In some instances you might want to send inbound calls, or transfer calls, from other communication systems into Spoke Phone.
For example you may have a contact center running on Amazon Connect, Twilio Flex, or RingCentral, etc., and you want contact center agents to be able to transfer a call to your Finance Team, or to a specific user on Spoke Phone.
Or your main company office in Sydney is using a legacy Cisco phone system, but the support team is in Manila on Spoke Phone. When customers call in to your Sydney office and your Cisco IVR says "Press 2 for Support", you want to send the customer call over to the Manila team, who use Spoke Phone.
You can easily achieve the above, and more, using industry standard SIP trunks. By configuring a SIP trunk into Spoke Phone you can then configure your existing platform to forward calls using SIP to a Spoke Phone user, device, or a team of people in a call/hunt group.
- SIP to a user - Rings the user directly
- SIP to device - Rings that one extension (which could be a kitchen phone)
- SIP to call group - Offers the call to the users in a call group. Offers the call based on the hunt and rollover rules that you set up for the call group in Spoke
You need to contact Spoke Phone in order to have a SIP Domain provisioned on your account. Contact us via the chat icon at the bottom right of this page.
Authentication limits access to your SIP Domain from only approved devices and users.
Spoke Phone authentication supports both IP Access Control List or a credential list. If you configure both, then both ACLs and credential lists are enforced to secure your account.
Preferred access method
Spoke Phone's default and preferred method, is IP Access Control List (IP ACL).
You will need to provide a list of IP addresses that will be sending SIP Invites to Spoke Phone (IP ACL).
Spoke will only accept SIP traffic originating from the IPs in this list - all other packets will be dropped. You must specify a full IP address; no IP wildcarding is supported.
Credential Lists are sets of usernames and passwords that will be accepted by your SIP Domain.
If a Credential List is configured, your SIP INVITE will be challenged with a 407 Proxy Authentication Required requesting the appropriate user name and password.
For each username, you must set a password that meets the following minimum requirements:
* Minimum of 12 characters * At least one mixed case * At least one digit
Spoke Phone does not store the passwords you provide for usernames in cleartext; instead, the passwords are MD5 hashed in accordance with the digest authentication specification. Once a password is set, Spoke Phone does not provide a way to retrieve the stored password.
Once provisioned, you will be provided with your SIP Domain.
The SIP Domain we provide you will follow this pattern:
In order to send the call to the correct user, device, or call group, you will append the extension number to the SIP Domain to create your SIP Invite.
Spoke Phone issues extension numbers to users, devices, and call/hunt groups. You can see and edit your list of extensions in the Spoke Account Portal.
You simply append the extension number to the SIP Domain like this:
so if the extension number is 200, then:
So if the SIP Domain we provide you is:
And the Spoke extension you want to send calls to is 200, then the SIP address to forward calls to will look like this:
Updated 2 months ago