This incidence shows a SIP message flow, where 200 OK and ACK are sent back and forth several times.

NOTE:

-You must replace ‹X.X.X.X› in the template with the IP address for Side #2.

-You must replace ‹MySharedKey› in the template with text string as the initial shared key

-You must replace ‹nnnn› in the template with unique number from 1-99999 like 12345

-You must purchase the appropriate software license for your SmartNode to support VPN service. To see the current licenses active for your SmartNode, execute the "show license" command.

The command sequence below defines a VPN Tunnel that uses DES to route traffic from Side #1 (192.168.1.0) to Side #2 (192.168.2.0).

NOTE: You must replace X.X.X.X in the template with the IP address for Side #2.

NOTE: You must purchase the appropriate software license for your SmartNode to support VPN service. To see the current licenses active for your SmartNode, execute the "show license" command.

Command Sequence to copy, edit and paste

profile ipsec-transform DES
esp-encryption des-cbc 64

profile ipsec-policy-manual VPN_DES
use profile ipsec-transform DES
session-key inbound esp-encryption A242353012531297
session-key outbound esp-encryption B242353012531297
spi inbound esp 1111
spi outbound esp 2222

# SET IP ADDRESS of SIDE #2
peer X.X.X.X
mode tunnel

profile acl VPN_Out
permit ip 192.168.1.0 0.0.0.255 192.168.2.0 0.0.0.255 ipsec-policy VPN_DES
permit ip any any

profile acl VPN_In
permit esp any any
permit ah any any
permit ip 192.168.2.0 0.0.255.255 192.168.1.0 0.0.0.255
permit ip any any

context ip
interface eth0
use profile acl VPN_In in
use profile acl VPN_Out out

context ip router
route 192.168.2.0 255.255.255.0 eth0

Command Sequence to copy, edit and paste

profile ipsec-transform DES
esp-encryption des-cbc 64

profile ipsec-policy-manual VPN_DES
use profile ipsec-transform DES
session-key inbound esp-encryption B242353012531297
session-key outbound esp-encryption A242353012531297
spi inbound esp 2222
spi outbound esp 1111

# SET IP ADDRESS of SIDE #1
peer X.X.X.X
mode tunnel

profile acl VPN_Out
permit ip 192.168.2.0 0.0.0.255 192.168.1.0 0.0.0.255 ipsec-policy VPN_DES
permit ip any any

profile acl VPN_In
permit esp any any
permit ah any any
permit ip 192.168.1.0 0.0.255.255 192.168.2.0 0.0.0.255
permit ip any any

context ip
interface eth0
use profile acl VPN_In in
use profile acl VPN_Out out

context ip router
route 192.168.1.0 255.255.255.0 eth0

This happens when the SmartNode involved is B side of the connection.
Looking at a debug log of such a call, it can be seen that several 200 OK and ACK messages are sent back and forth.
This is becuase the other device the SmartNode is communicating with, does not follow the SIP standard as per RFC3261.
See the following explanation;

The reason why the ACK for the 200 Ok from the SmartNode is not accepted, is due to an invalid branch ID in the via header.
Based on the description in RFC 3261, the branch ID for a 2xx response has to be different than in the former messages.
You will find this in section 8.1.1.7, in the RFC 3261.

This template can be used to register a fxs gateway with at Asterisk PBX.

How to force the SmartNode entering the bootloader/u-boot mode

While connected with your computer to the console interface, perform a restart of the SmartNode. The following will appear on the console
Hit any key to stop autoboot:
When doing so, the boot process will be aborted and the SmartNode remains in u-boot mode. 

• Download the required .bin file at

• Image for SN4660 and SN4670

• Image for SN4970, SN4980 and SN4990

1. Erase the potentially stored evironment variables from u-boot, to get the default behavior:

1. Set the IP of the SmartNode: setenv ipaddr <device ip address>
2. Set the netmask for the SmartNode: setenv netmask <device netmask>
3. Set the TFTP server IP: setenv serverip <tftp server ip address>
4. Specify the right folder of your TFTP server where the image file is located: setenv tftproot <path>
   The SmartNode will automatically select the image.bin 
5. Type in: run getimage
   wait for the image to be downloaded
   
6. Type in: reset

Now the SmartNode will perform a reboot. Afterwards it will boot up with the regular firmware.

