Rather than having to port over your number to Talkdesk, you can maintain your existing phone numbers with a local carrier and have those calls forwarded to Talkdesk via SIP, so that agents are able to answer those calls directly within Talkdesk.
The phone number will appear as normal in your account and standard configurations and setup can be made as if the number was hosted with Talkdesk.
For example, if the phone number +123456789 is with an existing local carrier, it will need to be changed so that when someone calls it, the local carrier forwards calls via SIP to an address provided by Talkdesk Support such as +123456789@EXAMPLE.COM
(The below diagram is just an example and is simplified: your exact SIP endpoint will be provided to you via ticket by contacting our support team on firstname.lastname@example.org).
In order for the local carrier to forward the call correctly via SIP to Talkdesk in a secure fashion, we need to add all the Public IP Addresses used by the local carrier to our Firewall’s whitelist (maximum 2500 addresses at this time). We only require signalling IP's not media IP's where applicable.
Please do bear in mind, support boundaries will of course be different due to us not hosting these numbers on our own platform.
- The external system must support the G.711/μ-law codec and all calls must be sent using this.
- Before we can configure the endpoints we need to know all the phone numbers that will be forwarded to Talkdesk.
- We will also need to know all the Public IP Addresses that will be used by the local carrier to send calls.
We will then do the following:
- Add those IPs to our firewall.
- Generate a SIP endpoint address for each phone number.
- Provide you with these details.
You can then:
- Pass the endpoint information to your external provider for configuration.
- Should they need further information please inform us so we can assist.
If you are using Flynumber as your external SIP provider, instructions are provided here on how to setup forwarding after we have given you your SIP details.
To send calls via SIP, your provider may need our providers' IPs for both SIP and media traffic. Also, check that your SIP endpoint supports the same media codec and method for sending and receiving DTMF.
Our provider receives SIP traffic to a set of IP addresses (see below). The IP address of a request may change from call to call. You can whitelist all of our providers SIP IPs to ensure traffic can flow without any issues.
Media is transported via RTP and is accepted on ports 10000 to 20000 for RTP traffic. RTP traffic can occur on a variable set of IPs so you will need to open up your system to receive RTP from any IP address. Should you need to limit access, you must review Amazon AWS IP ranges here.
G.711/μ-law for media.
RFC2833 for receiving DTMF.
Our provider supports TLS but does does not currently validate the certificates of the remote clients. This means that you may use self-signed certs on your clients.
5060 (UDP/TCP) & 5061 (TLS)
IP Addresses for SIP traffic:
Our provider will receive SIP traffic to the IP addresses below. We strongly recommend that you whitelist all of the IPs below even though you may only see traffic from a few IPs initially. The additional IPs will be used to enhance scalability and reliability.
Please check with your external carrier hosting the numbers how many channels are permitted for each phone number. Some may have limitations on this, and this will determine how many simultaneous inbound calls you can receive.