Oracle 4600 Acme Packet Session Border Controller, Skype for Business Technical Application Note
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Oracle Enterprise Session Border Controller
–
Acme Packet 4600 and Microsoft Skype for
Business for Enterprise SIP Trunking with
CenturyLink
Technical Application Note
Disclaimer
The following is intended to outline our general product direction. It is intended for information purposes only, and may not be incorporated into any contract. It is not a commitment to deliver any material, code, or functionality, and should not be relied upon in making purchasing decisions. The development, release, and timing of any features or functionality described for
Oracle’s products remains at the sole discretion of Oracle.
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Intended Audience
This document is intended for use by Oracle personnel, third party Systems Integrators, and end users of the Oracle Enterprise
Session Border Controller (E-SBC). It assumes that the reader is familiar with basic operations of the Oracle Enterprise Session
Border Controller
– Acme Packet 4600.
Document Overview
Microsoft Skype for Business offers the ability to connect to SIP based telephony trunks using an IP communication. This reduces the cost and complexity of extending an enterprise’s telephony system outside its network borders. Oracle Enterprise Session
Border Controllers (E-SBCs) play an important role in SIP trunking as they are used by many trunk providers and some enterprises as part of their SIP trunking infrastructure.
This application note has been prepared as a means of ensuring that SIP trunking between Microsoft Skype for Business servers,
Oracle E-SBCs and IP Trunking services are configured in the optimal manner.
It should be noted that the E-SBC configuration provided in this guide focuses strictly on the Skype for Business Server associated parameters. Many E-SBC users may have additional configuration requirements that are specific to other applications. These configuration items are not covered in this guide. Please contact your Oracle representative with any questions pertaining to this topic.
For additional information on Skype for Business Server, please visit http://www.skype.com/en/business/.
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Introduction
Audience
This is a technical document intended for telecommunications engineers with the purpose of configuring the Oracle Enterprise SBC and the Skype for Business Server. There will be steps that require navigating Microsoft Windows Server as well as the Acme
Packet Command Line Interface (ACLI). Understanding the basic concepts of TCP/UDP, IP/Routing, and SIP/RTP are also necessary to complete the configuration and for troubleshooting, if necessary.
Requirements
Fully functioning Skype for Business Server deployment, including Active Directory and DNS
A dedicated Mediation Server for the SIP trunking connection
Microsoft Skype for Business 2015
– Version 6.0.93190.0
Skype for Business 2015 client, Version 15.0.4753.1000
Oracle Enterprise Session Border Controller AP 4600 running Net-Net OS ECZ730m1p1.64.bz. Note: the configuration running on the SBC is backward/forward compatible with any release in the 7.3.0 stream.
Architecture
The following reference architecture shows a logical view of the connectivity between Skype for Business Server and the SBC.
The Enterprise Network represents the customer’s on-premise infrastructure, which includes the Active Directory, DNS and Skype for Business Server systems. The SIP Trunk Network represents the infrastructure of the SIP trunk provider, CenturyLink, which provides PSTN service via the SIP trunk. Area C represents the integration of these two environments over an IP network. This could be, through a VPN tunnel over the Internet, an MPLS managed network, or even a dedicated physical connection. The SFB
Mediation Server and the SBC are the edge components that form the boundary of the SIP trunk.
As per the CenturyLink network requirements, the customer devices are required to register and support authentication. The Oracle
ESBC supports authentication and performs registration on behalf of the Skype for Business Server. The registration can be done using either the single trunk architecture or the dual trunk architecture. In the single trunk architecture, the Oracle ESBC registers a single trunk to the CTL Carrier SBC and in the dual trunk architecture; the ESBC registers to a pair of geo-redundant CTL carriers
SBC. This guide documents both single and dual trunk architectures, starting with the configuration required for the testing of the single trunk architecture.
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The configuration, validation and troubleshooting of the areas B and C is the focus of this document and will be described in two phases:
Phase 1
– Configure the Skype for Business Server
Phase 2
– Configure the 4600
Lab Configuration
The following diagram, similar to the Reference Architecture described earlier in this document, illustrates the lab environment created to facilitate certification testing (IP addressing/Port below is only a reference, they can change per your network specification).
Lync client 2
192.168.4.142
Ph no: 4695737005
Lync client 1
192.168.4.141
Ph no: 4695737006
CORE-Microsoft SFB
Server 1
Domain Controller
192.168.4.150
Lync server
192.168.4.151
Exchange server
192.168.4.155
Server 2
Mediation server 1
192.168.4.152
Mediation server 2
192.168.4.153
Monitoring server
192.168.2.194
s0p0
192.168.4.135
4600 Series
VPN to trunk s0p1
192.168.1.220
PEER – CenturyLink
Trunk
Sip Trunk
192.168.1.170
PSTN
Ph no: 7813282518
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Phase 1
– Configuring the Skype for Business server
The enterprise will have a fully functioning Skype for Business Server infrastructure with Enterprise Voice deployed and a Mediation
Server dedicated to this installation. If there is no Mediation Server present for this purpose, one will have to be deployed.
There are two parts for configuring SFB Server to operate with the Oracle SBC:
Adding the SBC as a PSTN gateway to the SFB Server infrastructure
Creating a route within the SFB Server infrastructure to utilize the SIP trunk connected through the SBC.
To add the PSTN gateway, we will need:
IP addresses of the external facing NICs of the Mediation Servers
IP address of the sip interface of the SBC facing the Mediation servers
Rights to administer Lync Server Topology Builder
Access to the Lync Server Topology Builder
Note: This section of the Application note only walks you through adding Oracle E-
SBC to Microsoft’s Skype for Business config, The assumption is that Microsoft’s Skype for Business application is already installed and 100% functional.
Adding the SBC as a PSTN gateway
The following process details the steps to add the SBC as the PSTN gateway
1. On the server where the Topology Builder is located, start the console.
2. From the Start bar, select Lync Server Topology Builder.
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3. The opening screen of the Topology builder will be displayed. Click on the Cancel button.
4. The Topology Builder window will now be displayed. Click on Action and select Download Topology.
5. You will then see a screen showing that you have successfully imported the topology. Click the Ok button.
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6. Next you will be prompted to save the topology which you have imported. You should revision the name or number of the topology according to the standards used within the enterprise. Click the Save button
Note: This keeps track of topology changes and, if desired, will allow you to fall back from any changes you make during this installation
7.
You will now see the topology builder screen with the enterprise’s topology imported.
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8. In the upper left hand corner, expand the site in which the PSTN gateway will be added. In our case, the site is Bedford.
Then click on the PSTN Gateways
9. Right click on PSTN gateways and select New IP/PSTN Gateway.
10. In the Define New IP/PSTN Gateway window, enter the ip address of the SIP interface of the SBC in the FQDN text box and click Next.
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11. Select Enable IPv4 in the Define the IP address section and click Next.
12. In the next section, enter the ip address of the SBC
’s sip interface under Trunk name. Configure the Listening port for
IP/PSTN gateway as 5068, TCP as the SIP Transport Protocol and click Finish.
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13. The PSTN gateway for the Lync server has been added. It will be listed under PSTN gateways.
Expand the Mediation Pool list and click on the Mediation Server to be utilized. In our example the Mediation Server is
sfbmedpool.acmepacket.net.
You will see that the PSTN gateway is associated with the Mediation server.
14. In the upper right hand corner of your screen under Actions select Topology then select Publish.
You will now see the Publish Topology window. Click on the Next button
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You will now be at a window showing the databases associated with site. Click Next.
15. When complete you should see a window from Topology Builder stating that your topology was successfully published. Click the Finish button.
16. You will be at the Topology Builder main window, expand your site and double check that your PSTN entries are correct and that the appropriate Mediation Server has the PSTN gateway associated.
Creating a route within the Lync Server infrastructure
In order for the Lync Server Enterprise Voice clients to utilize the SIP trunking infrastructure that has been put in place, a route will need to be created to allow direction to this egress. Routes specify how Lync Server handles calls placed by enterprise voice users.
When a user places a call, the server, if necessary, normalizes the phone number to the E.164 format and then attempts to match that phone number to a SIP Uniform Resource Identifier (URI). If the server is unable to make a match, it applies outgoing call routing logic based on the number. That logic is defined in the form of a separate voice route for each set of target phone numbers listed in the location profile for a locale. For this document we are only describing how to set up a route. Other aspects which apply to Lync Server Enterprise Voice deployments such as dial plans, voice policies, and PSTN usages are not covered.
To add the route we will need:
Rights to administer Lync Server Control Panel
- Membership in the CS Administrator Active Directory Group
Access to the Lync Server Control Panel
The following process details the steps to create the route:
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1. From the Start bar, select Lync Server Control Panel.
You will be prompted for credential, enter your domain username and password.
2.
Once logged in, you will now be at the “Welcome Screen”. On the left hand side of the window, click on Voice Routing.
3. The Dial Plan tab in the Voice Routing section will be displayed. On the content area toolbar, click +New.
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4. Next you build a Dial Plan and a translation rule for the phone numbers you want this route to handle.
US Dial-plan
Match this pattern: ^(\d*)$
Translation rule: +1$1
Additional normalization rules will need to be created as shown below to allow dialing the feature codes.
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5. On the top row of the tabs, select Route. On the content area toolbar, click +New.
6. On the New Voice Route page, in the Name field, enter the name you have selected for the Route. In our example, it is US route.
7. Next you build a Pattern Match for the phone numbers you want this route to handle. Click Edit.
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8. Next you want to associate the Voice Route with the Trunk you have just created. Scroll down to Associated Trunks, click on the Add button.
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9. You will now be at a window showing available Trunks to associate your Voice Route. Click on the PSTN gateway that you just created and then click the OK button.
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10. You can now see that you have associated your trunk with the route you created. An appropriate PSTN usage record will need to be assigned as well. In our example, we use one that was already created in the enterprise. Click on the Select button under Associated PSTN Usages.
11. In the Select PSTN Usage Record window displayed, select the appropriate PSTN Usage Record and click OK.
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12. You will now see the Associated PSTN Usages which you have added. Click the OK button at the top of the New Voice
Route screen.
13. You will now be at the Routes page showing the US route. Click the Commit drop-down menu, and then Commit All.
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14. On the Uncommitted Voice Configuration Settings window, click OK.
If there are no errors, the new Voice Route has now been successfully created and the state will show as Committed.
Additional Steps
There are other aspects to a Lync Server Enterprise Voice deployment such as
Site, local, and global dial plans
Voice Policies
Assigning Voice Policies to users
PSTN usage policies
Refer to MSDN technet for relevant information.
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Phase 2
– Configuring the Oracle Enterprise SBC
In this section we describe the steps for configuring an Oracle Enterprise SBC, formally known as an Acme Packet Net-Net Session
Director (“SBC”), for use with Skype for Business Server in a SIP trunking scenario.
In Scope
The following guide configuring the Oracle SBC assumes that this is a newly deployed device dedicated to a single customer. If a service provider currently has the SBC deployed and is adding SFB Server customers, then please see the ACLI Configuration
Guide on http://docs.oracle.com/cd/E61547_01/index.html
for a better understanding of the Command Line Interface (CLI).
Note that Oracle offers several models of SBC. This document covers the setup for the 4600 platform series running Net-Net OS
ECZ7.3.0 or later. If instructions are needed for other Oracle SBC models, please contact your Oracle representative.
Out of Scope
Configuration of Network management including SNMP and RADIUS; and
What will you need
Serial Console cross over cable with RJ-45 connector
Terminal emulation application such as PuTTY or HyperTerm
Passwords for the User and Superuser modes on the Oracle SBC
IP address to be assigned to management interface (Wancom0) of the SBC - the Wancom0 management interface must be connected and configured to a management network separate from the service interfaces. Otherwise the SBC is subject to ARP overlap issues, loss of system access when the network is down, and compromising DDoS protection.
Oracle does not support SBC configurations with management and media/service interfaces on the same subnet.
IP address of Mediation Server external facing NIC
IP addresses to be used for the SBC internal and external facing ports (Service Interfaces)
IP address of the next hop gateway in the service provider network
IP address of the enterprise DNS server
SBC- Getting Started
Once the Oracle SBC is racked and the power cable connected, you are ready to set up physical network connectivity. Note: use
the console port on the front of the SBC, not the one on the back.
Plug the slot 0 port 0 (s0p0) interface into your outside (gateway facing) network and the slot 0 port 1 (s1p0) interface into your inside (SFB server-facing) network. Once connected, perform you are ready to power on and perform the following steps.
All commands are in bold, such as
configure terminal; parameters in bold red such as oraclesbc1 are parameters which are specific to an individual deployment. Note: The ACLI is case sensitive.
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Establish the serial connection and logging in the SBC
Confirm the SBC is powered off and connect one end of a straight-through Ethernet cable to the front console port (which is active by default) on the SBC and the other end to console adapter that ships with the SBC, connect the console adapter (a DB-9 adapter) to the DB-9 port on a workstation, running a terminal emulator application such as PuTTY. Start the terminal emulation application using the following settings:
Baud Rate=115200
Data Bits=8
Parity=None
Stop Bits=1
Flow Control=None
Power on the SBC and confirm that you see the following output from the bootup sequence.
Enter the following commands to login to the SBC and move to the configuration mode. Note that the default SBC password is
“acme” and the default super user password is “packet”.
