PSTN2SIP script handles incoming PSTN calls to your gateway, authorizes them in the billing, sends the outgoing call to the SIP UA, and disconnects the call when credit time is exceeded.

Allows:

• Authentication/authorization by DNIS
• Use flexible user-name translation rules (e.g. carrier reports DNIS in the format '0296355111', but we should authenticate '4202963555111')
• Choose which phone number should be used for the outgoing call - original DNIS) or the one after translation rule was applied (account ID)
• Use account's redirect number as a destination number

Requires IOS 12.2(11)T or later.

The price of the PSTN2SIP script is $499. For more information about PSTN2SIP script please contact our sales team.

Configuration parameters.

Specified as a VSA parameters, in a form of call application voice

• authenticate-by Specifies what shall be used as a User-Name for the authentication requests. Possible values are:
  • ani - Calling-Station-Id (ANI)
  • dnis - Called-Station-Id (DNIS)
  • fixed - Fixed string, provided as "user-name" parameter
• method-list-name Specifies which authentication methods should be used for authentication. h323 by default.
• update-accounting Whether script should attempt to update information in the accounting requests. aaa accounting update is not available in older IOSes, so you might need to turn this feature off if you are using one. Possible values on and off, by default is on.
• password Password to be used for authentication queries. By default is cisco.
• skip-password PortaBilling specific. In case of the incoming call from PSTN to SIP device we might not know the valid password for the account. Nevertheless we need to do the rest of the authentication. If this switch is turned on, scripts sends the special attribute which instructs PortaBilling to skip the password check (so any supplied password matches), but do the rest of the authentication
• warning-time When user should be warned by beep before the maximum allowed sesion duration. Default is 60 seconds.
• translate Allows to apply translation rules to the username, used for authentication. This allows to change username depending on the situation without actually modifying application source, just by using CLI. is a regexp in a format /pattern/replace-with/ (in the TCL regexp syntax. For example to add 1 in from of the number wil be done with /^.+/1&/
• user-name Specifies fixed string to be used as a User-Name for authentication.
• redirect Use the account's redirect number as a destination number instead of DNIS
• authorize Perform call authorization. By default is "no".
• dial-account-id Use the account ID as a outgoing phone number. So, if DNIS is 4257891105, account ID after the translation rules is 14257891105, dial 14257891105. By default is "no".

Configuration example

! Simple case - forward calls to the SIP device
call application voice pstn2sip flash:pstn2sip.tcl
call application voice pstn2sip skip-password yes
call application voice pstn2sip authenticate-by dnis

! Similar to the above, but translate DNIS from NANP to E.164 
! For example: 4257891105 -> 14257891105
call application voice pstn2sip_tr flash:pstn2sip.tcl
call application voice pstn2sip_tr skip-password yes
call application voice pstn2sip_tr authenticate-by dnis
call application voice pstn2sip_tr translate "/^/1/"

! Same as above, but authorize the call and disconnect
! when the maximum call duration is reached
call application voice pstn2sip_authz flash:pstn2sip.tcl
call application voice pstn2sip_authz skip-password yes
call application voice pstn2sip_authz authenticate-by dnis
call application voice pstn2sip_authz authorize yes
call application voice pstn2sip_authz translate "/^/1/"
call application voice pstn2sip_authz dial-account-id yes

! Redirect call to the account's redirect number
call application voice pstn2sip_redirect flash:pstn2sip.tcl
call application voice pstn2sip_redirect skip-password yes
call application voice pstn2sip_redirect authenticate-by dnis
call application voice pstn2sip_redirect authorize yes
call application voice pstn2sip_redirect redirect yes
!
 dial-peer voice 1 pots
  application pstn2sip
  incoming called-number 1202.T
!
 dial-peer voice 2 pots
  application pstn2sip_tr
  incoming called-number 206.T
!
 dial-peer voice 3 pots
  application pstn2sip_authz
  incoming called-number 800.T
!
 dial-peer voice 4 pots
  application pstn2sip_redirect
  incoming called-number 1866.T
!
!