This example includes a basic configuration for the SmartNode SN411X/FXS to be used as an FXS gateway in combination with a Cisco Call Manager. The configuration contains variables in brackets <>. Please Replace the variables including the brackets. Unused lines or variables have to be commented out or deleted

This cofiguration note gives an overview about configuring the SN411x with Cisco Unified Communications Manager (CUCM).

G723 and G729
 

On some of the SmartNodes, only one low bitrate codec is supported at the time, systemwide.
There the following setting is required to make the G729 codec work.

By default the G723 codec is systemwide enabled.

The following SmartNodes DO NOT require this setting, since they can handle both low bitrate codec at the same time in the system.

SN4940 / SN4950 / SN4960
SN4970 / SN4980 / SN4990
SN4660 / SN4670
SN4120/2V
SN-DTA/2V

Gateway/Routers will also route between two dissimilar network types, but they also regulate traffic between similar networks, in this case the TCP/IP network.  These devices will need to have at least two Ethernet ports so that the IP traffic can be routed from one network to another.

Most companies already have a router in place for the IP traffic, so they will need only the Gateway feature to convert their old Telephony equipment to IP (SIP) and vice versa.

http://www.3cx.com/blog/voip-howto/patton-fxs-gateway-at/



To configure a Trinity SmartNode FXS Gateway within 3CX, please visit the link below.

https://www.patton.com/support/kb_art.asp?art=478&p=0

First of all make sure an ACL is configured and enabled in the corresponding IP interface. In the ACL make sure that only known hosts and IP networks can access the SmartNode from outside to the well known ports.

For SIP calls coming from untrusted peers, the "trust remote" feature should be enabled.

Protecting the SmartNode from SIP DoS attacks, the SIP Peer Flood Blocking can be used.

How all this can be configured you can find in the software config guide as follows:

http://www.patton.com/manuals/SCG-r63.pdf

ACL - see page 290

Trust remote - see page 501

SIP peer flood blocking- see page 418

Note: To activate a configuration, you must be the Root user or have a user access level of 0.

To activate a system configuration:

1. Select Systems from the navigation panel.

2. Click Edit next to the system you wish to activate,in order to access the System Configuration screen.

3. Activate the configuration:

• Enter a name for the system
• Under Activate Configuration:
  • Select a Configuration on which to run the system
  • Click Activate

Note: Make certain that Lock Active Configuration box (located at the top of the page) is 'unchecked. Failure to do so will prevent the configuration from being saved.

Verify that an activation confirmation message is displayed. The system will remind you to back up your database when you are done configuring (at top of screen, in yellow):

• T1/E1/J1 interfaces
• DS3 interfaces
• SONET/SDH interfaces

To create a new line interface:

1- Select Tdm Interfaces-->Line Interfaces from the navigation panel

2- Click Create New Line Interface to create a single interface ,or Create Multiple Line Interfaces to create many interfaces at once

3- Create the new line interface:

• Enter a name for the interface
• Set a type for the interface
• Set a local index for the line
• Select a length and an encoding scheme
• Click Create

4- Verify that the "Line interface was successfully created" message is displayed

The E1/J1/T1 line services are also known as trunks or spans.

To create a new line service:

1- Select Tdm Interfaces-->Line Interfaces from the navigation panel

2- Click Edit next to the line service for which you wish to create a line service

3- In the following window, click Create New Child Line Service (also, a number can be created at once by selecting Create Multiple Child Line Service)

4- Create the new Child Line Service:

• Enter a name for the new line
• Select an available local index, from the drop-down box
• Select appropriate framing and loopback modes
• Click Create

5- Navigate back to the line interface page, by following link displayed in the upper-left corner of the screen. The new line service should be displayed in the Line Services list

Step 1: Create different MTP3 Point Code for signaling and for voice Traffic

Click Point Codes in the navigation panel

Click Create New Point Code

Configure the new point code:

   Enter a name for the point code

   Select a point code type (select type-specific parameters)

   Select a point code length (14, 16, 24 bits)

   Enter a point code

   Click Create Repeat above steps for creating a voice traffic PC

Step 2: Create different MTP3 Routes for signaling and voice traffic

Click Create New Mtp3 Route in the MTP3 network configuration window:

Configure the new MTP3 route:

   Enter a name for the route

   Select a point code for the route

   Select a point type

   Check the Route to Adjacent box, if this route is directed to an adjacent network

   Click Create

Repeat above steps for voice traffic MTP3 Route.

