Inbound Registration is when the SIP details, are automatically generated on our side. Once created, it is up to you, to REGISTER with us, just like a typical UAC to UAS scenario.
Dial Pattern : The dial pattern is the regular expression used to match again the number you are dialling out, allowing this connection to be selected when a particular number is dialed. The broadest possible match is simply a . (a full stop). You can have multiple connections and with each connection they can match on a different regular expression. For example; if you have two connections, one to Office A where your extension numbers all in the 200 to 299 range, and another connection to Office B where your extensions are all 300 to 399, then you would set the connection to Office A with a Dial Pattern of ^2\\d{2}$ and the connection to Office B with a dial pattern of ^3\\d{2}$. You could even have a third connection that is connected to an ISP with a catch-all dial pattern (.), but then you should make sure that it has a greater Weight than the other connections.
Weight : The Weight is used if you have more than one connection. The Weight value is a numeric value to determine the order of connections when performing the dialed number lookup. The lookup is performed in a loop starting with the lowest Weight, and moving higher, and when a match is found, the test ends. Typically a “catch-all” Dial Pattern (.) should have the highest value.
Allow Source IP/Subnet : The allow source IP or Subnet is a list of IP addresses (eg “192.168.1.0”) or Subnets (”192.168.0.1/24”), that restricts access to your account when a call comes in. Each inbound call on our SBC is inspected and matched against an account, once the match is found, the related connection is loaded and this property is matched against the IP represented as the source of the message. If there is a match the call is allowed, otherwise its rejected.
Note: this option can be set to 0.0.0.0/0 (all possible IP addresses), but this should only be used with Inbound Registration’s as they are challenged with Digest Authentication. It’s recommended to always make use of this option.
Transcoding
Call Transcoding is only available on a paid account. When the call is transcoded, the session is initiated via the Registration details as it would normally, except the media is pointed to our transcoding servers. This means the INVITE will appear as peer-to-peer, but the media will flow over our network. The media will also be transcoded to DTLS (with Opus codec) from RTP (with G722 codec) or from DTLS (with Opus codec) to RTP (with G722 codec).
IP Address : This is the live IP address of the selected Transcoding Server. We allow you to select an available transcoding server so that you can open this IP to media traffic inbound and outbound on your firewall.
Port: 10000 – 65000. We suggest opening the entire UDP range greater than 10000 to and from our Transcoding Server(s).
Protocol: (UDP) Media will always be in UDP protocol but the underlying media encoding will be set according to the media transcoding options.
SBC Server
The Session Border Controller (SBC) is the public facing interface of the Siperb Network. It’s the first of 3 Proxy Servers that allows calls to flow from your network through ours and on to your devices. We allow you to choose a different SBC for each of your own connections. Once a connection is setup on a SBC, it can be changed by selecting another SBC from the list. The choice of SBC is important for you to understand, as it determines the source or destination of the signalling traffic. This will be important when you configure your Asterisk (or other) PBX.
Host Address : You can choose the SBC server by selecting one from the dropdown above. This list may expand over time.
IP Address : This is the live IP address of the selected SBC. You may want to make note of this to allow SIP traffic in and out of your firewall.
Note: Never send a call to us as an IP address, always send your INVITE as a full domain INVITE <sip:123456789@eu-west-1-sbc-1.siperb.com> SIP/2.0.
Port: (5060) We will send SIP messages on port 5060, and you can send SIP messages to us on port 5060.
Protocol: (UDP) We will send SIP messages on an UDP protocol, and you can send SIP messages to us on an UDP protocol.
Registration
Registration is an important part of your connection. With Inbound Registration, the registration details below resemble the details that you will need to copy and insert into your Asterisk (or other) PBX. Typically this would be associated with the registration details that Asterisk would need to use in order to establish a registration with us. This process allows us to learn your location and port, allowing us to send calls to you. It would be up to you to identify and authorise calls originating from us.
Note: Inbound calls, from you to us, will always be challenged with Digest Authentication, however as an added precaution you should also set an Allow Source IP or Subnet.
Username : Copy the above username for the auth part of the Asterisk Registration on your server.
Password : Copy the above password for the auth part of the Asterisk Registration on your server.
[ ] Reset Username and Password (optional) By selecting this option, you will reset the current username and password. You would only need to perform this task if your username and password have not yet been created or would like to reset the details for security reasons.