Password: acme oraclesbc1> enable
Password: packet oraclesbc1# configure terminal oraclesbc1(configure)#
You are now in the global configuration mode.
Initial Configuration
– Assigning the management Interface an IP address
To assign an IP address, one has to configure the bootparams on the SBC by going to oraclesbc1#configure terminal --- >bootparams
Once you type “bootparam” you have to use “carriage return” key to navigate down
A reboot is required if changes are made to the existing bootparams
ACMESYSTEM(configure)# bootparam
'.' = clear field; '-' = go to previous field; q = quit
Boot File : /boot/nnECZ730m1p1.64.bz
IP Address : 192.65.79.44
VLAN :
Netmask : 255.255.255.224
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Gateway : 192.65.79.33
IPv6 Address :
IPv6 Gateway :
Host IP : 0.0.0.0
FTP username : vxftp
FTP password : vxftp123
Flags :
Target Name : ACMESYSTEM
Console Device : COM1
Console Baudrate : 115200
Other :
NOTE: These changed parameters will not go into effect until reboot.
Also, be aware that some boot parameters may also be changed through
PHY and Network Interface Configurations.
Configuring the SBC
The following section walks you through configuring the Oracle Enterprise SBC required to work with Skype for Business (SFB) and
CenturyLink SIP trunk. The information below is split into three sections:
The single trunk test plan utilizing the G711 codec - The transport protocol for signaling/media between SBC and SFB is
TCP and RTP
– the SIP trunk utilizes UDP/RTP.
The single trunk test plan utilizing the G729 codec - The transport protocol for signaling/media between SBC and SFB is
TCP and RTP using the G711 codec
– the SIP trunk utilizes UDP/RTP and supports G729 codec, the SBC performs transcoding in this scenario.
It is outside the scope of this document to include all the interoperability working information as it will differ in every deployment.
SIP PRACK interworking
In order to establish an early media session for outbound calls, Skype for Business gateway specification mandates the PSTN gateways to offer a reliable provisional response and for inbound calls offer INVITEs with a supported header. The SBC interworks the messaging and provide RFC 3262 PRACK interworking towards SFB and it is a mandatory configuration in all Oracle ESBC
–
Microsoft SFB deployments. The following need to be configured on ESBC
Configure option 100rel-interworking on the sip-interface facing mediation server
Configure a sip-feature to pass the 100rel in supported and require headers
Configure a sip-manipulation (ForEarlyMedia) to add a Require:100rel header in incoming SIP INVITE from mediation server and delete the Supported:100rel header
Local REFER handling by the SBC
In event of transferring calls from a SFB client (calls could be initially placed between 2 SFB Clients, but then transferred out to
PSTN), SFB has two OPTIONS:
Transfer using re-INVITE approach
Transfer using REFER method --- the info below is required when REFER based transfer is enabled.
Oracle E-SBC supports REFER method termination when received from SFB. After termination the ESBC uses the info in refer-to header and sends a newly generated INVITE towards the SFB Mediation server. SFB then processes the INVITE, authorizes the call transfer and sends a new INVITE (for calls transferred to PSTN) to the SBC or transfers call internally to the transferred SFB client
To handle the call transfer and refer scenarios
– when SFB client 1 refers/transfers the call to SFB Client 2 or to a party on the
PSTN, we will need two routes to route to the two mediation servers depending on the referred party:
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local-policy
from-address *
to-address sfbmedpool.acmepacket.net
source-realm SIP-Trunk
description For referred party header
activate-time
deactivate-time
state enabled
policy-priority none
policy-attribute
next-hop sfbmedpool.acmepacket.net
realm towards-sfb
action replace-uri
terminate-recursion disabled
carrier
start-time 0000
end-time 2400
days-of-week U-S
cost 0
state enabled
app-protocol SIP
methods
media-profiles
lookup single
next-key
eloc-str-lkup disabled
eloc-str-match
Ring-back tone during Transfers
During call transfer to a PSTN party, the transfer completes but the calling party does not hear a ring back tone during transfer. The
INVITE Lync sends to the SBC to initiate the transfer contains the SDP attribute, a=inactive which is forwarded to the trunk and as a result of which the SBC cannot play the ring back tone to the original PSTN caller (while call is being transferred). A sendonly attribute is required for MoH and transfer scenarios for the calling party to be able to hear ringback or MoH when it is kept on hold.
The SBC is able to signal appropriately towards the SIP trunk by changing the
“a=inactive” SDP attribute in the INVITE to sendonly towards PSTN.
Sip manipulations are configured to make the necessary changes. The manipulation ForEarlyMedia is configured to change the
SDP attribute from a=inactive to a=sendonly in the INVITEs sent to the calling party for transfer
sip-manipulation
name Changeinactosendonly
description Change inactive to sendonly for transfer
split-headers
join-headers
header-rule
name changeSDP
header-name Content-Type
action manipulate
comparison-type case-sensitive
msg-type request
methods INVITE
match-value
new-value
element-rule
name inacttosendonly
parameter-name application/sdp
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type mime
action find-replace-all
match-val-type any
comparison-type pattern-rule
match-value a=inactive
new-value a=sendonly
We utilize the local playback feature of the SBC to play ring back tone during transfers. The ringback tone is played based on
REFER termination. You must upload an audio file (in .raw format) to /code/media onto the ESBC for the media you want played during the transfer. A separate file is required for each different codec type, even if the media itself is the same.
The playback configuration is defined listing the media files that you want to play. The playback-config element is configured under media-manager. playback-config
name transferrbt
entry
encoding PCMU
filename US_ringbackPCMU.raw
bytes-per-sec 8000
The playback options can be applied to realms, sip-interfaces or session agents using the spl-options command. oraclesbc1(session-router)# sip-interface oraclesbc1(sip-interface)# sel
<realm-id>:
1: towards-sfb 192.168.2.225:5060
2: SIP-trunk 192.168.1.220:5060 selection: 1 oraclesbc1(sip-interface)# spl-options playback-on-refer="transferrbt" oraclesbc1(sip-interface)# done
SIP manipulations
SFB and SIP trunk carry their own SIP and SDP design
– not always these implementation methods align, causing a lot of mismatch in SIP and SDP signaling and call flow. The ESBC helps resolve these issues with SIP manipulation feature. In this section, we detail the changes that we implement to make sure the signaling is per the specifications of the trunk and SBC.
The HMR applied to the signaling towards the trunk performs the following changes:
- The Request-URI is modified to include the ip address and port of the trunk device
- The P-Asserted-Identity header sent by SFB is deleted and a new PAI header is added to include the pilot number that is used to register with the trunk, in our case we delete the original PAI header and add a PAI with the value - sip:[email protected]
- The uri-host portion of the From header is replaced with the FQDN of the trunk, in our case the uri-host is changed to voip.centirylink.com
- In the Contact header, we have header rules to strip +1 from the uri-user and replace the uri-host and uri-port portions with the SBC’s local ip and port of the interface facing the trunk.
- In the Route header we remove the +1 from the uri-user.
- For privacy enabled calls, SFB sends the phone number in the From header. It indicates that it is a privacy enabled calls using the
‘Privacy:id’ header. For such calls, we replace the phone number in the uri-user of the From header with ‘anonymous’.
To conform SFB’s signaling per the trunk’s specification, we modify the messages coming from SFB and also make some changes to messages before they are sent to SFB.
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The following changes are applied to the messages coming from SFB:
- We add a
‘Require:100rel’ header in incoming SIP INVITE from mediation server and delete the ‘Supported:100rel’ header as mentioned in the SIP PRACK interworking section.
- To enabled ringback on transfers, we replace the
‘a=inactive’ line in SDP of the INVITEs with ‘a=sendonly’. For more information, please refer to the Ring-back tone during Transfers section.
To the messages sent to SFB, the following changes are applied:
- The urihosts of the From and To headers are replaced with SBC’s local ip and SFB’s ip.
- In the From and To headers we remove the +1 from the uri-user, when the uri-user is anonymous.
- At last we have a rule to insert +1 in the uri-user of the Contact header as SFB server is configured for E.164 format.
SBC Configurations
SBC configuration for Single Trunk registration using PCMU towards trunk
Following is the configuration of the SBC: local-policy
from-address *
to-address *
source-realm core
policy-attribute
next-hop 192.168.1.220
realm ctl-trunk local-policy
from-address *
to-address SFBMed1.partnersfb.com
source-realm core
policy-attribute
next-hop SFBMed1.partnersfb.com
realm core
action replace-uri local-policy
from-address *
to-address SFBMed2.partnersfb.com
source-realm core
policy-attribute
next-hop SFBMed2.partnersfb.com
realm core
action replace-uri local-policy
from-address 4695550016
to-address *
source-realm core
policy-attribute
next-hop 192.168.1.220
realm ctl-trunk local-policy
from-address *
to-address *
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source-realm ctl-trunk
policy-attribute
next-hop medpool.partnersfb.com
realm core media-manager media-policy
name voip-default
tos-settings
media-type audio
tos-value 0x28
tos-settings
media-type message
media-sub-type sip
tos-value 0x18 network-interface
name s0p0
ip-address 192.168.1.220
netmask 255.255.255.0
gateway 192.,168.1.1
hip-ip-list 192.168.1.220
icmp-address 192.168.1.220 network-interface
name s1p0
hostname
ip-address 192.168.4.135
netmask 255.255.255.0
gateway 192.168.4.1
dns-ip-primary 192.168.4.150
dns-domain partnersfb.com
hip-ip-list 192.168.4.135
icmp-address 192.168.4.135 phy-interface
name s0p0
operation-type Media phy-interface
name s1p0
operation-type Media
slot 1 playback-config
name transferrbt
entry
encoding PCMU
filename US_ringbackPCMU.raw realm-config
identifier core
network-interfaces s1p0:0
mm-in-realm enabled
qos-enable enabled
restricted-latching sdp realm-config
identifier ctl-trunk
network-interfaces s0p0:0
mm-in-realm enabled
qos-enable enabled
media-policy voip-default session-agent
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hostname 192.168.1.220
ip-address 192.168.1.220
port 5100
realm-id ctl-trunk
out-translationid stripplus1
out-manipulationid Manipto7016 session-agent
hostname medpool.partnersfb.com
port 5068
transport-method StaticTCP
realm-id core
ping-method OPTIONS
ping-interval 30
out-translationid addplus1
in-manipulationid ForEarlyMedia
out-manipulationid outManipToSFB
refer-call-transfer enabled
auth-attributes
auth-realm voip.centurylink.com
username 257570-4695550016
password ********
in-dialog-methods INVITE session-translation
id addplus1
rules-calling addplus1
rules-called addplus1 session-translation
id stripplus1
rules-calling stripplus1
rules-called stripplus1 sip-config
home-realm-id core
registrar-domain *
registrar-host *
registrar-port 5060
options inmanip-before-validate
max-udp-length=0
sip-message-len 6000
refer-src-routing enabled sip-feature
name 100rel-interworking
realm ctl-trunk
require-mode-inbound Pass
require-mode-outbound Pass sip-interface
realm-id core
sip-port
address 192.168.4.135
transport-protocol TCP
allow-anonymous agents-only
registration-caching enabled
options 100rel-interworking
spl-options playback-on-refer="transferrbt" sip-interface
realm-id ctl-trunk
sip-port
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address 192.168.1.220
allow-anonymous agents-only sip-manipulation
name Changeinactosendonly
header-rule
name changeSDP
header-name Content-Type
action manipulate
msg-type request
methods INVITE
element-rule
name inacttosendonly
parameter-name application/sdp
type mime
action find-replace-all
comparison-type pattern-rule
match-value a=inactive
new-value a=sendonly sip-manipulation
name Check_privacy_header
header-rule
name ChechForPrivacy
header-name Privacy
action manipulate
msg-type request
methods INVITE
header-rule
name OverwriteFromDisplay
header-name From
action manipulate
comparison-type boolean
msg-type request
methods INVITE
match-value $ChechForPrivacy
element-rule
name OverwriteUser
type uri-user
action find-replace-all
new-value anonymous
element-rule
name OverwriteDisplay
type uri-display
action find-replace-all
new-value "\"Anonymous\" " sip-manipulation
name ForEarlyMedia
header-rule
name delsupported
header-name Supported
action delete
msg-type request
methods INVITE
header-rule
name addrequireinINVITE
header-name Require
action add
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msg-type request
methods INVITE
new-value 100rel
header-rule
name Fortransfer
header-name From
action sip-manip
new-value Changeinactosendonly sip-manipulation
name Manipto7016
header-rule
name modURI
header-name request-uri
action manipulate
element-rule
name mod2
type uri-host
action replace
match-val-type fqdn
new-value $REMOTE_IP+:+$REMOTE_PORT
header-rule
name DeletePAI
header-name P-Asserted-Identity
action delete
methods INVITE
header-rule
name ReplacePAI
header-name P-Asserted-Identity
action add
comparison-type pattern-rule
methods INVITE
new-value sip:[email protected]
header-rule
name from_voip_ctl_com
header-name From
action manipulate
methods INVITE
element-rule
name From_add_voip_ctl_com
type uri-host
action replace
new-value voip.centurylink.com
header-rule
name registercontactlocalipport
header-name Contact
action manipulate
msg-type request
methods REGISTER
element-rule
name registercontactlocalip
type uri-host
action replace
match-val-type ip
new-value $LOCAL_IP
element-rule
32 | P a g e
name registercontactlocalport
type uri-port
action replace
new-value $LOCAL_PORT
header-rule
name fixContact2
header-name Contact
action manipulate
element-rule
name updatecon2
type uri-user
action replace
comparison-type pattern-rule
match-value (.*)
new-value $ORIGINAL-^"+1"
header-rule
name fixRoute2
header-name Route
action manipulate
element-rule
name updateroute2
type uri-user
action replace
comparison-type pattern-rule
match-value (.*)
new-value $ORIGINAL-^"+1"
header-rule
name Privacy
header-name From
action sip-manip
new-value Check_privacy_header sip-manipulation
name outManipToSFB
header-rule
name From
header-name From
action manipulate
msg-type request
element-rule
name From_header
type uri-host
action replace
new-value $LOCAL_IP
element-rule
name modanonfrom
type uri-user
action replace
comparison-type pattern-rule
match-value \+1anonymous
new-value anonymous
header-rule
name To
header-name To
action manipulate
msg-type request
33 | P a g e
element-rule
name To
type uri-host
action replace
new-value $REMOTE_IP
header-rule
name fixcontactuser
header-name Contact
action manipulate
comparison-type pattern-rule
element-rule
name user
type uri-user
action replace
comparison-type pattern-rule
match-value (.*)
new-value "+1"+$ORIGINAL sip-monitoring
match-any-filter enabled steering-pool
ip-address 192.168.1.220
start-port 20000
end-port 40000
realm-id ctl-trunk steering-pool
ip-address 192.168.4.135
start-port 20000
end-port 40000
realm-id core surrogate-agent
register-host voip.centurylink.com
register-user 4695550016
realm-id core
customer-host 192.168.1.220
customer-next-hop 192.168.1.220
register-contact-host 192.168.1.220
register-contact-user 4695550016
password *************
register-expires 60
auth-user 257570-4695550016 system-config
hostname CenturyLink-IOT
process-log-level DEBUG
comm-monitor
state enabled
monitor-collector
address 172.18.255.71
default-gateway 192.168.1.1 translation-rules
id addplus1
type add
add-string +1 translation-rules
id stripplus1
type delete
delete-string +1
34 | P a g e
SBC configuration for Single Trunk registration using G729 towards trunk
For this configuration, the SBC will need to transcode between G729 and PCMU codecs on the turnk and SFB side respectively.
