Difference between revisions of "Providers"

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== Description ==
== Description ==


In old PSTN world they are called Trunks. These are your routes to outside world (out of your VoIP network).
In the old PSTN world, Providers were called Trunks. Providers are your route out of your VoIP network to the outside world.


Providers can be several types: Zap, SIP, IAX2, and H323. Type determines which technology is used to connect to Provider.
Providers can be one of several types: ZAP, SIP, IAX2, and H323. The 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.
At the very start, you need to create a Provider 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.
Every Provider will charge you for calling a destination through his service (although the rate might be zero). The to every destination is different. Because of this, we need to have a Tariff (or Price List) for each Provider. This Tariff will tell our billing what price we will pay to a destination when using this Provider's services. In other words, this will be our Selfcost.


Providers' configuration can by found in '''SETTINGS – Billing – Providers'''.
Providers' configurations can by found in '''SETTINGS – Billing – Providers'''.
 
Here enter the name for new provider, select its type, tariff and press Create.


Here, enter the name for anew provider, select its type and tariff, and click Create.
NOTE:  
NOTE:  
* You can't create new Provider if there are no Tariffs available for Providers.
* You can't create new a Provider if there are no Tariffs available for Providers.
* Provider is also able to send calls, not just receive. More info [[Configure Provider which can make calls | here]].
* A Provider is also able to send calls, not just receive them. More info [[Configure Provider which can make calls | here]].


== Settings ==  
== Settings ==  
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=== General ===
=== General ===


* Name – name for provider, for informational purposes
* Name – a name for the 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
* Technology – out of SIP/IAX2/ZAP/H323, choose which technology your provider uses - that is, the way you connect to the provider.
* Tariff – the rate list provider charges you
* Tariff – the list of rates the provider charges you.
* DTMF Mode – available options: inband, info, RFC2833, auto – choose which one your provider uses
* DTMF Mode – the available options are inband, info, RFC2833, and auto. Choose the one used by your provider.
* Location – which Localization rules set to apply to INCOMING calls COMING FROM this provider
* Location – which Localization rules are set to apply to INCOMING calls COMING FROM this provider.
* Automatic Number Identification - used in special case explained here: [[Provider with ANI]]
* Automatic Number Identification - used in a 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.
* Timeout - this setting lets you set for how long this provider should be dialed before giving up. The default value is 60 seconds, and the minimum value is 30 seconds.
* Device ID – informational data - (not used starting from MOR 0.8)
* Device ID – informational data. No longer used starting from MOR 0.8.
* Call limit - explained here: [[Simultaneous call limitation]]
* Call limit - explained here: [[Simultaneous call limitation]].




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==== For ZAP providers ====
==== 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.
* Channel – which channel (or channel group) to use on a PRI/BRI/PSTN card. Channels and groups should be configured in zapata.conf.


==== For SIP/IAX2 providers ====
==== For SIP/IAX2 providers ====


* Login – username if your provider asks for it
* Login – username if your provider asks for it.
* Password – password used for authentication by your provider
* Password – password used for authentication by your provider.
* Authenticate by IP – just check the checkbox (Login/Password can be left empty)
* Authenticate by IP – just check the checkbox (Login/Password can be left empty).
** Make sure you put correct Hostname/IP adress!
** Make sure you put the correct Hostname/IP address!
* Register - should we register to provider
* Register - should we register with the provider?
* Register extension - if provider asks to use some extension for registration (usually they don't) - put it here
* Register extension - if the provider asks to use an extension for registration (usually they don't), enter it here.




=== CallerID ===
=== 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.
CallerID – consists of two parts: Name and Number. The "Number" part is transferred by default on all technologies (GSM, PSTN, SIP etc), but the "Name" part only on some. The number you see on your mobile phone when someone is calling you is the "Number" part.


* Name – the "Name" part of CallerID
* Name – the "Name" part of CallerID.
* Number – the "Number" 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
NOTE: if you leave these fields empty, calls coming from this provider will have CallerID set by the Provider. It should almost always be this way.
be like this almost always.




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=== Network related ===
=== Network related ===


* Hostname – hostname of the provider
* 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.
* Server IP – provider's IP. The value can be "dynamic", which means that provider can change its IP. 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)
* Port – port used to connect to the 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)
* Media control – canreinvite/transfer. Do you want your server to stay in the media path between your clients and your provider? Disable if you have a lot of 1-second answered calls.
* NAT – available options: yes, no, never, route. For detailed explanation about these settings please refer [http://www.voip-info.org/wiki-Asterisk+sip+nat here].
* NAT – the available options are: yes, no, never, and route. For a detailed explanation of these settings, refer to [http://www.voip-info.org/wiki-Asterisk+sip+nat here].
* Qualify – how often "ask" provider about availability. More details [http://www.voip-info.org/wiki-Asterisk+sip+qualify here] for SIP providers and [http://www.voip-info.org/wiki/view/Asterisk+iax+qualify here] for IAX2 providers.
* Qualify – how often to "ask" the provider about availability. More details [http://www.voip-info.org/wiki-Asterisk+sip+qualify here] for SIP providers and [http://www.voip-info.org/wiki/view/Asterisk+iax+qualify here] for IAX2 providers.
* Fast Start - H323 specific option, on or off
* Fast Start - an option specific to H323; it is either on or off.
* h245 Tunneling - H323 specific option, on or off
* h245 Tunneling - an option specific to H323; it is either on or off.




