Routing calls on your own number via SIP to babelforce

If you have a voice telephone number and you want to route the calls to babelforce, all you need to do is the following:

•  Set up forward to SIP with your provider of the number
• Send the calls to this URI "sip:1234567893434@trunk.{yourEnvironment}.babelforce.com" (please contact support@babelforce.com if you are unsure about the environment), where "1234567893434" must be replaced with your number in international format (with the country code in the beginning, following the E.164 standard). With some providers you can leave out the leading "sip:" in the URI field, because the provider adds it automatically in the background.
• Write to support@babelforce.com and tell us the number, e.g. "1234567893434"
• We also need to know the name of your phone number provider and the IP addresses from which your carrier is sending SIP and RTP traffic to us so that we can whitelist it in our firewall.

That's about it. If you have other specific SIP trunking needs or require a different technical interface for SIP calls, then just write to us using the Support tab in the Help Center.

A few important things to note:

• We transmit all calls using the highest throughput rate possible. So generally for inbound calls we will expect codec G711a. This is to ensure that your calls are transmitted at 64kbps thereby giving you the possibility to achieve the best possible quality (assuming of course that your providers, network and devices are optimized for VoIP).
• We operate an international platform, so your inbound providers should send the callers number in full international format, e.g. a call from the UK should present the caller number to us like 44xxxxxxxxxxxxx, from Spain 34xxxxxxxxx, from USA 1xxxxxxxxxx, etc.
• It is the responsibility of you and your provider to deliver SIP calls with the correct international format (ITU E.164 International with country code). The data delivered via SIP must meet the requirements or the SIP calls will be rejected.
• If your provider is already registered on babelforce as a connection, then we will just have to verify the IP ranges from which SIP signalling and RTP media traffic will come. If on the other hand your provider is not yet connected, then we will need to add a one-way peer definition and also whitelist IP addresses.
• Note that direct SIP interconnections (peer-to-peer) and SIP trunking is also supported by babelforce, these require a custom configuration and are dependent on babelforce validating and accepting the provider in question.
• Finally, you must have a live account and existing contract with babelforce, in order to send call traffic into an account using SIP forwarding or call forwarding or any other custom method.