For transcoding, we configure the codec polices, to-trunk and to-sfb, mentioned below and apply them on the trunk and SFB realms respectively. The following configuration shows the configuration elements that need to be changed for the transcoding configuration to take effect codec-policy
name to-trunk
allow-codecs * PCMU:no G729::vadoff
add-codecs-on-egress G729
order-codecs
packetization-time 20
force-ptime disabled
dtmf-in-audio disabled
last-modified-by [email protected]
last-modified-date 2016-08-08 16:43:07 codec-policy
name to-sfb
allow-codecs * G729:no
add-codecs-on-egress PCMU
order-codecs
packetization-time 20
force-ptime disabled
dtmf-in-audio disabled
last-modified-by [email protected]
last-modified-date 2016-08-08 16:43:07 media-profile
name G729
subname vadoff
payload-type 18
parameters annexb=no playback-config
name transferg729
entry
encoding G729
filename ringback-us-g729.raw realm-config
identifier core
network-interfaces s1p0:0
mm-in-realm enabled
qos-enable enabled
restricted-latching sdp
codec-policy core realm-config
identifier ctl-trunk
network-interfaces s0p0:0
mm-in-realm enabled
qos-enable enabled
media-policy voip-default
codec-policy access
35 | P a g e
Test Plan Executed
Following is the test plan executed against this setup and results have been documented below.
Title Description Test Setup
Status (Passed or Failed etc)
Anonymous
Call
Rejection
Activate
PBX User dials *77
PSTN Calls PBX User with Caller ID Block
Should receive an announcement
*77 is Dialed PBX and leaves PBX Phones gets an announcement
Calling Party blocks caller ID
Calling party makes a call to PBX User
Calling Party receives an announcement when PBX user is dialed
Passed
Anonymous
Call
Rejection
Deactivate
PBX User dials *87
PSTN Calls PBX User with Caller ID block
Call Should Complete
*87 is dialed PBX User receives and announcement
PSTN calls PBX User
PSTN Phone receives ringback
PBX Phone gets ringing
PBX Phone get Caller ID
PBX Phone answer the Call
2 way audio is received
PBX Phone releases Calls
PSTN receives a Bye
Alien TNs
A call PBX call originate where the from TN that is not part of the customer trunk group. As long as the pilot number is identified in outgoing call by PAI, the BroadWorks will accept and route the call.
After Alien TN is set up on a Trunk in
CenturyLink Network
PBX User Places a Call to PSTN
PBX User receives ringback
PSTN receives ringing
PSTN receives caller id of the Alien TN
PSTN answers the call
2 way audio is received
PBX Phone releases Calls
PSTN receives a Bye
Passed
Anonymous
Call PBX-BW
PBX sends anonymous call to BW
BW delivers the calls
Private or unknown or anonymous to PSTN
PBX is configured to send a call to BW as anonymous with TN as PSTN
BW delivers the call to PSTN as Private or
Anonymous
PSTN phone shows the call as Private or
Anonymous
Call is answered by PSTN
PBX user hangs up the call
Passed
Passed
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Barge In
Create a Pick Up Group with 2 PBX Users
PSTN Calls PBX User 1
PBX User 2 dials *33
+PBX User Ext
PSTN, User 1, and User
2 should be conf
PSTN calls PBX User 1
PSTN Phone receives ringback
PBX Phone gets ringing
PBX Phone get Caller ID
PBX Phone answer the Call
2 way audio is received
PBX User 2 Dials *33 + PBX User 1
Extension
PSTN, PBX User 1, and PBX User 2 are conferenced together
2 Way Audio is heard by all Legs
PBX User 1 drops from Call
2 way Audio is heard by PSTN and PBX
User 2
PSTN drops call
PBX User 2 receives a Bye
Barge In
Exempt
In the Portal Enable
Barge In Exempt
Create a Pick Up Group with 2 PBX Users
PSTN Calls PBX User 1
PBX User 2 dials *33
+PBX User Ext
User 2 Should not be conf
Barge in Exempt is set on PBX user 1
PSTN calls PBX User 1
PSTN Phone receives ringback
PBX Phone gets ringing
PBX Phone get Caller ID
PBX Phone answer the Call
2 way audio is received
PBX User 2 Dials *33 + PBX User 1
Extension
PBX user 2 is not allowed to barge in
PSTN drops the call
PBX User 1 receives a Bye
Passed
Passed
PSTN to
BWA
PSTN calls BWA Number
Enter Calling Number
(2nd Phone Location)
Enter Called Number
(PSTN)
PSTN should Ring with
Caller ID of 2nd Phone
Answer Call
BroadWorks Anywhere is set up in Portal
PSTN 1 Calls BWA Number
Announcement is received
Enter calling Number (2nd Phone created in BWA)
Announcement received
Enter Called Number (PSTN 2)
PSTN 1 receives ringback
PSTN 2 receives ringing
PSTN 2 receives caller ID of 2nd Phone
(Not of PSTN 1)
PSTN 2 Answers Call
2 way audio is received
PSTN 2 releases Calls
PSTN receives a Bye
Passed
37 | P a g e
PSTN to PBX user with
BWA
PSTN Calls User with
BWA
PBX User and 2nd
Location should Ring
Answer phone for 2nd location
BroadWorks Anywhere is set up in Portal
PSTN 1 Calls BWA Number
PSTN 1 receives ringback
Both PBX User and 2nd Phone Location
Number gets ringing
Both PBX User and 2nd Phone Location
Number gets Caller ID of PSTN
Call is answered on Location 2
PBX User no longer gets ringing (cancel)
2 way Audio
Location 2 releases call
PSTN receives a Bye
Passed
Call
Forwarding
Always
Activate
PBX User dials *72
Enter the CFA
Destination TN
PSTN calls PBX User with CFA
PBX User 1 Dials *72
Announcement is heard
PBX User enter PBX User 2 TN
Announcement is heard
PBX Receives a Bye
PSTN Calls PBX User 1
PSTN receives ringback
PBX User 1 does not ring
PBX User 2 gets ringing
PBX user 2 receives Caller ID (PSTN
Originator Caller)
PBX User answers call
2 way Audio
PSTN releases call
PBX User 2 receives a Bye
Passed
Call
Forwarding
Always
Interrogate
PBX User with CFA dials
*21*
Announcement received
PBX User 1 Dials *21*
Announcement is Heard
After announcement completes PBX User 1 receives a Bye
Call
Forwarding
Always
Deactivate
PBX User with CFA dials
*73
PSTN Calls PBX User
PBX User 1 Dials *73
Announcement is Heard
After announcement completes PBX User 1 receives a Bye
Passed
Passed
38 | P a g e
PSTN call is
CFB to PSTN with ID
Restricted
PBX configured to send
CFB to BW for identified
Station.
BW is configured with
CFB to PSTN2.
PSTN 1 Calls PBX with
Caller ID Restricted
PSTN 1 hears ring back
PBX send 486 Busy to
BW
BW forwards the call to
PSTN2
PSTN 2 hears ringing
PSTN 2 Caller ID displays
Private/Anonymous
PSTN 2 Answers the call.
Two way voice path is established between
PSTN 1 and PSTN 2
PSTN 2 hangs up
PSTN2 should receive Private/Anonymous as CLID
Passed
PSTN with
Privacy call to PBX is
CFA to PSTN
PBX User is configured with CFA to PSTN 2
PSTN 1 Calls PBX with
Caller ID Restricted
PSTN 1 hears ring back
PBX sends a new call to
BW with PSTN 2 Number,
From as Anonymous and
PAI set to Pilot Number
BW forwards the call to
PSTN2
PSTN 2 hears ringing
PSTN 2 Caller ID displays
Pilot Number
PSTN 2 Answers the call.
