SIP Phone Integration's SIP Phone integration allows you to make browser based outgoing calls from the CRM to your Accounts and Contacts. Once your call is completed a Meeting record of category of Call will be auto logged in for you automatically showing the details of the call that took place.... pretty cool huh!  Well lets begin by looking at what you will need to get up and running.



In order to make use of the SIP integration you will need the following.

  • A SIP service that supports WebRTC technology so that it can make calls via the browser
  • Your SIP service must support the following SIP client (
  • Your SIP account details including your Web Proxy URL
  • Microphone and speakers or equivalent headset
  • Firefox or Chrome web browsers

If you are unsure if your SIP service supports the requirements above then please check and confirm with your SIP provider.  See the FAQs section for some example questions you can ask your SIP provider.


SIP Integration Set-up

Before setting up the SIP integration is is important you have read and understand the requirements above and have your SIP user details to hand.  The Configuration is in two parts, SIP must be enabled globally in the administration are and then details of the your SIP account must be entered on your user record in  Each user who wishes to use the Click to dial feature will need to have their SIP details entered on their user record.

SIP Admin Configuration

The administrator must enable SIP for online phone integration with in the admin area under Administration-->Plugins-->SIP.


SIP User Configuration

 Each user can configure their personal SIP settings under their My Profile-->Configuration-->SIP Configuration.  Admin users can configure the SIP configuration for other users if necessary by accessing that users record and following the same steps:


Here are some example values for an on-sip user configuration to give you an idea. 

  • Domain name:
  • SIP URI: sip:
  • Display Name:  Jim Smith
  • Authorization name:  myusername
  • Authorization password:  mypassword
  • Web Socket Proxy Url:  ws://
  • Outbound Proxy Url:  this is optional depending on your setup

Note - this is not a real account and will not work, the details above are just an example.  Sip configuration details will vary for different providers, you should confirm your configuration details with your sip provider

Making a Call

Once your settings are set up, you can now make an outgoing call by clicking any phone type field in the application.

  • Make outgoing calls (outgoing only, no incoming).
  • Call any phone type field in the Contacts, Leads, or Accounts modules (from the detail view, list view, portlets).
Be sure to allow a connection if your browser requests access in order for the call to go through.


To end the call, click the Hang up button.


A meeting record is auto created with the following fields auto populated (currently only supported in Chrome):

  • Name: Called [[NAME]] via SIP
  • Start Time: Start date and time of the call.
  • End Time: End date and time of the call.
  • Category: The category is populated as 'Call'
  • Attendees: populates the name of the Contact/Lead if calling from a contact/lead record.
  • Account: Populates the name of the Account if calling from an account record.

You can edit the meeting record to add any details in the description box if desired or to modify any of the other fields.




Q. We have physical IP phones in the office, will our calls be logged automatically in

A. Only outbound calls initiated from within and completed using WebRTC within the browser will create a meeting record in for you automatically.  Calls received or made using a physical IP phone will not create a meeting record in

Q. Will inbound calls be recorded automatically in

A. no, only outbound calls initiated from within and completed using WebRTC within the browser will create a meeting record in for you automatically. 

Q. Will automatically display contact information for any inbound calls I receive.

A. Currently our SIP integration only supports click to dial functionality for outbound calls. does not currently monitor incoming calls in any way and therefore cannot query the DB and return details of the inbound caller.

Q. What should I ask my SIP provider to confirm my service meets the requirements?

A. If you are unsure if your SIP service meets the requirements in this article then here are some example questions that you can put to them to confirm.

  1. Does my SIP service support WebRTC?
  2. Can I make SIP calls to external numbers via the browser?
  3. Do you support the following SIP client (
  4. What is my Web Proxy URL

Q. What is a Web Proxy URL?

A. If your SIP service can make calls to external phone numbers via the browser then it should be using a Web Proxy URL.  The Web Proxy URL in not mandatory for internal Sip account to Sip account calls, however if you plan to make Sip account to external Phone number calls via the browser then you will need to use a Web proxy URL.  If you are not sure what your Web Proxy URL address is then you should check with your SIP provider.  Since most calls made from are expected to be external calls to your Contacts we do require you provide a Web Proxy URL value.

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