller
      • Collect Response
    • didML reference doc
      • Voice API reference params
      • Voice API response
      • didML Verbs
        • <dial>
          • <client>
          • <conference>
          • <number>
          • <queue>
          • <sip>
        • <enqueue>
        • <fax>
        • <gather>
        • <hangup>
        • <pause>
        • <play>
        • <postcall>
        • <record>
        • <recordcall>
        • <redirect>
        • <reject>
        • <say>
        • <sms>
Powered by GitBook
On this page
  • Attributes
  • Troubleshooting

Was this helpful?

  1. API
  2. didML reference doc
  3. didML Verbs
  4. <dial>

<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:

  • 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 " ".

Previous<queue>Next<enqueue>

Last updated 5 years ago

Was this helpful?