Two way voice path is established between
PSTN 1 and PSTN 2
PSTN 2 hangs up
Pilot Number should be shown as CLID on
PSTN2
Passed
39 | P a g e
Call
Forwarding
Busy Activate
PBX User dials *90
Enter the CFB
Destination TN
PSTN calls PBX User with CFB
PBX User 1 Dials *90
Announcement is heard
PBX User enter PBX User 2 TN
Announcement is heard
PBX Receives a Bye
Busy PBX User 1
PSTN Calls PBX User 1
PSTN receives ringback
PBX User 1 does not ring
PBX User 2 gets ringing
PBX user 2 receives Caller ID (PSTN
Originator Caller)
PBX User answers call
2 way Audio
PSTN releases call
PBX User 2 receives a Bye
Passed
Call
Forwarding
Busy
Interrogate
PBX User with CFB dials
*67*
Announcement received
PBX User 1 Dials *67*
Announcement is Heard
After announcement completes PBX User 1 receives a Bye
Call
Forwarding
Busy
Deactivate
PBX User with CFB dials
*91
PSTN Calls PBX User
PBX User 1 Dials *91
Announcement is Heard
After announcement completes PBX User 1 receives a Bye
Passed
Passed
Call
Forwarding
No Answer
Activate
PBX User dials *92
Enter the CFNA
Destination TN
PSTN calls PBX User with CFNA
PBX User 1 Dials *92
Announcement is heard
PBX User enters PBX User 2 TN
Announcement is heard
PBX Receives a Bye
PSTN Calls PBX User 1
PSTN receives ringback
PBX User 1 gets ringing
PBX User 1 receives Caller ID
After timer is RNA is received
PBX User 2 gets ringing
PBX user 2 receives Caller ID
PBX User 2 answers call
2 way Audio
PSTN releases call
PBX User 2 receives a Bye
Passed
40 | P a g e
Call
Forwarding
No Answer-
RNA Timer
PBX User dials *610
Enter 1 #
PSTN calls PBX User with CFNA
Verify Call is forwarded
PBX User 1 Dials *610
Announcement is Heard
PBX User enter 1 for amount of Rings
After announcement completes PBX User 1 receives a Bye
PSTN Calls PBX User 1
PSTN receives ringback
PBX User 1 gets ringing
PBX User 1 receives Caller ID
After timer is RNA is received
PBX User 2 gets ringing
PBX user 2 receives Caller ID
PBX User 2 answers call
2 way Audio
PSTN releases call
PBX User 2 receives a Bye
Call
Forwarding
No Answer
Interrogate
PBX User with CFNA dials
*61*
Announcement received
PBX User 1 Dials *61*
Announcement is Heard
After announcement completes PBX User 1 receives a Bye
Call
Forwarding
No Answer
Deactivate
PBX User with CFNA dials *93
PSTN Calls PBX User
PBX User 1 Dials *93
Announcement is Heard
After announcement completes PBX User 1 receives a Bye
Call
Forwarding
No Answer to
Voicemail
Activate
PBX User Dials *41
PSTN Dials PBX User with CFNA
Verify Call goes to
Voicemail
PBX User 1 Dials *41
Announcement is received
When announcement completes PBX User receives a Bye
Busy PBX User 1
PSTN User Calls PBX User 1
Call should go to voicemail after RNA timer is reached
Announcement is Heard
Leave voicemail
After leaving voicemail PSTN should receive a Bye
PBX User 1 should receive and MWI
Passed
Passed
Passed
Passed
41 | P a g e
Call
Forwarding
Not
Reachable
Activate
PBX User dials *94
Enter the CFNR
Destination TN
Unregister Pilot TNs
PSTN calls PBX User with CFNR
Verify Call is forwarded
Register Pilot TNs
PBX User 1 Dials *94
Announcement is heard
PBX User enter PBX User 2 TN
Announcement is heard
PBX Receives a Bye
Unplug SBC Lan Cable
PSTN Calls PBX User 1
PSTN receives ringback
PBX User 1 does not ring
PSTN User 2 gets ringing
PSTN user 2 receives Caller ID (PSTN
Originator Caller)
PSTN User answers call
2 way Audio
PSTN User 1 releases call
PSTN User 2 receives a Bye
Passed
Call
Forwarding
Not
Reachable
Interrogate
PBX User with CFNR dials
*63*
Announcement received
PBX User 1 Dials *63*
Announcement is Heard
After announcement completes PBX User 1 receives a Bye
Call
Forwarding
Not
Reachable
Deactivate
PBX User with CFNR dials *95
PSTN Calls PBX User
PBX User 1 Dials *95
Announcement is Heard
After announcement completes PBX User 1 receives a Bye
Passed
Passed
Call Return by PBX User
PBX User dials *69
PSTN 1 Calls PBX User 1
PSTN 1 receives ringback
PBX User 1 receives ringing
PBX User 1 receives caller ID
PBX User 1 answers call
2 way Audio
PSTN 1 ends the call
PBX User 1 receives a Bye
PBX User 1 Dials *69
PBX User receives Ringback
PSTN 1 receives Ringing
PSTN receives Caller ID
PSTN answers
2 way Audio
PSTN releases call
PBX User 1 receives a Bye
Passed
42 | P a g e
Consultative
Transfer with
SIP REFER
PBX User Calls PSTN
PBX User transfers PSTN to PSTN2
PBX User has Audio with
PSTNs
PSTN 1 has MOH
PBX User Transfers Call
PSTN and PSTN2 now have audio
Unattended
Transfer with
SIP REFER
PBX User Calls PSTN
PBX User transfers PSTN to PSTN2
During Ringback PBX
User transfers
PSTN 1 has MOH
PSTN2 answers call
PSTN and PSTN2 now have audio
Passed
Passed
Consultative
Transfer
PBX User Calls PSTN
PBX User transfers PSTN to PBX User 2
PBX User 1 has Audio with PBX User 2
PSTN 1 has MOH
PBX User Transfers Call
PSTN and PBX 2 now have audio
PBX User 1 Calls PSTN
PBX User receives Ringback
PSTN 1 receives Ringing
PSTN 1 receives Caller ID
PSTN 1 answers
2 way Audio
PBX User transfers call to PBX User 2
PSTN User gets MOH
PBX User 1 gets Dial tone
PBX User 1 dials PBX User 2 Extension
PBX User 1 receives Ringback
PBX User 2 receives Ringing
PBX User 2 receives Caller ID of PBX User
1
PBX User 2 answers the Call
2 way Audio
PBX User 1 transfers the call
MOH Ends
PSTN 1 and PBX User 2 are now connected
2 Way Audio
PSTN 1 Ends the call
PBX User 2 receives the Bye
Passed
43 | P a g e
Unattended
Transfer
PBX User Calls PSTN
PBX User transfers PSTN to PBX User 2
During Ringback PBX
User transfers
PSTN 1 has MOH
PBX User 2 answers call
PSTN and PBX User 2 now have audio
PBX User 1 Calls PSTN
PBX User receives Ringback
PSTN 1 receives Ringing
PSTN 1 receives Caller ID
PSTN 1 answers
2 way Audio
PBX User transfers call to PBX User 2
PSTN User gets MOH
PBX User 1 gets Dial tone
PBX User 1 dials PBX User 2 Extension
PBX User 1 receives Ringback
PBX User 2 receives Ringing
PBX User 2 receives Caller ID of PSTN 1
PBX User 1 release call
PBX User 2 answers the Call
MOH Ends
2 way Audio
PSTN 1 release the call
PBX User 2 receives the Bye
Passed
Call Waiting
Persistent
Activate
PBX User dials *43
PSTN Calls PBX User
PSTN 2 Calls PBX User
Verify Call Waiting Tone
PBX User 1 Dials *43
Announcement is heard
PBX Receives a Bye
PSTN User 1 Calls PBX User 1
PSTN User 1 receives ringback
PBX User 1 gets ringing
PBX user 1 receives Caller ID
PBX User answers call
2 way Audio
PSTN User 2 Calls PBX User 1
PSTN User 2 receives ringback
PBX User 1 receives caller ID
PBX User 1 hear Call Waiting Tone
PBX User Places PSTN User 1 on Hold
PSTN User 1 hears MOH
PBX User 1 answers Call from PSTN 2
2 way Audio
Verify PBX User 1 can swap between to callers
While on PBX User 1 and PSTN User 1
PSTN 1 releases Call
PBX User 1 receives a Bye
Call 2 should still be up with PSTN 2 hearing MOH
Passed
44 | P a g e
Customer
Originated
Trace
PSTN Calls PBX User
PBX User Answers the
Call
PBX User Hangs up call
PBX User enters *57
Verify announcement
PSTN Calls PBX User 1
PSTN receives ringback
PBX User 1 gets ringing
PBX user 1 receives Caller ID
PBX User answers call
2 way Audio
PSTN releases call
PBX User 2 receives a Bye
PBX User 1 Dial *57
Announcement received
Announcement Completes PBX User receives a Bye
Enhanced
Call Logs
Log into portal and verify
Call logs
Log into the portal for PBX User 1
On main screen verify calls Logs are displayed
Missed
Received
Placed
Last Number
Redial
PBX User dials *66
The last number dialed should be called
PSTN Calls PBX User 1
PSTN receives ringback
PBX User 1 gets ringing
PBX user 1 receives Caller ID
PBX User answers call
2 way Audio
PSTN releases call
PBX User 1 receives a Bye
PBX User 1 Dial *66
PSTN receives ringback
PBX User 1 gets ringing
PBX user 1 receives Caller ID
PBX User answers call
2 way Audio
PSTN releases call
PBX User 1 receives a Bye
Passed
Passed
Passed
45 | P a g e
MOH
Verify MOH for conference, transfer, and hold
PSTN Calls PBX User 1
PSTN receives ringback
PBX User 1 gets ringing
PBX user 1 receives Caller ID
PBX User answers call
2 way Audio
PBX User 1 Places call on Hold
PSTN receives MOH
PBX User retrieves call from Hold
2 way Audio
PSTN releases call
PBX User 2 receives a Bye
Passed
Remote
Office - Like
CFA
Provision Remote office for a SIP Trunk user on the BroadWorks portal to use PSTN number A.
Place a call from a PSTN number B to the SIP
Trunk user's DID and verify that it is forwarded to PSTN number A (the destination configured in
BroadWorks).
Log into the portal for PBX User 1
Set up remote Office to PSTN Number 1
PSTN User 2 Calls PBX User 1
PSTN 2 receives ringback
PSTN User 1 gets ringing with PSTN 2
Caller ID and Diversion header for PBX
User1
PSTN User 1 answers call
2 way Audio
PSTN 1 releases call
Passed
Remote
Office - Quick
Call
Provision Remote office for a SIP Trunk user 1 on the BroadWorks portal to use PSTN number A. On the BW portal, Manage
Users, select Configure
Features of User 1, under
Quick Call, add PSTN B number and click on the
Call Button. PSTN A should Start Ringing with
PBX User 1 Caller ID.
Log into the portal for PBX User 1
Set up remote Office to PSTN Number 1
Initiate a Quack Call to PSTN 2 on the portal
PSTN User 1 gets ringing with PBX User 1
Caller ID
PSTN user 1 answers the call.
Now PSTN2 should start ringing with PBX
User1 as Caller ID.
PSTN 1 might hear ringback based on how long PSTN 2 rings.
PSTN 2 answers the call
2 way Audio
PSTN 1 releases call
Passed
46 | P a g e
Remote
Office - Click to Call
Provision Remote office for a SIP Trunk user 1 on the BroadWorks portal to use PSTN number A. On the BW portal, Manage
Users, select Configure
Features of User 1, under
Call Logs, select either incoming/outgoing/missed calls and Click on a Call under "Phone Number
Click To call" column.
PSTN A should Start
Ringing with PBX User 1
Caller ID.
Log into the portal for PBX User 1
Set up remote Office to PSTN Number 1
Review call logs and identify a call log that needs to be called via Click to Call.
Click on the identified call log under "Click to Call"
PSTN User 1 gets ringing with PBX User 1
Caller ID
PSTN user 1 answers the call.
Now PSTN2 should start ringing with PBX
User1 as Caller ID.
PSTN 1 might hear ringback based on how long PSTN 2 rings.
PSTN 2 answers the call
2 way Audio
PSTN 1 releases call
Passed
Selective Call
Acceptance
Provision selective call acceptance in the
BroadWorks portal. Place a call from an accepted
TN to the SIP Trunk User.
Verify that the call completes normally.
Place a call from a TN that is not on the accept list and verify that the call does not reach the SBC.
Log into the portal for PBX User 1
Set up Selected Call Acceptance to PSTN
Number 1
PSTN Calls PBX User 1
PSTN receives ringback
PBX User 1 gets ringing
PBX user 1 receives Caller ID
PBX User answers call
2 way Audio
PSTN releases Call
PBX User 1 receives a Bye
Selective Call
Rejection
Provision selective call rejection in the
BroadWorks portal. Place a call from a TN not on the reject list to the SIP
Trunk User. Verify that the call completes normally. Place a call from a TN that is on the reject list and verify that the call does not reach the SBC.
Log into the portal for PBX User 1
Set up Selected Call rejection to PSTN
Number 1
PSTN Calls PBX User 1
Verify PSTN gets an announcement
PSTN receives a Bye
Passed
Passed
47 | P a g e
Sequential
Ring
Provision sequential ring in the BroadWorks portal.
Place a call to the SIP trunk user. Verify that the numbers in the sequential ring list are dialed in order.
Log into the Portal for PBX User 1
Set up Sequential Ring with PBX User 2 and PBX User 3
PSTN Calls PBX User 1
PSTN receives ringback
PBX User 1 gets ringing
PBX user 1 receives Caller ID
After RNA Timer PBX User 1 receives a
Cancel
PBX User 2 gets ringing
PBX user 2 receives Caller ID
After RNA Timer PBX User 1 receives a
Cancel
PBX User 3 gets ringing
PBX user 3 receives Caller ID
PBX User 3 answers call
2 way Audio
PSTN releases Call
PBX User 3 receives a Bye
Passed
Simultaneous
Ring
Provision Simultaneous ring in the BroadWorks portal. Place a call to the
SIP trunk user. Verify that the numbers in the
Simultaneous ring list are dialed at once.