Associate the newly created routes with your MTP3 linkset:

   Select a linkset from the list provided at the bottom of the route configuration window

   Click the "<<" button to associate the linkset with the newly created route

This example includes a basic configuration for the SmartNode SN4118/JS to be used as a FXS gateway in combination with two Cisco Call Manager. The configuration contains variables in brackets <>. Please Replace the variables including the brackets. Unused lines or variables have to be commented out or deleted

This configuration note gives an overview about configuring the SN4118/JS with Cisco Unified Communications Manager (CUCM).

Commande Sequence to copy, edit and paste

#----------------------------------------------------------------#
# #
# SN4118/JS/EUI #
# R6.X #
# Handmade configuration file #
# #
# Cisco Call Manager #
# #
# This example includes a basic configuration for the SmartNode #
# SN4118/JS in combination with two Cisco Call Manager with #
# SIP authentication and registration. #
# This example can be used for other SmartNodes too. The only #
# thing which needs to be adjusted, are the ports and interfaces #
# as far as required. #
# #
# Author: Thomas Tlapak #
# Patton-Inalp Neworks #
# CH-3172 Niederwangen #
# www.patton.com; support@patton.com #
# #
# Date: 11/06/2014 #
# #
#----------------------------------------------------------------#

cli version 3.20
clock local default-offset +00:00
timer PROVISIONING now + 3 minutes "provisioning execute PF_PROVISIONING_CONFIG"
webserver port 80 language en
sntp-client
sntp-client server primary pool.ntp.org port 123 version 4

system

ic voice 0
low-bitrate-codec g729

profile ppp default

profile tone-set default

profile voip default
codec 1 g711alaw64k rx-length 20 tx-length 20
codec 2 g711ulaw64k rx-length 20 tx-length 20

profile pstn default

profile ringing-cadence default
play 1 1000
pause 2 4000

profile sip default
no autonomous-transitioning

profile aaa default
method 1 local
method 2 none

profile provisioning PF_PROVISIONING_CONFIG
destination configuration
location 1 http://redirect.patton.com/$(system.mac);mac=$(system.mac);serial=$(system.serial);hwMajor=$(system.hw.major);hwMinor=$(system.hw.minor);swMajor=$(system.sw.major);swMinor=$(system.sw.minor);swDate=$(system.sw.date);productName=$(system.product.name);cliMajor=$(cli.major);cliMinor=$(cli.minor);osName=$(cli.major>=4|Trinity|SmartWare);subDirTrinity=$(cli.major>=4|/Trinity);subDirSmartWare=$(cli.major<4|/SmartWare);dhcp66=$(dhcp.66);dhcp67=$(dhcp.67)
location 2 $(dhcp.66)
location 3 $(dhcp.66)/$(system.mac).cfg
location 4 http://$(dhcp.66)/$(dhcp.67)
location 5 http://$(dhcp.66)/$(system.mac).cfg
location 6 tftp://$(dhcp.66)/$(dhcp.67)
location 7 tftp://$(dhcp.66)/$(system.mac).cfg
activation reload immediate

context ip router

interface eth0
ipaddress dhcp
tcp adjust-mss rx mtu
tcp adjust-mss tx mtu

context ip router

context cs switch

routing-table called-e164 RT_TO_SIP_00
route .T dest-interface IF_SIP

routing-table called-e164 RT_TO_SIP_01
route .T dest-interface IF_SIP_01

routing-table called-e164 RT_TO_SIP_02
route .T dest-interface IF_SIP_02

routing-table called-e164 RT_TO_SIP_03
route .T dest-interface IF_SIP_03

routing-table called-e164 RT_TO_SIP_04
route .T dest-interface IF_SIP_04

routing-table called-e164 RT_TO_SIP_05
route .T dest-interface IF_SIP_05

routing-table called-e164 RT_TO_SIP_06
route .T dest-interface IF_SIP_06

routing-table called-e164 RT_TO_SIP_07
route .T dest-interface IF_SIP_07

routing-table called-e164 RT_TO_FXS
route <username 2> dest-interface IF_FXS_01
route <username 3> dest-interface IF_FXS_02
route <username 4> dest-interface IF_FXS_03
route <username 5> dest-interface IF_FXS_04
route <username 6> dest-interface IF_FXS_05
route <username 7> dest-interface IF_FXS_06
route <username 8> dest-interface IF_FXS_07
route <username 1> dest-interface IF_FXS_00

