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Registering a CloudCo Partner SIP Trunk on your PBX

General Overview:

In order to setup the CloudCo Partner SIP Trunk with your PBX, there are several items that you need to be aware of. We will go into configuration with specific PBX’s later on in this article. CloudCo Partner uses a pair of IP proxies for redundancy of signaling. You will want to ensure that both of the following IP addresses can be accessed.



Keep in mind that for inbound traffic, both IP addresses need to be white-listed, as signaling could come from either proxy. You will also want to ensure that both IP addresses are configured for outbound traffic as well, in the event that one may be offline.


Bandwidth does not support REGISTRATION. In order to utilize CloudCo Partner’s SIP trunk, you will need to provide your Public Static IP Address to CloudCo Partner  that they will send calls to and in the same regard, you will need a public static address to send CloudCo Partner calls. Ultimately, your public static IP address is going to be used to authenticate with CloudCo Partner. When the registration method is sent you may see a 200 OK from CloudCo Partner, however, it is not guaranteed.


In the event that your PBX is going to be protected by a firewall, you will need to forward several ports, in order to allow for full 2-way audio as well as signaling to pass through correctly.

  • ·         UDP Port 5060 – Default port used by CloudCo Partner to allow for SIP Signaling
  • ·         UDP Ports 1024 to 64000 – Need to be open for audio traffic


CloudCo Partner SIP Trunks also have several supported attributes including DTMF, Dial Plans, Codecs, Signaling Protocol and IP Protocol.

DTMF AKA Dual-tone multi frequency signaling, is used to detect dialed digits dialed during a call and can be detected on both inbound and outbound calls. Bandwidth supports in-band or out-of-band DTMF as specified in RFC 2833. For dial plans, Bandwidth only supports the use of E.164 for inbound and outbound calls. E.164 is an international standard for calls. It can be recognized by a + followed by the country code, for the United States this would be +1. You will have to verify that your phone system supports using this method for dialing if you choose to utilize it. If you choose to use this method, Bandwidth will send all calls into the PBX with this format and will expect that calls sent back are also formatted this way.


CloudCo Partner requires that all SIP and audio traffic are delivered using UDP. TCP is not accepted or delivered by CloudCo Partner. Your UDP packets also cannot be larger 1350 bytes or else you will get a “Message Size Too Big” error that will prevent the call from completing successfully. Bandwidth also only supports 2 audio codecs, these are G.729a and G.711u.


There are also a few SIP features that are supported by CloudCo Partner, these include; Inbound-Caller-ID, Outbound Caller-ID, Call Transferring, Conference Calling and Forwarding in specific ways.

CloudCo Partner SIP uses the FROM field to present caller ID name and number. In the case that a Remote-Party ID field is included in the SIP INVITE message then that will be used for Caller ID. In either case they will need to be in a 10-digit format. P-Asserted Identity and Privacy headers are also supported.




  • Setting Up a CloudCo Partner SIP Trunk with 3CX
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  1. Steve Stoveld

  2. Posted