Log into the Portal for PBX User 1
Set up Simultaneous Ring with PBX User 2 and PBX User 3
PSTN Calls PBX User 1
PSTN receives ringback
PBX User 1 gets ringing
PBX user 1 receives Caller ID
PBX User 2 gets ringing
PBX user 2 receives Caller ID
PBX User 3 gets ringing
PBX user 3 receives Caller ID
PBX User 3 Answers Call
PBX User 1 and 2 receive a Cancel
2 way Audio
PSTN releases Call
PSTN User 3 receives a Bye
Passed
48 | P a g e
PBX Initiate
Conference
PBX User Calls PSTN
PBX User Conferences
PBX User 2
PBX User 1 Calls PSTN
PBX User receives Ringback
PSTN 1 receives Ringing
PSTN 1 receives Caller ID
PSTN 1 answers
2 way Audio
PBX User conferences call to PBX User 2
PSTN User gets MOH
PBX User 1 gets Dial tone
PBX User 1 dials PBX User 2 Extension
PBX User 1 receives Ringback
PBX User 2 receives Ringing
PBX User 2 receives Caller ID of PBX User
1
PBX User 2 answers the Call
2 way Audio
PBX User 1 conferences the call
MOH Ends
PSTN 1, PBX User 1 and PBX User 2 are now connected
2 Way Audio
PBX User 1 Ends the call
PBX User 2 and PSTN receives the Bye
Passed
49 | P a g e
PSTN Initiate
Conference
PBX User calls PSTN
PSTN conferences PBX
User2
"PBX User 1 Calls PSTN
PBX User receives Ringback
PSTN 1 receives Ringing
PSTN 1 receives Caller ID
PSTN 1 answers
2 way Audio
PSTN User 1 conferences call to PBX User
2
PBX User 1 gets MOH
PSTN User 1 gets Dial tone
PSTN User 1 dials PBX User 2 Extension
PSTN User 1 receives Ringback
PBX User 2 receives Ringing
PBX User 2 receives Caller ID of PSTN
User 1
PBX User 2 answers the Call
2 way Audio
PSTN User 1 conferences the call
MOH Ends
PSTN 1, PBX User 1 and PBX User 2 are now connected
2 Way Audio
PSTN User 1 Ends the call
PBX User 1 and PBX User 2 Still Have
Audio
PBX User 1 End the Call
PBX User 2 receives a Bye
Passed
Huntgroup
Sim Ring
PSTN calls Huntgroup
Sim ring 3 members
Answer Call
Log into Admin Portal
Create Huntgroup with 3 members with
Sequential Ring
PSTN Calls Huntgroup
PSTN receives ringback
PBX User 1 gets ringing
PBX user 1 receives Caller ID
PBX User 2 gets ringing
PBX user 2 receives Caller ID
PBX user 3 receives Caller ID
PBX User 3 Answers the call
PBX User 3 Answers the Call
2 way Audio
PSTN ends the call
PBX User 2 receives a Bye
Pass
50 | P a g e
PBX to PBX
PBX User Calls PBX
User2 Same Trunk
Verify RTP is dropped to
SBC
PBX User 1 Calls PBX User 2
PBX User 1 receives ringback
PBX User 2 gets ringing
PBX user 2 receives Caller ID
PBX User 2 answers call
2 way Audio RTP is on SBC/PBX
PBX User 1 End the call
PBX User 2 receives a Bye
PSTN to PBX PSTN to PBX User
PSTN User 1 Calls PBX User 1
PSTN User 1 receives ringback
PBX User 1 gets ringing
PBX user 1 receives Caller ID
PBX User answers call
2 way Audio
PSTN User Ends The Call
PBX User 1 receives a Bye
PBX to PSTN PBX User to PSTN
PBX User 1 Calls PSTN User 1
PBX User 1 receives ringback
PSTN User 1 gets ringing
PSTN user 1 receives Caller ID
PSTN User answers call
2 way Audio
PSTN User Ends The Call
PBX User 1 receives a Bye
PBX to PSTN
- Packet
Marking for
SIG packets
PBX to PSTN Call to verify that signaling packets are marked with
DSCP = 24 (0x18)
All outgoing SIP Signaling packets are marked with DSCP=24
PBX to PSTN
- Packet
Marking for
RTP packets
PBX to PSTN Call to verify that rtp packets are marked with DSCP = 40
(0x28)
All outgoing SIP RTP packets are marked with DSCP=40
PBX to PSTN
- Directory assistance
PBX User Calls PBX 411 and speaks with directory assistant
PBX to PSTN
- Toll Free
PBX User Calls
800.366.8201 to test toll free numbers
PBX User 1 dials 411
Call is delivered to Directory Assistant for enquiry
Once the user hears an announcement or speaks with an operator, PBX user hangs up the call
PBX User 1 dials 800.366.8201 (CTL
Support)
Call is delivered to CenturyLink Support
Once the user hears an announcement or speaks with an operator, PBX user hangs up the call
Passed
Passed
Passed
Passed
Passed
Passed
Passed
51 | P a g e
Anonymous
Call
Rejection
Activate
PBX User dials *77
PSTN Calls PBX User with Caller ID Block
Should receive an announcement
*77 is Dialed PBX and leaves PBX Phones gets an announcement
Calling Party blocks caller ID
Calling party makes a call to PBX User
Calling Party receives an announcement when PBX user is dialed
Passed
Anonymous
Call
Rejection
Deactivate
PBX User dials *87
PSTN Calls PBX User with Caller ID block
Call Should Complete
*87 is dialed PBX User receives and announcement
PSTN calls PBX User
PSTN Phone receives ringback
PBX Phone gets ringing
PBX Phone get Caller ID
PBX Phone answer the Call
2 way audio is received
PBX Phone releases Calls
PSTN receives a Bye
Alien TNs
A call PBX call originate where the from TN that is not part of the customer trunk group. As long as the pilot number is identified in outgoing call by PAI, the BroadWorks will accept and route the call.
After Alien TN is set up on a Trunk in
CenturyLink Network
PBX User Places a Call to PSTN
PBX User receives ringback
PSTN receives ringing
PSTN receives caller id of the Alien TN
PSTN answers the call
2 way audio is received
PBX Phone releases Calls
PSTN receives a Bye
Passed
Anonymous
Call PBX-BW
PBX sends anonymous call to BW
BW delivers the calls
Private or unknown or anonymous to PSTN
PBX is configured to send a call to BW as anonymous with TN as PSTN
BW delivers the call to PSTN as Private or
Anonymous
PSTN phone shows the call as Private or
Anonymous
Call is answered by PSTN
PBX user hangs up the call
Passed
Passed
52 | P a g e
Barge In
Create a Pick Up Group with 2 PBX Users
PSTN Calls PBX User 1
PBX User 2 dials *33
+PBX User Ext
PSTN, User 1, and User
2 should be conf
PSTN calls PBX User 1
PSTN Phone receives ringback
PBX Phone gets ringing
PBX Phone get Caller ID
PBX Phone answer the Call
2 way audio is received
PBX User 2 Dials *33 + PBX User 1
Extension
PSTN, PBX User 1, and PBX User 2 are conferenced together
2 Way Audio is heard by all Legs
PBX User 1 drops from Call
2 way Audio is heard by PSTN and PBX
User 2
PSTN drops call
PBX User 2 receives a Bye
Barge In
Exempt
In the Portal Enable
Barge In Exempt
Create a Pick Up Group with 2 PBX Users
PSTN Calls PBX User 1
PBX User 2 dials *33
+PBX User Ext
User 2 Should not be conf
Barge in Exempt is set on PBX user 1
PSTN calls PBX User 1
PSTN Phone receives ringback
PBX Phone gets ringing
PBX Phone get Caller ID
PBX Phone answer the Call
2 way audio is received
PBX User 2 Dials *33 + PBX User 1
Extension
PBX user 2 is not allowed to barge in
PSTN drops the call
PBX User 1 receives a Bye
Passed
Passed
PSTN to
BWA
PSTN calls BWA Number
Enter Calling Number
(2nd Phone Location)
Enter Called Number
(PSTN)
PSTN should Ring with
Caller ID of 2nd Phone
Answer Call
BroadWorks Anywhere is set up in Portal
PSTN 1 Calls BWA Number
Announcement is received
Enter calling Number (2nd Phone created in BWA)
Announcement received
Enter Called Number (PSTN 2)
PSTN 1 receives ringback
PSTN 2 receives ringing
PSTN 2 receives caller ID of 2nd Phone
(Not of PSTN 1)
PSTN 2 Answers Call
2 way audio is received
PSTN 2 releases Calls
PSTN receives a Bye
Passed
53 | P a g e
PSTN to PBX user with
BWA
PSTN Calls User with
BWA
PBX User and 2nd
Location should Ring
Answer phone for 2nd location
BroadWorks Anywhere is set up in Portal
PSTN 1 Calls BWA Number
PSTN 1 receives ringback
Both PBX User and 2nd Phone Location
Number gets ringing
Both PBX User and 2nd Phone Location
Number gets Caller ID of PSTN
Call is answered on Location 2
PBX User no longer gets ringing (cancel)
2 way Audio
Location 2 releases call
PSTN receives a Bye
Passed
Call
Forwarding
Always
Activate
PBX User dials *72
Enter the CFA
Destination TN
PSTN calls PBX User with CFA
PBX User 1 Dials *72
Announcement is heard
PBX User enter PBX User 2 TN
Announcement is heard
PBX Receives a Bye
PSTN Calls PBX User 1
PSTN receives ringback
PBX User 1 does not ring
PBX User 2 gets ringing
PBX user 2 receives Caller ID (PSTN
Originator Caller)
PBX User answers call
2 way Audio
PSTN releases call
PBX User 2 receives a Bye
Passed
Call
Forwarding
Always
Interrogate
PBX User with CFA dials
*21*
Announcement received
PBX User 1 Dials *21*
Announcement is Heard
After announcement completes PBX User 1 receives a Bye
Call
Forwarding
Always
Deactivate
PBX User with CFA dials
*73
PSTN Calls PBX User
PBX User 1 Dials *73
Announcement is Heard
After announcement completes PBX User 1 receives a Bye
Passed
Passed
54 | P a g e
PSTN with
Privacy call to PBX is
CFA to PSTN
PBX User is configured with CFA to PSTN 2
PSTN 1 Calls PBX with
Caller ID Restricted
PSTN 1 hears ring back
PBX sends a new call to
BW with PSTN 2 Number,
From as Anonymous and
PAI set to Pilot Number
BW forwards the call to
PSTN2
PSTN 2 hears ringing
PSTN 2 Caller ID displays
Pilot Number
PSTN 2 Answers the call.
Two way voice path is established between
PSTN 1 and PSTN 2
PSTN 2 hangs up
Pilot Number should be shown as CLID on
PSTN2
Passed
PSTN call is
CFB to PSTN with ID
Restricted
PBX configured to send
CFB to BW for identified
Station.
BW is configured with
CFB to PSTN2.
PSTN 1 Calls PBX with
Caller ID Restricted
PSTN 1 hears ring back
PBX send 486 Busy to
BW
BW forwards the call to
PSTN2
PSTN 2 hears ringing
PSTN 2 Caller ID displays
Private/Anonymous
PSTN 2 Answers the call.
Two way voice path is established between
PSTN 1 and PSTN 2
PSTN 2 hangs up
PSTN2 should receive Private/Anonymous as CLID
Passed
55 | P a g e
Call
Forwarding
Busy Activate
PBX User dials *90
Enter the CFB
Destination TN
PSTN calls PBX User with CFB
PBX User 1 Dials *90
Announcement is heard
PBX User enter PBX User 2 TN
Announcement is heard
PBX Receives a Bye
Busy PBX User 1
PSTN Calls PBX User 1
PSTN receives ringback
PBX User 1 does not ring
PBX User 2 gets ringing
PBX user 2 receives Caller ID (PSTN
Originator Caller)
PBX User answers call
2 way Audio
PSTN releases call
PBX User 2 receives a Bye
Passed
Call
Forwarding
Busy
Interrogate
PBX User with CFB dials
*67*
Announcement received
PBX User 1 Dials *67*
Announcement is Heard
After announcement completes PBX User 1 receives a Bye
Call
Forwarding
Busy
Deactivate
PBX User with CFB dials
*91
PSTN Calls PBX User
PBX User 1 Dials *91
Announcement is Heard
After announcement completes PBX User 1 receives a Bye
Passed
Passed
Call
Forwarding
No Answer
Activate
PBX User dials *92
Enter the CFNA
Destination TN
PSTN calls PBX User with CFNA
PBX User 1 Dials *92
Announcement is heard
PBX User enters PBX User 2 TN
Announcement is heard
PBX Receives a Bye
PSTN Calls PBX User 1
PSTN receives ringback
PBX User 1 gets ringing
PBX User 1 receives Caller ID
After timer is RNA is received
PBX User 2 gets ringing
PBX user 2 receives Caller ID
PBX User 2 answers call
2 way Audio
PSTN releases call
PBX User 2 receives a Bye
Passed
56 | P a g e
Call
Forwarding
No Answer-
RNA Timer
PBX User dials *610
Enter 1 #
PSTN calls PBX User with CFNA
Verify Call is forwarded
PBX User 1 Dials *610
Announcement is Heard
PBX User enter 1 for amount of Rings
After announcement completes PBX User 1 receives a Bye
PSTN Calls PBX User 1
PSTN receives ringback
PBX User 1 gets ringing
PBX User 1 receives Caller ID
After timer is RNA is received
PBX User 2 gets ringing
PBX user 2 receives Caller ID
PBX User 2 answers call
2 way Audio
PSTN releases call
PBX User 2 receives a Bye
Passed
Call
Forwarding
No Answer
Interrogate
PBX User with CFNA dials
*61*
Announcement received
PBX User 1 Dials *61*
Announcement is Heard
After announcement completes PBX User 1 receives a Bye
Call
Forwarding
No Answer
Deactivate
PBX User with CFNA dials *93
PSTN Calls PBX User
PBX User 1 Dials *93
Announcement is Heard
After announcement completes PBX User 1 receives a Bye
Passed
Passed
Call
Forwarding
No Answer to
Voicemail
Activate
PBX User Dials *41
PSTN Dials PBX User with CFNA
Verify Call goes to
Voicemail
PBX User 1 Dials *41
Announcement is received
When announcement completes PBX User receives a Bye
Busy PBX User 1
PSTN User Calls PBX User 1
Call should go to voicemail after RNA timer is reached
Announcement is Heard
Leave voicemail
After leaving voicemail PSTN should receive a Bye
PBX User 1 should receive and MWI
Passed
57 | P a g e
Call
Forwarding
Not
Reachable
Activate
PBX User dials *94
Enter the CFNR
Destination TN
Unregister Pilot TNs
PSTN calls PBX User with CFNR
Verify Call is forwarded
Register Pilot TNs
PBX User 1 Dials *94
Announcement is heard
PBX User enter PBX User 2 TN
Announcement is heard
PBX Receives a Bye
Unplug SBC Lan Cable
PSTN Calls PBX User 1
PSTN receives ringback
PBX User 1 does not ring
PSTN User 2 gets ringing
PSTN user 2 receives Caller ID (PSTN
Originator Caller)
PSTN User answers call
2 way Audio
PSTN User 1 releases call
PSTN User 2 receives a Bye
Passed
Call
Forwarding
Not
Reachable
Interrogate
PBX User with CFNR dials
*63*
Announcement received
PBX User 1 Dials *63*
Announcement is Heard
After announcement completes PBX User 1 receives a Bye
Call
Forwarding
Not
Reachable
Deactivate
PBX User with CFNR dials *95
PSTN Calls PBX User
PBX User 1 Dials *95
Announcement is Heard
After announcement completes PBX User 1 receives a Bye
Passed
Passed
Call Return by PBX User
PBX User dials *69
PSTN 1 Calls PBX User 1
PSTN 1 receives ringback
PBX User 1 receives ringing