interface sip IF_SIP
bind context sip-gateway GW_SIP_00
route call dest-table RT_TO_FXS
remote <IP of the callmanager>
local <IP of the callmanager>

interface sip IF_SIP_01
bind context sip-gateway GW_SIP_01
route call dest-table RT_TO_FXS
remote <IP of the callmanager>
local <IP of the callmanager>

interface sip IF_SIP_02
bind context sip-gateway GW_SIP_02
route call dest-table RT_TO_FXS
remote <IP of the callmanager>
local <IP of the callmanager>

interface sip IF_SIP_03
bind context sip-gateway GW_SIP_03
route call dest-table RT_TO_FXS
remote <IP of the callmanager>
local <IP of the callmanager>

interface sip IF_SIP_04
bind context sip-gateway GW_SIP_04
route call dest-table RT_TO_FXS
remote <IP of the callmanager>
local <IP of the callmanager>

interface sip IF_SIP_05
bind context sip-gateway GW_SIP_05
route call dest-table RT_TO_FXS
remote <IP of the callmanager>
local <IP of the callmanager>

interface sip IF_SIP_06
bind context sip-gateway GW_SIP_06
route call dest-table RT_TO_FXS
remote <IP of the callmanager>
local <IP of the callmanager>

interface sip IF_SIP_07
bind context sip-gateway GW_SIP_07
route call dest-table RT_TO_FXS
remote <IP of the callmanager>
local <IP of the callmanager>

interface fxs IF_FXS_00
route call dest-table RT_TO_SIP_00
subscriber-number <username 1>

interface fxs IF_FXS_01
route call dest-table RT_TO_SIP_01
subscriber-number <username 2>

interface fxs IF_FXS_02
route call dest-table RT_TO_SIP_02
subscriber-number <username 3>

interface fxs IF_FXS_03
route call dest-table RT_TO_SIP_03
subscriber-number <username 4>

interface fxs IF_FXS_04
route call dest-table RT_TO_SIP_04
subscriber-number <username 5>

interface fxs IF_FXS_05
route call dest-table RT_TO_SIP_05
subscriber-number <username 6>

interface fxs IF_FXS_06
route call dest-table RT_TO_SIP_06
subscriber-number <username 7>

interface fxs IF_FXS_07
route call dest-table RT_TO_SIP_07
subscriber-number <username 8>

context cs switch
no shutdown

authentication-service AUTH_SRV_00
username <username 1> password <password>

authentication-service AUTH_SRV_01
username <username 2> password <password>

authentication-service AUTH_SRV_02
username <username 3> password <password>

authentication-service AUTH_SRV_03
username <username 4> password <password>

authentication-service AUTH_SRV_04
username <username 5> password <password>

authentication-service AUTH_SRV_05
username <username 6> password <password>

authentication-service AUTH_SRV_06
username <username 7> password <password>

authentication-service AUTH_SRV_07
username <username 8> password <password>

location-service LOC_SRV_00
domain 1 <IP of the callmanager> 5060

identity-group default
identity <username 1>

authentication outbound
authenticate 1 authentication-service AUTH_SRV_00 username <username 1>

registration outbound
registrar <IP of the callmanager>
lifetime 300
register auto
retry-timeout on-system-error 10
retry-timeout on-client-error 10
retry-timeout on-server-error 10

location-service LOC_SRV_01
domain 1 <IP of the callmanager> 5060

identity <username 2>

authentication outbound
authenticate 1 authentication-service AUTH_SRV_01 username <username 2>

registration outbound
registrar <IP of the callmanager>
lifetime 300
register auto
retry-timeout on-system-error 10
retry-timeout on-client-error 10
retry-timeout on-server-error 10

