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This guide is meant to document the additional configurations required to configure NSG to pass the SPIROU certification test

Requires NSG 4.1.2 or greater, found at

IMPORTANT: These steps are in addition to the steps outlined on

  1. When configuring an SS7 Link:
    1. Set MTP2 "Link Type" to ETSI (default ITU92)
    2. Set MTP3 "Switch Type" to ETSI v3 (default ITU00)
    3. **note** SLC numbering starts at 1, not 0. (ie you have two links to the same APC, you will need to assign them an SLC of 1 and 2 respectively (default 0 and 1))
  2. When configuring the SS7 Linkset:
    1. Ensure that "Minimum Active Signalling Links" is equal to the number of signalling links connecting to the same APC
  3. When setting the SS7 Channel Configuration for each span:
    1. Click on "Show Advanced Options"
    2. Set ITX/TXA Auto Reply to Enable (default Disable)
    3. Ensure CPG on Progress to Disable (default Disable)
    4. Set CPG on Progress Media to Disable (default Enable)

After configuring the system through the WebUI, you will need to manually edit the following files to configure settings which are not exposed in the WebUI.  You can modify these files directly through the system through SSH, or if you prefer you can open up the "File Editor" in the WebUI.  You will not need to remove or edit any existing parameters, you will only need to add in new lines.

In the file /usr/local/nsg/conf/autoload_configs/freetdm.conf.xml:
configuration > signaling_configs > sngss7_configs > sng_isup > isup_interfaces > isup_interface block add the following lines:
      <param name="isup.t6" value="600"/>
         (in WebGui, value is in second. In this case, it needs to be 60 configured in webgui)
      <param name="isup.t9" value="1200"/>      (in WebGui, value is in second. In this case, it needs to be 120 configured in webgui)

In the file /usr/local/nsg/conf/dialplan/default.xml:
context name=from-pstn > extension name=to-sip > condition field="^(.*)$" field="destination number" block add the following lines ABOVE the bridge:
      <action data="ringback=%(1500,3500,440)" application="export"/>
      <action data="ss7_rel_loc=4" application="set"/>         (valid values are 1-15)

context name=from-sip > extension name=to-pstn > condition expression="^(.*)$" field="destination number" block add the following lines:

      <action application="export" data="sip_h_X-FreeTDM-LOC=NULL" />
      <action application="export" data="freetdm_iam_fwd_ind_isdn_access_ind=0" />
      <action application="export" data="freetdm_bearer_capability=3" />
      <action application="export" data="sip_h_X-FreeTDM-RDNIS-Presentation=0" />

Original Called Number:

If incoming IAM does not contain this IE, it’s required to add this IE to redirected outgoing IAM. This can be added to native-bridged IAM by using x-headers:



Configure these x-headers’ value in your Asterisk. See

ISDN user part indicator in ACM:

To set the backward call indicator’s isdn user part indicator value in outgoing ACM message, set the variable “acm_bi_iup” value as needed in dialplan. For example:

      <action application="export" data="acm_bi_iup=0" />


At this point all configuration changes on NSG have been made, and we need to modify our Asterisk dialplan to notify NSG when to natively bridge two channels to enable the redirection of calls.  This is needed to enable "redirection" and "portability" functionality.

On your Asterisk system, in the file /etc/asterisk/extensions.conf in the context [from-nsg] you will need to make sure you have the following line:
exten => <number>, n, SIPAddHeader(X-FreeTDM-TransUUID:${SIP_HEADER(X-FreeTDM-TransUUID)})

This will ensure that NSG properly forwards all messages on a call that has been redirected back out to the PSTN.

**note** This is not the full configuration.  You will need to create your dialplan to trigger "redirected" numbers which also modifies the Redirection X-Headers, and also set conditions for "ported" numbers which does not add those headers.  These lines are there only to enable the bridging of SS7 messages from one "side" of a redirected call to another.  For more information on the available SIP X-Headers, please visit

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