PBX User 1 receives caller ID
PBX User 1 answers call
2 way Audio
PSTN 1 ends the call
PBX User 1 receives a Bye
PBX User 1 Dials *69
PBX User receives Ringback
PSTN 1 receives Ringing
PSTN receives Caller ID
PSTN answers
2 way Audio
PSTN releases call
PBX User 1 receives a Bye
Passed
58 | P a g e
Consultative
Transfer with
SIP REFER
PBX User Calls PSTN
PBX User transfers PSTN to PSTN2
PBX User has Audio with
PSTNs
PSTN 1 has MOH
PBX User Transfers Call
PSTN and PSTN2 now have audio
Unattended
Transfer with
SIP REFER
PBX User Calls PSTN
PBX User transfers PSTN to PSTN2
During Ringback PBX
User transfers
PSTN 1 has MOH
PSTN2 answers call
PSTN and PSTN2 now have audio
Passed
Passed
Consultative
Transfer
PBX User Calls PSTN
PBX User transfers PSTN to PBX User 2
PBX User 1 has Audio with PBX User 2
PSTN 1 has MOH
PBX User Transfers Call
PSTN and PBX 2 now have audio
PBX User 1 Calls PSTN
PBX User receives Ringback
PSTN 1 receives Ringing
PSTN 1 receives Caller ID
PSTN 1 answers
2 way Audio
PBX User transfers call to PBX User 2
PSTN User gets MOH
PBX User 1 gets Dial tone
PBX User 1 dials PBX User 2 Extension
PBX User 1 receives Ringback
PBX User 2 receives Ringing
PBX User 2 receives Caller ID of PBX User
1
PBX User 2 answers the Call
2 way Audio
PBX User 1 transfers the call
MOH Ends
PSTN 1 and PBX User 2 are now connected
2 Way Audio
PSTN 1 Ends the call
PBX User 2 receives the Bye
Passed
59 | P a g e
Unattended
Transfer
PBX User Calls PSTN
PBX User transfers PSTN to PBX User 2
During Ringback PBX
User transfers
PSTN 1 has MOH
PBX User 2 answers call
PSTN and PBX User 2 now have audio
PBX User 1 Calls PSTN
PBX User receives Ringback
PSTN 1 receives Ringing
PSTN 1 receives Caller ID
PSTN 1 answers
2 way Audio
PBX User transfers call to PBX User 2
PSTN User gets MOH
PBX User 1 gets Dial tone
PBX User 1 dials PBX User 2 Extension
PBX User 1 receives Ringback
PBX User 2 receives Ringing
PBX User 2 receives Caller ID of PSTN 1
PBX User 1 release call
PBX User 2 answers the Call
MOH Ends
2 way Audio
PSTN 1 release the call
PBX User 2 receives the Bye
Call Waiting
Persistent
Activate
PBX User dials *43
PSTN Calls PBX User
PSTN 2 Calls PBX User
Verify Call Waiting Tone
PBX User 1 Dials *43
Announcement is heard
PBX Receives a Bye
PSTN User 1 Calls PBX User 1
PSTN User 1 receives ringback
PBX User 1 gets ringing
PBX user 1 receives Caller ID
PBX User answers call
2 way Audio
PSTN User 2 Calls PBX User 1
PSTN User 2 receives ringback
PBX User 1 receives caller ID
PBX User 1 hear Call Waiting Tone
PBX User Places PSTN User 1 on Hold
PSTN User 1 hears MOH
PBX User 1 answers Call from PSTN 2
2 way Audio
Verify PBX User 1 can swap between to callers
While on PBX User 1 and PSTN User 1
PSTN 1 releases Call
PBX User 1 receives a Bye
Call 2 should still be up with PSTN 2 hearing MOH
Customer
Originated
Trace
PSTN Calls PBX User
PBX User Answers the
Call
PBX User Hangs up call
PBX User enters *57
Verify announcement
PSTN Calls PBX User 1
PSTN receives ringback
PBX User 1 gets ringing
PBX user 1 receives Caller ID
PBX User answers call
2 way Audio
PSTN releases call
PBX User 2 receives a Bye
PBX User 1 Dial *57
Announcement received
Announcement Completes PBX User receives a Bye
Passed
Passed
Passed
60 | P a g e
Enhanced
Call Logs
Log into portal and verify
Call logs
Log into the portal for PBX User 1
On main screen verify calls Logs are displayed
Missed
Received
Placed
Last Number
Redial
PBX User dials *66
The last number dialed should be called
PSTN Calls PBX User 1
PSTN receives ringback
PBX User 1 gets ringing
PBX user 1 receives Caller ID
PBX User answers call
2 way Audio
PSTN releases call
PBX User 1 receives a Bye
PBX User 1 Dial *66
PSTN receives ringback
PBX User 1 gets ringing
PBX user 1 receives Caller ID
PBX User answers call
2 way Audio
PSTN releases call
PBX User 1 receives a Bye
Passed
Passed
MOH
Verify MOH for conference, transfer, and hold
PSTN Calls PBX User 1
PSTN receives ringback
PBX User 1 gets ringing
PBX user 1 receives Caller ID
PBX User answers call
2 way Audio
PBX User 1 Places call on Hold
PSTN receives MOH
PBX User retrieves call from Hold
2 way Audio
PSTN releases call
PBX User 2 receives a Bye
Passed
61 | P a g e
Remote
Office - Like
CFA
Provision Remote office for a SIP Trunk user on the BroadWorks portal to use PSTN number A.
Place a call from a PSTN number B to the SIP
Trunk user's DID and verify that it is forwarded to PSTN number A (the destination configured in
BroadWorks).
Log into the portal for PBX User 1
Set up remote Office to PSTN Number 1
PSTN User 2 Calls PBX User 1
PSTN 2 receives ringback
PSTN User 1 gets ringing with PSTN 2
Caller ID and Diversion header for PBX
User1
PSTN User 1 answers call
2 way Audio
PSTN 1 releases call
Passed
Remote
Office - Quick
Call
Provision Remote office for a SIP Trunk user 1 on the BroadWorks portal to use PSTN number A. On the BW portal, Manage
Users, select Configure
Features of User 1, under
Quick Call, add PSTN B number and click on the
Call Button. PSTN A should Start Ringing with
PBX User 1 Caller ID.
Log into the portal for PBX User 1
Set up remote Office to PSTN Number 1
Initiate a Quick Call to PSTN 2 on the portal
PSTN User 1 gets ringing with PBX User 1
Caller ID
PSTN user 1 answers the call.
Now PSTN2 should start ringing with PBX
User1 as Caller ID.
PSTN 1 might hear ringback based on how long PSTN 2 rings.
PSTN 2 answers the call
2 way Audio
PSTN 1 releases call
Passed
Remote
Office - Click to Call
Provision Remote office for a SIP Trunk user 1 on the BroadWorks portal to use PSTN number A. On the BW portal, Manage
Users, select Configure
Features of User 1, under
Call Logs, select either incoming/outgoing/missed calls and Click on a Call under "Phone Number
Click To call" column.
PSTN A should Start
Ringing with PBX User 1
Caller ID.
Log into the portal for PBX User 1
Set up remote Office to PSTN Number 1
Review call logs and identify a call log that needs to be called via Click to Call.
Click on the identified call log under "Click to Call"
PSTN User 1 gets ringing with PBX User 1
Caller ID
PSTN user 1 answers the call.
Now PSTN2 should start ringing with PBX
User1 as Caller ID.
PSTN 1 might hear ringback based on how long PSTN 2 rings.
PSTN 2 answers the call
2 way Audio
PSTN 1 releases call
Passed
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Selective Call
Acceptance
Provision selective call acceptance in the
BroadWorks portal. Place a call from an accepted
TN to the SIP Trunk User.
Verify that the call completes normally.
Place a call from a TN that is not on the accept list and verify that the call does not reach the SBC.
Log into the portal for PBX User 1
Set up Selected Call Acceptance to PSTN
Number 1
PSTN Calls PBX User 1
PSTN receives ringback
PBX User 1 gets ringing
PBX user 1 receives Caller ID
PBX User answers call
2 way Audio
PSTN releases Call
PBX User 1 receives a Bye
Selective Call
Rejection
Provision selective call rejection in the
BroadWorks portal. Place a call from a TN not on the reject list to the SIP
Trunk User. Verify that the call completes normally. Place a call from a TN that is on the reject list and verify that the call does not reach the SBC.
Log into the portal for PBX User 1
Set up Selected Call rejection to PSTN
Number 1
PSTN Calls PBX User 1
Verify PSTN gets an announcement
PSTN receives a Bye
Passed
Passed
Sequential
Ring
Provision sequential ring in the BroadWorks portal.
Place a call to the SIP trunk user. Verify that the numbers in the sequential ring list are dialed in order.
Log into the Portal for PBX User 1
Set up Sequential Ring with PBX User 2 and PBX User 3
PSTN Calls PBX User 1
PSTN receives ringback
PBX User 1 gets ringing
PBX user 1 receives Caller ID
After RNA Timer PBX User 1 receives a
Cancel
PBX User 2 gets ringing
PBX user 2 receives Caller ID
After RNA Timer PBX User 1 receives a
Cancel
PBX User 3 gets ringing
PBX user 3 receives Caller ID
PBX User 3 answers call
2 way Audio
PSTN releases Call
PBX User 3 receives a Bye
Passed
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Simultaneous
Ring
Provision Simultaneous ring in the BroadWorks portal. Place a call to the
SIP trunk user. Verify that the numbers in the
Simultaneous ring list are dialed at once.
Log into the Portal for PBX User 1
Set up Simultaneous Ring with PBX User 2 and PBX User 3
PSTN Calls PBX User 1
PSTN receives ringback
PBX User 1 gets ringing
PBX user 1 receives Caller ID
PBX User 2 gets ringing
PBX user 2 receives Caller ID
PBX User 3 gets ringing
PBX user 3 receives Caller ID
PBX User 3 Answers Call
PBX User 1 and 2 receive a Cancel
2 way Audio
PSTN releases Call
PSTN User 3 receives a Bye
PBX Initiate
Conference
PBX User Calls PSTN
PBX User Conferences
PBX User 2
PBX User 1 Calls PSTN
PBX User receives Ringback
PSTN 1 receives Ringing
PSTN 1 receives Caller ID
PSTN 1 answers
2 way Audio
PBX User conferences call to PBX User 2
PSTN User gets MOH
PBX User 1 gets Dial tone
PBX User 1 dials PBX User 2 Extension
PBX User 1 receives Ringback
PBX User 2 receives Ringing
PBX User 2 receives Caller ID of PBX User
1
PBX User 2 answers the Call
2 way Audio
PBX User 1 conferences the call
MOH Ends
PSTN 1, PBX User 1 and PBX User 2 are now connected
2 Way Audio
PBX User 1 Ends the call
PBX User 2 and PSTN receives the Bye
Passed
Passed
64 | P a g e
PSTN Initiate
Conference
PBX User calls PSTN
PSTN conferences PBX
User2
PBX User 1 Calls PSTN
PBX User receives Ringback
PSTN 1 receives Ringing
PSTN 1 receives Caller ID
PSTN 1 answers
2 way Audio
PSTN User 1 conferences call to PBX User
2
PBX User 1 gets MOH
PSTN User 1 gets Dial tone
PSTN User 1 dials PBX User 2 Extension
PSTN User 1 receives Ringback
PBX User 2 receives Ringing
PBX User 2 receives Caller ID of PSTN
User 1
PBX User 2 answers the Call
2 way Audio
PSTN User 1 conferences the call
MOH Ends
PSTN 1, PBX User 1 and PBX User 2 are now connected
2 Way Audio
PSTN User 1 Ends the call
PBX User 1 and PBX User 2 Still Have
Audio
PBX User 1 End the Call
PBX User 2 receives a Bye
Huntgroup
Sim Ring
PSTN calls Huntgroup
Sim ring 3 members
Answer Call
Log into Admin Portal
Create Huntgroup with 3 members with
Sequential Ring
PSTN Calls Huntgroup
PSTN receives ringback
PBX User 1 gets ringing
PBX user 1 receives Caller ID
PBX User 2 gets ringing
PBX user 2 receives Caller ID
PBX user 3 receives Caller ID
PBX User 3 Answers the call
PBX User 3 Answers the Call
2 way Audio
PSTN ends the call
PBX User 2 receives a Bye
Passed
Passed
PBX to PBX
PBX User Calls PBX
User2 Same Trunk
Verify RTP is dropped to
SBC
PBX User 1 Calls PBX User 2
PBX User 1 receives ringback
PBX User 2 gets ringing
PBX user 2 receives Caller ID
PBX User 2 answers call
2 way Audio RTP is on SBC/PBX
PBX User 1 End the call
PBX User 2 receives a Bye
Passed
65 | P a g e
PSTN to PBX PSTN to PBX User
PSTN User 1 Calls PBX User 1
PSTN User 1 receives ringback
PBX User 1 gets ringing
PBX user 1 receives Caller ID
PBX User answers call
2 way Audio
PSTN User Ends The Call
PBX User 1 receives a Bye
PBX to PSTN PBX User to PSTN
PBX User 1 Calls PSTN User 1
PBX User 1 receives ringback
PSTN User 1 gets ringing
PSTN user 1 receives Caller ID
PSTN User answers call
2 way Audio
PSTN User Ends The Call
PBX User 1 receives a Bye
PBX to PSTN
- Packet
Marking for
SIG packets
PBX to PSTN Call to verify that signaling packets are marked with
DSCP = 24 (0x18)
All outgoing SIP Signaling packets are marked with DSCP=24
PBX to PSTN
- Packet
Marking for
RTP packets
PBX to PSTN Call to verify that rtp packets are marked with DSCP = 40
(0x28)
All outgoing SIP RTP packets are marked with DSCP=40
PBX to PSTN
- Directory assistance
PBX User Calls PBX 411 and speaks with directory assistant
PBX to PSTN
- Toll Free
PBX User Calls
800.366.8201 to test toll free numbers
PBX User 1 dials 411
Call is delivered to Directory Assistant for enquiry
Once the user hears an announcement or speaks with an operator, PBX user hangs up the call
PBX User 1 dials 800.366.8201 (CTL
Support)
Call is delivered to CenturyLink Support
Once the user hears an announcement or speaks with an operator, PBX user hangs up the call
Passed
Passed
Passed
Passed
Passed
Passed
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Configuration for Dual trunk registration and test plan
In the dual trunk architecture, the ESBC registers to a pair of geo-redundant CTL carriers SBC. The architecture for dual trunk registration is as follows:
CORE-Microsoft SFB
Lync client 2
192.168.4.142
Ph no: 4695737005
Lync client 1
192.168.4.141
Ph no: 4695737006
Server 1
Domain Controller
192.168.4.150
Lync server
192.168.4.151
Exchange server
192.168.4.155
Server 2
Mediation server 1
192.168.4.152
Mediation server 2
192.168.4.153
Monitoring server
192.168.2.194
s0p0
192.168.4.135
s0p0
192.168.4.136
4600 Series
VPN to trunk s0p1
192.168.1.220
s0p1
192.168.1.221
PEER – CenturyLink
Trunk
Sip Trunk
192.168.1.170
PSTN
Ph no: 7813282518
Sip Trunk
192.168.1.171
This architecture consists of a single SBC with two realms on each side of the SBC. Two realms and two sip-interfaces on each side replicate two trunks on each side. This test plan, like the single trunk registration, was tested for both PCMU and G729.