location-service LOC_SRV_02
domain 1 <IP of the callmanager> 5060

identity <username 3>

authentication outbound
authenticate 1 authentication-service AUTH_SRV_02 username <username 3>

registration outbound
registrar <IP of the callmanager>
lifetime 300
register auto
retry-timeout on-system-error 10
retry-timeout on-client-error 10
retry-timeout on-server-error 10

location-service LOC_SRV_03
domain 1 <IP of the callmanager> 5060

identity <username 4>

authentication outbound
authenticate 1 authentication-service AUTH_SRV_03 username <username 4>

registration outbound
registrar <IP of the callmanager>
lifetime 300
register auto
retry-timeout on-system-error 10
retry-timeout on-client-error 10
retry-timeout on-server-error 10

location-service LOC_SRV_04
domain 1 <IP of the callmanager> 5060

identity <username 5>

authentication outbound
authenticate 1 authentication-service AUTH_SRV_04 username <username 5>

registration outbound
registrar <IP of the callmanager>
lifetime 300
register auto
retry-timeout on-system-error 10
retry-timeout on-client-error 10
retry-timeout on-server-error 10

location-service LOC_SRV_05
domain 1 <IP of the callmanager> 5060

identity <username 6>

authentication outbound
authenticate 1 authentication-service AUTH_SRV_05 username <username 6>

registration outbound
registrar <IP of the callmanager>
lifetime 300
register auto
retry-timeout on-system-error 10
retry-timeout on-client-error 10
retry-timeout on-server-error 10

location-service LOC_SRV_06
domain 1 <IP of the callmanager> 5060

identity <username 7>

authentication outbound
authenticate 1 authentication-service AUTH_SRV_06 username <username 7>

registration outbound
registrar <IP of the callmanager>
lifetime 300
register auto
retry-timeout on-system-error 10
retry-timeout on-client-error 10
retry-timeout on-server-error 10

location-service LOC_SRV_07
domain 1 <IP of the callmanager> 5060

identity <username 8>

authentication outbound
authenticate 1 authentication-service AUTH_SRV_07 username <username 8>

registration outbound
registrar <IP of the callmanager>
lifetime 300
register auto
retry-timeout on-system-error 10
retry-timeout on-client-error 10
retry-timeout on-server-error 10

context sip-gateway GW_SIP_00

interface IF_GW_SIP
bind interface eth0 context router port 5060

context sip-gateway GW_SIP_00
bind location-service LOC_SRV_00
no shutdown

context sip-gateway GW_SIP_01

interface IF_GW_SIP_5061
bind interface eth0 context router port 5061

context sip-gateway GW_SIP_01
bind location-service LOC_SRV_01
no shutdown

context sip-gateway GW_SIP_02

interface IF_GW_SIP_5062
bind interface eth0 context router port 5062

context sip-gateway GW_SIP_02
bind location-service LOC_SRV_02
no shutdown

context sip-gateway GW_SIP_03

interface IF_GW_SIP_5063
bind interface eth0 context router port 5063

context sip-gateway GW_SIP_03
bind location-service LOC_SRV_03
no shutdown

context sip-gateway GW_SIP_04

interface IF_GW_SIP_5064
bind interface eth0 context router port 5064

context sip-gateway GW_SIP_04
bind location-service LOC_SRV_04
no shutdown

context sip-gateway GW_SIP_05

interface IF_GW_SIP_5065
bind interface eth0 context router port 5065

context sip-gateway GW_SIP_05
bind location-service LOC_SRV_05
no shutdown

context sip-gateway GW_SIP_06

interface IF_GW_SIP_5066
bind interface eth0 context router port 5066

context sip-gateway GW_SIP_06
bind location-service LOC_SRV_06
no shutdown

context sip-gateway GW_SIP_07

interface IF_GW_SIP_5067
bind interface eth0 context router port 5067

context sip-gateway GW_SIP_07
bind location-service LOC_SRV_07
no shutdown

port ethernet 0 0
medium auto
encapsulation ip
bind interface eth0 router
no shutdown

port fxs 0 0
use profile fxs us
encapsulation cc-fxs
bind interface IF_FXS_00 switch
no shutdown

