Gateways
- Samir Doshi
- Alpesh Kaklottar
- Tushar Devani (Unlicensed)
- Devang Nathwani (Unlicensed)
Overview
A gateway in ASTPP acts as a crucial link between the traditional Public Switched Telephone Network (PSTN) and Internet Protocol (IP) networks. It serves as a bridge facilitating the connection between the caller and the destination. Essentially, it involves a set of configurations to establish connections between upstream/downstream providers and the ASTPP switch.
Purpose
The primary purpose of configuring gateways in ASTPP is to enable seamless communication between different networks. By setting up gateways, ASTPP establishes connections with upstream and downstream providers, allowing the smooth flow of call traffic. These configurations are essential for routing calls effectively and ensuring a reliable link between the traditional telephony infrastructure and modern IP networks.
How will it work?
Carriers →Gateway
Basic Information | ||
|---|---|---|
Field | Description | |
Name | Convenient and self-explanatory name for admin to understand. | |
SIP Profile | Drop-down to select switch's SIP-Profile to bind the gateway with. In other words, if you want the gateway to connect via another network interface card or the connection between public and private interface. | |
Username | If the 'Register' set to 'True' meaning the gateway is allowed with device based access in that case configure the username value shared by up/down stream provider | |
Password | Same as above, just configure the shared password of above username here | |
Proxy | The IP(or IP:Port) or domain name of up/down stream provider, to/from where the call traffic is switching from/to ASTPP | |
Outbound-Proxy | The SIP proxy IP/Domain configuration, if any | |
Register | True | If up/down stream provider having device based access authentication |
False | If up/down stream provider having IP based access authentication | |
Caller-Id-In-From | True | If caller ID wanted to be in the "From:", which is CID type=none |
False | If caller ID not needed to be modified and pass as is, as per the generated dialplan | |
Status | Active | If the configured gateway is needed to be Active |
Inactive | If the configured gateway is not needed to be Active | |
Optional Information | ||
|---|---|---|
Field | Description | |
From-Domain | If blank, same as 'Realm' | |
From User | If blank, same as 'Username' | |
Realm | If blank, same as 'Gateway Name' | |
Extension-In-Contact | True | If 'Extension' value is blank then set Contact-User as 'Username' |
False | If Contact-User not needed to be modified and pass as is, as per the generated dialplan | |
Extension | If above is 'True', the value here will replace the Contact-User | |
Expire Seconds | If blank, 3600 seconds will be considered between Register requests | |
Reg-Transport | Which transport to use for Register | |
Contact Params | If any extra SIP parameters to send in the contact | |
Ping | To send options ping every configured seconds, failure will Unregister and/or mark it down | |
Retry-Seconds | Seconds before a retry when a failure or timeout occurs | |
Register-Proxy | Send Register via this proxy; Same as 'Proxy' if blank | |
Dialplan Variable | Additional Parameters for Gateway Configuration in Directory Push We are introducing the ability to set custom variables that can be executed within the call script when pushing the gateway configuration to the switch.
| |
Custom Dialplan Variable :
Example 1 : SIP Diversion Header for DID Call Transfers to PSTN
We will add the SIP Diversion header to the SIP INVITE when an incoming DID call is transferred to a PSTN number.
In the Gateway create/edit form, users can configure a dial plan variable, such as sip:#diversion_from_number#@sip.netmode.com.au;reason=unavailable, in the "Dialplan Variable" field. The placeholder #diversion_from_number# will be replaced by the actual DID number where the call is received.
For example, if a call is received on the DID number 109990, the dial plan variable will become: sip:109990@sip.netmode.com.au;reason=unavailable.
This header will be applied whenever a DID call is transferred to a PSTN number through any routing method, such as call forwarding, PBX features, or other types of transfers. The SIP Diversion header will be displayed in the second call leg when viewed in sngrep.
It will not apply to standard direct SIP-to-PSTN calls.
Example 2 : Set header variables by calling function & Static Singapore DID to Singapore trunk routing (Static Singapore DID Routing)
This can also be achieved using a dial plan variable called function=custom_set_variables.
Role of the Provider
The provider is responsible for managing calls by:
Identifying which provider handles the incoming call based on the DID number
Routing calls through appropriate trunks linked to that provider.
Accessing specific trunks (routes) that can be used for outgoing calls.
Determining the termination rates for the call, ensuring the correct pricing is applied.
In essence, the ASTPP ensures that calls are properly managed, routed, and billed according to their specific DID and trunks.