SBC configuration for Dual Trunk registration using PCMU towards trunk
local-policy
from-address *
to-address *
source-realm ctl-trunk2
policy-attribute
next-hop sag:SFBmed
realm core2 local-policy
from-address *
to-address *
source-realm core
policy-attribute
next-hop 192.168.1.170
realm ctl-trunk local-policy
from-address *
to-address SFBMed1.partnersfb.com
source-realm core
policy-attribute
next-hop SFBMed1.partnersfb.com
realm core
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action replace-uri local-policy
from-address *
to-address SFBMed2.partnersfb.com
source-realm core
policy-attribute
next-hop SFBMed2.partnersfb.com
realm core
action replace-uri local-policy
from-address 4695550016
to-address *
source-realm core
policy-attribute
next-hop 192.168.1.170
realm ctl-trunk local-policy
from-address *
to-address *
source-realm core2
policy-attribute
next-hop 192.168.1.171
realm ctl-trunk2 local-policy
from-address *
to-address SFBMed1.partnersfb.com
source-realm core2
policy-attribute
next-hop SFBMed1.partnersfb.com
realm core2
action replace-uri local-policy
from-address *
to-address SFBMed2.partnersfb.com
source-realm core2
policy-attribute
next-hop SFBMed2.partnersfb.com
realm core2
action replace-uri local-policy
from-address 4695737015
to-address *
source-realm core2
policy-attribute
next-hop 192.168.1.171
realm ctl-trunk2 local-policy
from-address *
to-address *
source-realm ctl-trunk
policy-attribute
next-hop medpool.partnersfb.com
realm core media-manager media-policy
name voip-default
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tos-settings
media-type audio
tos-value 0x28
tos-settings
media-type message
media-sub-type sip
tos-value 0x18 network-interface
name s0p0
ip-address 192.168.1.220
netmask 255.255.255.0
gateway 192.168.1.1
hip-ip-list 192.168.1.220
192.168.1.221
icmp-address 192.168.1.220
192.168.1.221 network-interface
name s1p0
hostname attsbc.partnersfb.com
ip-address 192.168.4.135
netmask 255.255.255.0
gateway 192.168.4.1
dns-ip-primary 192.168.4.150
dns-domain partnersfb.com
hip-ip-list 192.168.4.135
192.168.4.136
icmp-address 192.168.4.135
192.168.4.136 network-interface
name s1p1
ip-address 192.168.3.220
netmask 255.255.255.0
gateway 192.168.3.1
hip-ip-list 192.168.3.220
icmp-address 192.168.3.220 phy-interface
name s0p0
operation-type Media phy-interface
name s1p0
operation-type Media
slot 1 phy-interface
name s1p1
operation-type Media
port 1
slot 1 playback-config
name transferrbt
entry
encoding PCMU
filename US_ringbackPCMU.raw realm-config
identifier ctl-trunk2
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network-interfaces s0p0:0
mm-in-realm enabled
qos-enable enabled
media-policy voip-default realm-config
identifier core
network-interfaces s1p0:0
mm-in-realm enabled
qos-enable enabled
restricted-latching sdp realm-config
identifier core2
network-interfaces s1p0:0
mm-in-realm enabled
qos-enable enabled realm-config
identifier ctl-trunk
network-interfaces s0p0:0
mm-in-realm enabled
qos-enable enabled
media-policy voip-default session-agent
hostname 192.168.1.171
ip-address 192.168.1.171
port 5100
realm-id ctl-trunk2
out-translationid stripplus1
out-manipulationid Manipto7015 session-agent
hostname 192.168.1.170
ip-address 192.168.1.170
port 5100
realm-id ctl-trunk
out-translationid stripplus1
out-manipulationid Manipto7016 session-agent
hostname SFBMed1.partnersfb.com
ip-address 192.168.4.152
port 5069
transport-method StaticTCP
realm-id core2
ping-method OPTIONS
ping-interval 30
out-translationid addplus1
in-manipulationid ForEarlyMedia
out-manipulationid outManipToSFB
refer-call-transfer enabled
auth-attributes
auth-realm voip.centurylink.com
username 257570-4695737015
password ********
in-dialog-methods INVITE session-agent
hostname SFBMed2.partnersfb.com
ip-address 192.168.4.153
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port 5069
transport-method StaticTCP
realm-id core2
ping-method OPTIONS
ping-interval 30
out-translationid addplus1
in-manipulationid ForEarlyMedia
out-manipulationid outManipToSFB
refer-call-transfer enabled
auth-attributes
auth-realm voip.centurylink.com
username 257570-4695737015
password ********
in-dialog-methods INVITE session-agent
hostname medpool.partnersfb.com
port 5068
transport-method StaticTCP
realm-id core
ping-method OPTIONS
ping-interval 30
out-translationid addplus1
in-manipulationid ForEarlyMedia
out-manipulationid outManipToSFB
refer-call-transfer enabled
auth-attributes
auth-realm voip.centurylink.com
username 257570-4695550016
password ********
in-dialog-methods INVITE session-group
group-name SFBmed
strategy RoundRobin
dest SFBMed1.partnersfb.com
SFBMed2.partnersfb.com session-group
group-name ctl
dest 192.168.1.171
192.168.1.170 session-translation
id addplus1
rules-calling addplus1
rules-called addplus1 session-translation
id stripplus1
rules-calling stripplus1
rules-called stripplus1 sip-config
home-realm-id core
registrar-domain *
registrar-host *
registrar-port 5060
options inmanip-before-validate
max-udp-length=0
sip-message-len 6000
refer-src-routing enabled
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sip-feature
name 100rel-interworking
realm ctl-trunk
require-mode-inbound Pass
require-mode-outbound Pass sip-interface
realm-id ctl-trunk2
sip-port
address 192.168.1.221
allow-anonymous agents-only sip-interface
realm-id core
sip-port
address 192.168.4.135
transport-protocol TCP
allow-anonymous agents-only
registration-caching enabled
options 100rel-interworking
spl-options playback-on-refer="transferrbt" sip-interface
realm-id core2
sip-port
address 192.168.4.136
transport-protocol TCP
registration-caching enabled
options 100rel-interworking
spl-options playback-on-refer="transferrbt" sip-interface
realm-id ctl-trunk
sip-port
address 192.168.1.220
allow-anonymous agents-only sip-manipulation
name Changeinactosendonly
header-rule
name changeSDP
header-name Content-Type
action manipulate
msg-type request
methods INVITE
element-rule
name inacttosendonly
parameter-name application/sdp
type mime
action find-replace-all
comparison-type pattern-rule
match-value a=inactive
new-value a=sendonly sip-manipulation
name Check_privacy_header
header-rule
name ChechForPrivacy
header-name Privacy
action manipulate
msg-type request
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methods INVITE
header-rule
name OverwriteFromDisplay
header-name From
action manipulate
comparison-type boolean
msg-type request
methods INVITE
match-value $ChechForPrivacy
element-rule
name OverwriteUser
type uri-user
action find-replace-all
new-value anonymous
element-rule
name OverwriteDisplay
type uri-display
action find-replace-all
new-value "\"Anonymous\" " sip-manipulation
name ForEarlyMedia
header-rule
name delsupported
header-name Supported
action delete
msg-type request
methods INVITE
header-rule
name addrequireinINVITE
header-name Require
action add
msg-type request
methods INVITE
new-value 100rel
header-rule
name Fortransfer
header-name From
action sip-manip
new-value Changeinactosendonly sip-manipulation
name Manipto7015
header-rule
name modURI
header-name request-uri
action manipulate
element-rule
name mod2
type uri-host
action replace
match-val-type fqdn
new-value $REMOTE_IP+:+$REMOTE_PORT
header-rule
name DeletePAI
header-name P-Asserted-Identity
action delete
methods INVITE
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header-rule
name ReplacePAI
header-name P-Asserted-Identity
action add
comparison-type pattern-rule
methods INVITE
new-value sip:[email protected]
header-rule
name from_voip_ctl_com
header-name From
action manipulate
methods INVITE
element-rule
name From_add_voip_ctl_com
type uri-host
action replace
new-value voip.centurylink.com
header-rule
name registercontactlocalipport
header-name Contact
action manipulate
msg-type request
methods REGISTER
element-rule
name registercontactlocalip
type uri-host
action replace
match-val-type ip
new-value $LOCAL_IP
element-rule
name registercontactlocalport
type uri-port
action replace
new-value $LOCAL_PORT
header-rule
name fixContact1
header-name Contact
action manipulate
element-rule
name updatecon
type uri-user
action replace
comparison-type pattern-rule
match-value (.*)
new-value $ORIGINAL-^"+1"
header-rule
name fixRoute1
header-name Route
action manipulate
element-rule
name updateroute
type uri-user
action replace
comparison-type pattern-rule
match-value (.*)
new-value $ORIGINAL-^"+1"
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header-rule
name Privacy
header-name From
action sip-manip
new-value Check_privacy_header sip-manipulation
name Manipto7016
header-rule
name modURI
header-name request-uri
action manipulate
element-rule
name mod2
type uri-host
action replace
match-val-type fqdn
new-value $REMOTE_IP+:+$REMOTE_PORT
header-rule
name DeletePAI
header-name P-Asserted-Identity
action delete
methods INVITE
header-rule
name ReplacePAI
header-name P-Asserted-Identity
action add
comparison-type pattern-rule
methods INVITE
new-value sip:[email protected]
header-rule
name from_voip_ctl_com
header-name From
action manipulate
methods INVITE
element-rule
name From_add_voip_ctl_com
type uri-host
action replace
new-value voip.centurylink.com
header-rule
name registercontactlocalipport
header-name Contact
action manipulate
msg-type request
methods REGISTER
element-rule
name registercontactlocalip
type uri-host
action replace
match-val-type ip
new-value $LOCAL_IP
element-rule
name registercontactlocalport
type uri-port
action replace
new-value $LOCAL_PORT
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header-rule
name fixContact2
header-name Contact
action manipulate
element-rule
name updatecon2
type uri-user
action replace
comparison-type pattern-rule
match-value (.*)
new-value $ORIGINAL-^"+1"
header-rule
name fixRoute2
header-name Route
action manipulate
element-rule
name updateroute2
type uri-user
action replace
comparison-type pattern-rule
match-value (.*)
new-value $ORIGINAL-^"+1"
header-rule
name Privacy
header-name From
action sip-manip
new-value Check_privacy_header sip-manipulation
name outManipToSFB
header-rule
name From
header-name From
action manipulate
msg-type request
element-rule
name From_header
type uri-host
action replace
new-value $LOCAL_IP
element-rule
name modanonfrom
type uri-user
action replace
comparison-type pattern-rule
match-value \+1anonymous
new-value anonymous
header-rule
name To
header-name To
action manipulate
msg-type request
element-rule
name To
type uri-host
action replace
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new-value $REMOTE_IP
header-rule
name fixcontactuser
header-name Contact
action manipulate
comparison-type pattern-rule
element-rule
name user
type uri-user
action replace
comparison-type pattern-rule
match-value (.*)
new-value "+1"+$ORIGINAL sip-monitoring
match-any-filter enabled steering-pool
ip-address 192.168.1.220
start-port 20000
end-port 40000
realm-id ctl-trunk steering-pool
ip-address 192.168.1.221
start-port 20000
end-port 40000
realm-id ctl-trunk2 steering-pool
ip-address 192.168.3.220
start-port 20000
end-port 40000
realm-id Nice-SIPREC steering-pool
ip-address 192.168.4.135
start-port 20000
end-port 40000
realm-id core steering-pool
ip-address 192.168.4.136
start-port 20000
end-port 40000
realm-id core2 surrogate-agent
register-host voip.centurylink.com
register-user 4695737015
realm-id core2
customer-host 192.168.1.171
customer-next-hop 192.168.1.171
register-contact-host 192.168.1.171
register-contact-user 4695737015
password *************
register-expires 60
auth-user 257570-4695737015 surrogate-agent
register-host voip.centurylink.com
register-user 4695550016
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realm-id core
customer-host 192.168.1.170
customer-next-hop 192.168.1.170
register-contact-host 192.168.1.170
register-contact-user 4695550016
password *************
register-expires 60
auth-user 257570-4695550016 system-config
hostname CenturyLink-IOT
process-log-level DEBUG
comm-monitor
state enabled
monitor-collector
address 172.18.255.71
default-gateway 192.168.1.1 translation-rules
id addplus1
type add
add-string +1 translation-rules
id stripplus1
type delete
delete-string +1
SBC configuration for Dual Trunk registration using G729 towards trunk
For this configuration, the SBC will need to transcode between G729 and PCMU codecs on the turnk and SFB side respectively.