port fxs 0 1
use profile fxs us
encapsulation cc-fxs
bind interface IF_FXS_01 switch
no shutdown

port fxs 0 2
use profile fxs us
encapsulation cc-fxs
bind interface IF_FXS_02 switch
no shutdown

port fxs 0 3
use profile fxs us
encapsulation cc-fxs
bind interface IF_FXS_03 switch
no shutdown

port fxs 0 4
use profile fxs us
encapsulation cc-fxs
bind interface IF_FXS_04 switch
no shutdown

port fxs 0 5
use profile fxs us
encapsulation cc-fxs
bind interface IF_FXS_05 switch
no shutdown

port fxs 0 6
use profile fxs us
encapsulation cc-fxs
bind interface IF_FXS_06 switch
no shutdown

port fxs 0 7
use profile fxs us
encapsulation cc-fxs
bind interface IF_FXS_07 switch
no shutdown

The command sequence below enables SmartNode to provide the Austria 450Hz style tone set for dial-tones and call-progress tones.

Command Sequence to copy, edit and paste

profile call-progress-tone AUT450_Dialtone
  play 5000 450 -12 450 -12

profile call-progress-tone AUT450_Alertingtone
  play 1000 450 -12
  pause 5000

profile call-progress-tone AUT450_Busytone
  play 300 450 0
  pause 300
 
profile tone-set default
  map call-progress-tone dial-tone AUT450_Dialtone
  map call-progress-tone ringback-tone AUT450_Alertingtone
  map call-progress-tone busy-tone AUT450_Busytone
  map call-progress-tone release-tone AUT450_Busytone
  map call-progress-tone congestion-tone AUT450_Busytone

Route Retry feature is applied when more than one matching route is returned by the active routing script. Algorithm works in the following way:

• outgoing call will be made based on the first matching route
• if call fails while there are other matching routes left, new outgoing call attempt is made using next matching route 

Call attempt is considered failed in the following conditions:

• Call is terminated by remote side before reaching "Answered" state. Termination cause will determine if route retry is needed or not (enabled or disabled in 'Profiles -> Edit -> Reason Cause Mapping')
• Timeout occurs before call reaches specified call state (this timeout is ignored when last matching route is attempted)

Route retry timeout can be customized using two parameters:

• 'Menu -> Gateway -> Configuration -> Advanced Parameters -> Route retry timeout' - timeout delay in seconds to reach 'minimum call state' (value of 0 makes timeout never occur)
• 'Menu -> Gateway -> Configuration -> Advanced Parameters -> Route retry mode' - minimum call state to reach before the specified timeout occurs:
  • Call accepted
  • Call progress received
  • Call alerted
  • Call answered

If the timeout delay is expired before call reaches the specified state, call is considered failed and retry occurs. If the call reaches the minimal state, timeout isn't taken under consideration anymore, but explicit termination from remote side with error cause may still trigger retry with next route.

The 4 states above correspond to the following protocol-specific events:

SIP:
 Accepted: 100 Trying
 Progress: 183 with SDP (*)
 Alerted:  180 Ringing
 Answered: 200 OK
ISDN:
 Accepted: CALL PROCEEDING
 Progress: PROG
 Alerted:  ALERTING
 Answered: CONNECT
SS7:
 Accepted: <none>
 Progress: CPG
 Alerted:  ACM
 Answered: CONN/ANM
CAS R2:
 Accepted: SeizureAckCasBit
 Progress: <none>
 Alerted:  GroupB SuccessDigit
 Answered: AnswerCasBit

(*) 183 with SDP is usually interpreted as a "ALERT" (not a PROGRESS) unless explicitly configured on NAP level ("NAP -> Edit -> Advanced Parameters ->183 triggers call progress")

Route Retry termination cause configuration

To control if the route retry algorithm should stop or continue to the next route, each termination cause can be configured in the profile and set to either Continue or Stop.

Configuration is delivered via WebPortal in Profile setup 'Menu -> Profiles -> Edit -> Reason Cause Mapping -> Edit Reason Cause Mapping' and applied per NAP or Route basis.

