Providers
Description
In old PSTN world they are called Trunks. These are your routes to outside world (out of your VoIP network).
Providers can be several types: Zap, SIP, IAX2, and H323. Type determines which technology is used to connect to Provider.
At the very start you have to create Providers through which you will dial out to PSTN or other VoIP networks.
Every Provider will charge you some money for calling through him (rate could be 0 also) to various destinations. And to every destination rate is different. Following this logic we need to have Tariff (or Price List) for Provider. This Tariff will tell our billing what price we pay to some destination if we are using this Provider's services. In other words - that would be our Selfcost.
Providers' configuration can by found in SETTINGS – Billing – Providers.
Here enter the name for new provider, select its type, tariff and press Create.
NOTE:
- You can't create new Provider if there are no Tariffs available for Providers.
- Provider is also able to send calls, not just receive. More info here.
Settings
General
- Name – name for provider, for informational purposes
- Technology – SIP/IAX2/ZAP/H323 – choose which one is used by your provider, e.g. in which way do you connect to it
- Tariff – the rate list provider charges you
- DTMF Mode – available options: inband, info, RFC2833, auto – choose which one your provider uses
- Location – which Localization rules set to apply to INCOMING calls COMING FROM this provider
- Automatic Number Identification - used in special case explained here: Provider with ANI
- Timeout - This setting lets set how long this provider should be dialed before giving up. Default value is 60 seconds. Minimum value is 30 seconds.
- Device ID – informational data - (not used starting from MOR 0.8)
- Call limit - explained here: Simultaneous call limitation
Authentication
For ZAP providers
- Channel – which channel (or channel group) to use on PRI/BRI/PSTN card. Channels and groups should be configured in zapata.conf.
For SIP/IAX2 providers
- Login – username if your provider asks for it
- Password – password used for authentication by your provider
- Authenticate by IP – just check the checkbox (Login/Password can be left empty)
- Make sure you put correct Hostname/IP adress!
- Register - should we register to provider
- Register extension - if provider asks to use some extension for registration (usually they don't) - put it here
CallerID
CallerID – consists of two parts: name and number. The "Number" part gets transferred by default by all technologies: GSM, PSTN, SIP etc, the "Name" part only on some. The number you see on your mobile phone when somebody is calling you is "Number" part.
- Name – the "Name" part of CallerID
- Number – the "Number" part of CallerID
NOTE: if you leave these fields empty – calls coming from this provider will have CallerID set by Provider. It should be like this almost always.
- Hostname – hostname of the provider
- Server IP – provider's IP, the value can be "dynamic", which means that provider can change its. To discover this value do ping to the provider's hostname.
- Port – port used to connect to provider (default: 5060 for SIP, 4569 for IAX2, 1720 for H323)
- Media control – canreinvite/transfer – do you want your server to stay in media path between your clients and your provider (disable if you have a lot of 1s answered calls)
- NAT – available options: yes, no, never, route. For detailed explanation about these settings please refer here.
- Qualify – how often "ask" provider about availability. More details here for SIP providers and here for IAX2 providers.
- Fast Start - H323 specific option, on or off
- h245 Tunneling - H323 specific option, on or off
Codecs
Choose the codecs your provider uses.
NOTE: When no fields are checked – that means that all codecs are available. (
E.g. settings in sip.conf or iax.conf are effective)
Allowed addresses
IP, MASK – permit traffic from these IP only. You can find detailed explanation here.
SIP Specific
- Fromuser - more details
- Fromdomain - more details
- Trustrpid - This defines whether or not Remote-Party-ID is trusted. It's defined in http://tools.ietf.org/id/draft-ietf-sip-privacy-04.txt
- Sendrpid - Defines whether a Remote-Party-ID SIP header should be sent. Defaults to no. This field is often used by wholesale VoIP providers to provide calling party identity regardless of the privacy settings (the From SIP header).
- Insecure - more details
- T.38 Support - should we support T.38 pass-through?
- SIP 302 Reinvite Support - turns on/off this SIP feature
