SIP providers

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Contents

General settings

In Pre-configuration, you may choose from a variety of providers in order to use the predefined settings provided by AskoziaPBX. If your provider is not included in that list, the configuration is to be done manually starting with the Name of the provider and the Host. The Host is either a URL or an IP address. Username and Password are required next to complete the basic provider setup.

From the Language list, the language for the audio prompts for this account can be chosen. The specified Public Number will be read back to the caller when reaching voicemail. By default, the account's username is used if it is numeric. If the username is not numeric, the internal extension the call was routed to will be read back. It is advantageous to enter your entire call number in this field to make sure the caller is not just told the extension.

General settings for SIP providers


Call routing

The Call Routing settings are identical for all provider types. Therefore, they are explained in a separate section and not individually for each provider type.


Regional settings

The regional settings are explained on a separate page.


Caller IDs

The Caller ID Options are identical for all provider types. Therefore, they are explained in a separate section and not individually for each provider type.


Codecs

AskoziaPBX supports a variety of free and licensed Codecs for both audio and video. Codecs are explained in detail in Help for Integrators for experienced users.


Advanced options

Advanced Options are additional parameters, required by some providers. For most users, these settings are not required. The following advanced options are available for SIP providers.

  • Multiple Hosts: If this provider has multiple servers to forward calls to you, specify a host for each server.
Advanced SIP provider settings, part 1, multiple hosts
  • With Blacklists and Greylists you can prevent specific numbers from calling in. Blacklisted numbers receive the busy signal immediately, grey listed numbers are forwarded to the voicemail of the extension the caller tried to reach. Of course, this only works if voicemail for this phone was configured. Specify one black/greylisted number per line. Select Blacklist anonymous calls or Greylist anonymous calls if you want all calls with withheld number to be black/greylisted.
Advanced SIP provider settings, part 2, blacklist and greylist
  • Add PAI: Add P-Asserted-Identity header field as defined in RFC 3325.
  • Add PPI: Add P-Preferred-Identity header field as defined in RFC 3325.
  • Add RPI: Add remote-party-identity header field
Advanced SIP provider settings, part 3, header fields
  • Public Access: Allow this number to be reachable over the Internet. This option is required if the IP of your provider is changing with every call.
  • Register Separately: Register every incoming number separately. This is necessary when you don't have a SIP trunk and don't want to add a provider for every single number.
  • T.38 capability: If supported by the provider, T.38 mode should be used. T.38 is used to transfer faxes over data networks in real-time. In order to do so, T.38 converts the analog fax signal into an image and transfers it to a T.38 compatible device over the network. If this device is an IP phone system or a gateway, it can transform the signal back from T.38 into an analog signal and send it via PSTN (the public telephone network) to an analog fax machine.
Advanced SIP provider settings, part 4
  • SIP TO as DDI: Use SIP TO-Header as incoming DDI.
  • SIP CPI as DDI: Use P-Called-Party-Identity header as incoming DDI.
  • SIP Permit/Deny: activate permit/deny mask for SIP traffic.
    • SIP Deny Network: Limit SIP traffic to and from this peer to a certain IP or network. Set to '0.0.0.0 / 0.0.0.0' to deny everything.
    • SIP Permit Network: Limit SIP traffic to and from this peer to a certain IP or network. Set to '0.0.0.0 / 0.0.0.0' to permit everything.
  • Account code: This code is generated individually for each account. Usually there is no need to change it.
  • Registration: In some cases a registration is not permitted by a provider. Select Do not register with this provider if this applies to your provider. In most cases, AskoziaPBX is able to do the formatting of the registration correctly. Just a few providers expect a special formatting of the registration string. You can specify a registration string in the same format Asterisk uses in the text box.
Advanced SIP provider settings, part 5
  • Qualify Frequency: The peer will be contacted this often to check its availability. The default values is 60 seconds, i.e. the peer would be contacted every 60 seconds.
  • Qualify: Sets how often the provider is contacted to check its availability. Choose a short interval if your firewall has a short timeout for UDP sessions (User Datagram Protocol). SIP is using UDP. The unit of time is seconds.
  • NAT: If your network is using Network Address Translation (NAT) and you have problems with one-way audio you might have to change the NAT settings because different factors (providers settings, the SIP phone or router being used, etc.) influence these settings. If you use AskoziaPBX in a public network you might want to disable NAT completely. You can choose between different settings: always use NAT mode, only use NAT mode according to RFC3581 (rport), never attempt NAT mode or RFC3581 support or assume NAT mode but do not send rport.
  • DTMF-Mode: There are different standards used to transmit DTMF (Dual-tone multi-frequency) to SIP providers. Try a different standard when keypresses are not transferred correctly. inband transfers the keystrokes as tones. Make sure the audio codec is of high enough quality. auto, rfc and info transfer the keystrokes within the SIP coding. More about this in Codecs.
  • Session Timer: SIP Session-Timers provide an end-to-end keep-alive mechanism for active SIP sessions.
  • Default Expiry: Defines the default duration in seconds of incoming and outgoing registration.
Advanced SIP provider settings, part 6
  • Authorization User: Some Providers require a separate username for authorization.
  • From User: Some SIP providers differentiate between the username for authorization and the actual username. This can be used in conjunction with outgoing Caller ID specific settings. If this is the case with your provider, enter the correct value here.
  • Disable From User: Activating this option disables the fromuser field in sip.conf. By default, the username of this account is used.
  • From Domain: By default the standard domain name is used. If another name should be used enter it in the text box.
Advanced SIP provider settings, part 7

Further options can be activated by clicking on Expert Options shown in the image below. First, you should save your configurations by clicking on the Save button. A click on Expert Options leads to the GUI options within the Miscellaneous section of the Advanced chapter, where the Enable Expert Options checkbox can be activated.

Expert provider settings notification

The following additional options are now available. These options should only be used by experienced users and only if they are actually required.

  • Manual Dialplan Global: Changes for the global dialplan can be entered here, which then appear in the top of extensions.conf.
  • Manual Dialplan Incoming: Changes for the incoming dialplan can be entered here, which then appear in the extensions.conf in the context of this configuration.
  • Manual Dialplan Outgoing: Changes for the outgoing dialplan can be entered here, which then appear in the extensions.conf in the context of this configuration. As an example, if you want to globally block numbers, the following code can be entered. In this case, all numbers, beginning with 0900 would be blocked.
ExecIf($["${REGEX("^0900[0-9]+" ${EXTENSION_FAILOVER})}" = "1"]?Hangup()) 
  • Manual Attributes: Here, you can directly change Asterisk's configuration files. Manual key-value pairs can be entered in addition to the generated configuration (i.e. configoption=value). These settings will be appended to this item's Asterisk configuration file or context.
Advanced SIP provider settings, part 8, manual attributes

More information about manual attributes can be found in the Manual Attributes section of the Help for Integrators chapter.

Click Save to finish the configuration.

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