Route Retry global configuration

Route Retry mode and timeout parameters can be configured globally, per NAP, or per Route. Configuration precedence takes following order:

1. Route configuration
2. NAP configuration
3. Global clonfiguration

The global Route Retry mode and timeout parameters can be found in the SmartMedia WebPortal Gateway application configuration menu:

• Go to the 'Menu -> Gateway -> Configuration' menu.
• Expand the 'Advanced Parameters' section
• Modify accordingly the values of 'Route retry mode' and 'Route retry timeout'
• Press Save

Route Retry per-NAP configuration

Per-NAP Route Retry mode and timeout parameters are configured by adding a custom 'NAP column':

• Go to the 'Menu -> Gateway -> Routes -> NAP Columns' menu.
• Click on Create New Nap Column in the 'Editing Naps' section
  • In the Name text box, enter 'route_retry_mode'
  • In the Type attributes text box, enter '|accept|call_progress|alert|answer'
  • Leave the Default text box blank
• Click Save
• Click again on Create New Nap Column in the 'Editing Naps' section
  • In the Name text box, enter 'route_retry_timeout'
  • In the Type attributes text box, enter 'integer'
  • Leave the Default text box blank
• Click Save

After creating above columns, each time you create or edit a NAP in NAP Columns menu, Route Retry mode and timeout paramters will be available. In case those are left blank for a particular NAP, global configuration will take precedence. Assigning a value in either of these parameters will override the global value for that parameter.

Route Retry per-route configuration

Per-route Route Retry mode and timeout parameters are configured by adding a custom 'Route column':

• Go to the 'Menu -> Gateway -> Routes' menu.
• Click on Create New Route Column in the 'Editing Routes' section
  
  • In the Name text box, enter 'route_retry_mode'
  • In the Type attributes text box, enter '|accept|call_progress|alert|answer'
  • Leave the Default text box blank
• Click Save
• Click again on Create New Route Column in the Editing Routes section
  • In the Name text box, enter 'route_retry_timeout'
  • In the Type attributes text box, enter 'integer'
  • Leave the Default text box blank
• Click Save

After creating above columns, each time you create or edit a Route, Route Retry mode and timeout parameters will be shown under Custom Params. In case those are left blank for a particular Route, global configuration will take precedence. Assigning a value in either of these parameters will override the global value for that parameter.

Customize Route retry parameters from routing script

Any part of the routing script that has access to the route object can read or modify below parameters:

• route[:route_retry_mode]
• route[:route_retry_timeout]

Those fields initially contain values of 'Custom Route Columns' (as explained above), but can be overwritten by the routing script.

STEP 1>>enable recording

STEP 2>>>retrieve recorded files

Select Call Trace under Maintenance.

STEP 3>>>

Select which recording you would like to analyze or download.

Step 4>>>>

• Selects the best matching entry
• Sets/Changes the value of the defined Information Element
• Mapping tables together with regular expressions are more powerful

Mapping-Table Types:





Creating/Entering and changing a Mapping-Table:

To do this, you must be in the 'context cs' section of the configuration.

(ctx-cs)# mapping-table <input-type> to <output-type> <name>
(map-tab)[MAP]#map <input-value> to <output-value>



Sample Mapping-Tables:

mapping-table called-e164 to called-e164 REM_PREFIX_911
map .(.%) to \1

ex.  If you have to dial a 9 to get an outside line
9911 converts to 911 
(Helps comply with the newly enacted 'Keri's Law')


mapping-table calling-e164 to calling-e164 ADD_00
map (.%) to 00\1

ex.  If you want to have a prefix added to every number dialed (For accounting purposes, least call routing, etc...)
3019751000 gets mapped to 003019751000


mapping-table called-e164 to calling-pi CNPN_PI
map *31*(.%) to restricted

ex.  If you need to change the presentation for Privacy reasons
When *31*1234567 is dialed, the Presentation Indicator is mapped to restricted.


To use multiple mapping-tables, a Complex-Function can be configured. This is an ordered list so be aware of this when creating this complex-function.

complex-function CF_TO_PSTN
execute 1 CNPN_PI
execute 2 REM_PREFIX


How to use a Mapping-Table:

Mapping-Tables are best used within a routing-table.  See the example below that uses the mapping-table listed above (REM_PREFIX_911).

routing-table called-e164 OUTBOUND
     route .T dest-interface IF_SIP REM_PREFIX_911