Ejoin VoIP Gateway SIP Protocol
About ejointech VoIP And SMS gateway device, SIP protocol information, Running Parameters.
Running Parameters
Figure 3.3-1 Running Parameters
Items | Description |
Protocol Mode | It is the same as that in Basic Settings. The modification here also apply to Basic Settings page. |
Encryption Method | It is the same as that in Basic Settings. |
SIP Server | It is the same as that in Basic Settings. |
SIP Server Port | It is the same as that in Basic Settings. |
Primary Proxy IP | Proxy server will receive requests from client, and make the signaling and media streams are able to penetrate the firewall. It is usually used when gateway can’t registered with the softswitch because of network blockade. |
Proxy Port | The proxy server port. Ejoin default proxy port is 25600. |
Secondary Proxy IP | It is the same as primary proxy, don’t need set it. |
Expiration Period | Gateway will send a register request to the softswitch during every expiration period. |
Multiple Port Support | Disabled: all 16 ports will be used one SIP account. Enabled: all 16 ports SIP account will be separate. |
Use Phone Number | If the username is not the same with user id, enable it. |
Receive All Calls | Disabled: only the SIP server address which is type in basic settings or phone book can send traffic to this gateway. Enabled: traffic from any server can send traffic to this gateway (same LAN or both gateway and server have a public IP). It’s dangerous when eabled, hackers may send traffic to the gateway then steal SIM balance. |
Drop Account Prefix | If it is enabled, it will remove the account prefix presented in callee number. |
Auto Resp 183 | If it is enabled, gateway will send 183-Session-Progress immediatey for a incoming INVITE. |
Route By From | If it is enabled, gateway will only accept the call whose “From” header is matched. Note: if the gateway is just used as call termination, please disable it. |
No Line Code | Gateway will send this SIP code as response to SIP server when no available line. |
Custom User Agent | The User Agent header which is used in SIP message. |
STUN
STUN (Simple Traversal of UDP through NAT) is a protocol for assisting devices behind a NAT firewall or router with their packet routing. If you have the STUN server, enable STUN support, fill the server IP and port (default port is 3478), then it will work.
Figure 3.3-2 STUN Settings
MNP
Figure 3.3-3 MNP Settings
Items | Description |
MNP support | Mobile Number Portability (MNP) enables mobile telephone users to retain their mobile telephone numbers when changing from one mobile network operator to another. |
Select Order | When the traffic send to the gateway, it can select ascending order, descending order or random ports. |
Route | There are two choices of route: 1. Route calls after manipulation. 2. Route calls before manipulation. Note: route calls by allow prefix, callee number prefix manipulation by inward translation. |
Server URL | MNP server address |
Username | MNP server username |
Password | Password of the username |
SIP Accounts Figure
3.3-4 SIP Accounts Settings
Items | Description |
Allowed Prefix | Intelligent routing, gateway will route calls by the allowed prefix. for example: channel 1 is with prefix 070 and 075, this channel will only accept the calls with prefix 070 and 075, others will not be routed to this channel. If allowed prefix is blank, it can accept any calls. If all prefixes don’t match, the call will be rejected. |
Phone Number | When enable route by from, the channel will only accept the call which caller ID is input in phone number. |
Account | SIP registration account. |
Password | The password of SIP registration account. |
Status | The status of registration. When gateway is registered with softswitch, it will show ready. |