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Choose the codecs your provider uses.
Choose the codecs your provider uses.


NOTE: When no fields are checked – that means that all codecs are available. ([[Image:asterisk_icon.png]] E.g. settings in sip.conf or iax.conf are effective)
NOTE: When no fields are checked, all codecs are available. ([[Image:asterisk_icon.png]] For example, settings in sip.conf or iax.conf are effective).




=== Allowed addresses ===
=== Allowed addresses ===


IP, MASK – permit traffic from these IP only. You can find detailed explanation [http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+permit-deny-mask here].
IP, MASK – permit traffic from these IPs only. You can find a detailed explanation [http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+permit-deny-mask here].




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* Insecure - [http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+insecure  more details]
* Insecure - [http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+insecure  more details]
* T.38 Support - should we support T.38 pass-through?
* T.38 Support - should we support T.38 pass-through?
* SIP 302 Reinvite Support - turns on/off this SIP feature
* SIP 302 Reinvite Support - turns this SIP feature on or off.




Revision as of 11:46, 11 May 2010

Description

In the old PSTN world, Providers were called Trunks. Providers are your route out of your VoIP network to the outside world.

Providers can be one of several types: ZAP, SIP, IAX2, and H323. The type determines which technology is used to connect to Provider.

At the very start, you need to create a Provider through which you will dial out to PSTN or other VoIP networks.

Every Provider will charge you for calling a destination through his service (although the rate might be zero). The to every destination is different. Because of this, we need to have a Tariff (or Price List) for each Provider. This Tariff will tell our billing what price we will pay to a destination when using this Provider's services. In other words, this will be our Selfcost.

Providers' configurations can by found in SETTINGS – Billing – Providers.

Here, enter the name for anew provider, select its type and tariff, and click Create.

NOTE:

  • You can't create new a Provider if there are no Tariffs available for Providers.
  • A Provider is also able to send calls, not just receive them. More info here.

Settings

General

  • Name – a name for the provider, for informational purposes.
  • Technology – out of SIP/IAX2/ZAP/H323, choose which technology your provider uses - that is, the way you connect to the provider.
  • Tariff – the list of rates the provider charges you.
  • DTMF Mode – the available options are inband, info, RFC2833, and auto. Choose the one used by your provider.
  • Location – which Localization rules are set to apply to INCOMING calls COMING FROM this provider.
  • Automatic Number Identification - used in a special case explained here: Provider with ANI
  • Timeout - this setting lets you set for how long this provider should be dialed before giving up. The default value is 60 seconds, and the minimum value is 30 seconds.
  • Device ID – informational data. No longer 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 a 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 the correct Hostname/IP address!
  • Register - should we register with the provider?
  • Register extension - if the provider asks to use an extension for registration (usually they don't), enter it here.


CallerID

CallerID – consists of two parts: Name and Number. The "Number" part is transferred by default on all technologies (GSM, PSTN, SIP etc), but the "Name" part only on some. The number you see on your mobile phone when someone is calling you is the "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 the Provider. It should almost always be this way.


Network related

  • Hostname – hostname of the provider.
  • Server IP – provider's IP. The value can be "dynamic", which means that provider can change its IP. To discover this value, do ping to the provider's hostname.
  • Port – port used to connect to the provider (default: 5060 for SIP, 4569 for IAX2, 1720 for H323).
  • Media control – canreinvite/transfer. Do you want your server to stay in the media path between your clients and your provider? Disable if you have a lot of 1-second answered calls.
  • NAT – the available options are: yes, no, never, and route. For a detailed explanation of these settings, refer to here.
  • Qualify – how often to "ask" the provider about availability. More details here for SIP providers and here for IAX2 providers.
  • Fast Start - an option specific to H323; it is either on or off.
  • h245 Tunneling - an option specific to H323; it is either on or off.


Codecs

Choose the codecs your provider uses.

NOTE: When no fields are checked, all codecs are available. (Asterisk icon.png For example, settings in sip.conf or iax.conf are effective).


Allowed addresses

IP, MASK – permit traffic from these IPs only. You can find a 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 this SIP feature on or off.




See also

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