For transcoding, we configure the codec polices, to-trunk and to-sfb, mentioned below and apply them on the trunk and SFB realms respectively. The following configuration shows the configuration elements that need to be changed for the transcoding configuration to take effect codec-policy
name to-trunk
allow-codecs * PCMU:no G729::vadoff
add-codecs-on-egress G729
order-codecs
packetization-time 20
force-ptime disabled
dtmf-in-audio disabled
last-modified-by [email protected]
last-modified-date 2016-08-08 16:43:07 codec-policy
name to-sfb
allow-codecs * G729:no
add-codecs-on-egress PCMU
order-codecs
packetization-time 20
force-ptime disabled
dtmf-in-audio disabled
last-modified-by [email protected]
last-modified-date 2016-08-08 16:43:07 media-profile
name G729
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subname vadoff
payload-type 18
parameters annexb=no playback-config
name transferg729
entry
encoding G729
filename ringback-us-g729.raw realm-config
identifier core
network-interfaces s1p0:0
mm-in-realm enabled
qos-enable enabled
restricted-latching sdp
codec-policy core realm-config
identifier ctl-trunk
network-interfaces s0p0:0
mm-in-realm enabled
qos-enable enabled
media-policy voip-default
codec-policy realm-config
access
identifier core2
network-interfaces s1p0:0
mm-in-realm enabled
qos-enable enabled
restricted-latching sdp
codec-policy core realm-config
identifier ctl-trunk2
network-interfaces s0p0:0
mm-in-realm enabled
qos-enable enabled
media-policy voip-default
codec-policy access
On the SFB server, add the ip address of the sip-interface of the core2 realm as a second PSTN gateway in the topology builder and also in the Routes tab of the Voice Routing section of the SFB Control Panel following the steps mentioned in the section
–
Configuring the Skype for Business server.
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Test Plan
Following is the test plan executed against this setup and results have been documented below
Title Description Test Setup
Configure Dual Trunk on
PBX
Configure Dual Trunk on
ITSP
PBX is configured and connected to 2 PSTN
GW/SBCs
ITSP is configured and connected to 2 PSTN
GW/SBCs
Regitration of Dual Trunks Ensure that both trunks to
ITSP are registered successfully using the individual trunk registration information
The steps will be based on the type of PBX being utilized.
Ensure that trunks are configured between PBX and SBC.
Verify OPTIONS msgs from either PBX or SBC are being responded correctly by the other entity
The steps will be based on the type of SBC being utilized.
Ensure the TWO SBCs are configured with individual trunks to ITSP
1. Each SBC is configured with a trunk to ITSP and associated authentiation/digest and registration information.
2. Invoke a command on
SBC to register the trunk with ITSP.
3. Verify that 200 OK is received from ITSP for both the trunks.
Status
Passed
Passed
Passed
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Inbound PSTN calls pick correct trunk to SBC
Verify that PSTN to PBX inbound calls arrive on both the trunks when multiple calls are made
1. Dial an inbound call to the PBX.
2. Verify ringing is heard by calling and called parties.
3. Verify the trace shows a valid ringing indication message
4. Take called party phone off-hook.
5. Verify that a media path is established in both directions.
6. Hang up calling party
7. Verify the IP/PBX receives a Bye message.
8. Make a note of the
Trunk on which the call arrived to the SBC and
PBX.
9. Repeat the above steps 3 more times (total
4 calls).
10. Verify that calls to
PBX arrive on both the trunks.
11. Document Test
Results.
12. Save Trace.
Passed
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PBX calls are delivered to
PSTN on both the trunks
Calls from PBX to PSTN are delivered to
ITSP/PSTN utilizing both the configured trunks
Alien TN calls on 1st trunk Verify calls are successful with Alien TNs on 1st trunk
1. Dial an outbound call from the PBX.
2. Verify ringing is heard by calling and called parties.
3. Verify the trace shows a valid ringing indication message
4. Take called party phone off-hook.
5. Verify that a media path established in both directions.
6. Hang up Calling Party
7. Verify the IP/PBX sends a Bye message.
8. Make a note of the
Trunk on which the call was sent to ITSP.
9. Repeat the above steps 3 more times (total
4 calls).
10. Verify that calls from
PBX are sent out on both the trunks to ITSP.
11. Verify each call has
PAI sent per the trunk configuration
12. Document Test
Results.
13. Save Trace.
1. After Alien TN is set up on a Trunk1 in
CenturyLink Network
2. PBX User Places a Call to PSTN
3. PBX User receives ring back
4. PSTN receives ringing
5. PSTN receives caller id of the Alien TN
6. PSTN answers the call
7. 2 way audio is received
8. PBX Phone releases
Calls
9. PSTN receives a Bye
Passed
Passed
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Alien TN calls on 2nd trunk
Failover of 1st trunk WAN
- PSTN-PBX
Verify calls are successful with Alien TNs on 2nd trunk
Ensure that calls are delivered from PSTN to
PBX when the first trunk has failed on the WAN side
1. After Alien TN is set up on a Trunk2 in
CenturyLink Network
2. PBX User Places a Call to PSTN
3. PBX User receives ring back
4. PSTN receives ringing
5. PSTN receives caller id of the Alien TN
6. PSTN answers the call
7. 2 way audio is received
8. PBX Phone releases
Calls
9. PSTN receives a Bye
1. Down the WAN interface associated with
Trunk 1.
2. Make 3 calls from PSTN to PBX (one after the other)
3. Ensure that all 3 calls are delivered to PBX utilizing Trunk 2
4. PBX user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
Passed
Passed
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Failover of 1st trunk WAN
- PBX-PSTN
Ensure that calls are delivered from PBX to
PSTN when the first trunk has failed on the WAN side
Restore 1st trunk WAN:
PSTN-PBX
Ensure that calls are delivered from PSTN to
PBX when the first trunk has has been restored
1. Down the WAN interface associated with
Trunk 1.
2. Make 3 calls from PBX to PSTN (one after the other)
3. Ensure that all 3 calls are delivered to PSTN utilizing Trunk 2
4. PSTN user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
1. WAN interface associated with Trunk 1 is brought back into service.
2. Make 3 calls from PSTN to PBX (one after the other)
3. Ensure that at least one call is delivered to the PBX via Trunk 1
4. PBX user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
Passed
Passed
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Restore 1st trunk WAN:
PBX-PSTN
Failover of 2nd trunk
WAN: PSTN-PBX
Ensure that calls are delivered from PBX to
PSTN when the first trunk has has been restored
Ensure that calls are delivered from PSTN to
PBX when the second trunk has failed on the
WAN side
1. WAN interface associated with Trunk 1 is brought back into service.
2. Make 3 calls from PBX to PSTN (one after the other)
3. Ensure that at least one call is delivered to the
PSTN via Trunk 1
4. PSTN user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
1. Down the WAN interface associated with
Trunk 2.
2. Make 3 calls from PSTN to PBX (one after the other)
3. Ensure that all 3 calls are delivered to PBX utilizing Trunk 1
4. PBX user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
Passed
Passed
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Failover of 2nd trunk
WAN: PBX-PSTN
Restore 2nd trunk WAN:
PSTN-PBX
Ensure that calls are delivered from PBX-PSTN when the second trunk has failed on the WAN side
Ensure that calls are delivered from PSTN to
PBX when the second trunk has has been restored
1. Down the WAN interface associated with
Trunk 2.
2. Make 3 calls from PBX to PSTN (one after the other)
3. Ensure that all 3 calls are delivered to PBX utilizing Trunk 1
4. PSTN user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
1. WAN interface associated with Trunk 2 is brought back into service.
2. Make 3 calls from PSTN to PBX (one after the other)
3. Ensure that at least one call is delivered to the PBX via Trunk 2
4. PSTN user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
Passed
Passed
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Restore 2nd trunk WAN:
PBX-PSTN
Failover of 1st trunk LAN -
PBX to PSTN
Ensure that calls are delivered from PBX to
PSTN when the second trunk has has been restored
Ensure that calls are delivered from PBX to
PSTN when the first trunk has failed on the LAN side
1. WAN interface associated with Trunk 2 is brought back into service.
2. Make 3 calls from PBX to PSTN (one after the other)
3. Ensure that at least one call is delivered to the PBX via Trunk 2
4. PBX user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
1. Down the LAN interface associated with Trunk 1.
2. Make 3 calls from PBX to PSTN (one after the other)
3. Ensure that all 3 calls are delivered to PSTN utilizing Trunk 2
4. PSTN user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
Passed
Passed
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Failover of 1st trunk LAN -
PSTN to PBX
Ensure that calls are delivered from PSTN to
PBX when the first trunk has failed on the LAN side
Restore 1st trunk LAN -
PBX to PSTN
Ensure that calls are delivered from PBX to
PSTN when the first trunk has has been restored
1. Down the LAN interface associated with Trunk 1.
2. Make 3 calls from PSTN to PBX (one after the other)
3. Ensure that all 3 calls are delivered to PBX utilizing Trunk 2
4. PBX user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
1. LAN interface associated with Trunk 1 is brought back into service.
2. Make 3 calls from PBX to PSTN (one after the other)
3. Ensure that at least one call is delivered to the
PSTN via Trunk 2
4. PSTN user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
Passed
Passed
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Restore 1st trunk LAN -
PSTN to PBX
Failover of 2nd trunk LAN
- PBX to PSTN
Ensure that calls are delivered from PSTN to
PBX when the first trunk has has been restored
Ensure that calls are delivered from PBX to
PSTN when the second trunk has failed on the
LAN side
1. LAN interface associated with Trunk 1 is brought back into service.
2. Make 3 calls from PSTN to PBX (one after the other)
3. Ensure that at least one call is delivered to the PBX via Trunk 2
4. PBX user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
1. Down the LAN interface associated with Trunk 2.
2. Make 3 calls from PBX to PSTN (one after the other)
3. Ensure that all 3 calls are delivered to PSTN utilizing Trunk 1
4. PSTN user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
Passed
Passed
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Failover of 2nd trunk LAN
- PSTN to PBX
Ensure that calls are delivered from PSTN to
PBX when the second trunk has failed on the
LAN side
Restore 2nd trunk LAN -
PBX to PSTN
Ensure that calls are delivered from PBX to
PSTN when the second trunk has has been restored
1. Down the LAN interface associated with Trunk 2.
2. Make 3 calls from PSTN to PBX (one after the other)
3. Ensure that all 3 calls are delivered to PBX utilizing Trunk 1
4. PBX user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
1. LAN interface associated with Trunk 2 is brought back into service.
2. Make 3 calls from PBX to PSTN (one after the other)
3. Ensure that at least one call is delivered to the
PSTN via Trunk 2
4. PSTN user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
Passed
Passed
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Restore 2nd trunk LAN -
PSTN to PBX
Configure Dual Trunk on
PBX
Configure Dual Trunk on
ITSP
Ensure that calls are delivered from PSTN to
PBX when the second trunk has has been restored
PBX is configured and connected to 2 PSTN
GW/SBCs
ITSP is configured and connected to 2 PSTN
GW/SBCs
Regitration of Dual Trunks Ensure that both trunks to
ITSP are registered successfully using the individual trunk registration information
1. LAN interface associated with Trunk 2 is brought back into service.
2. Make 3 calls from PSTN to PBX (one after the other)
3. Ensure that at least one call is delivered to the PBX via Trunk 2
4. PBX user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
The steps will be based on the type of PBX being utilized.
Ensure that trunks are configured between PBX and SBC.
Verify OPTIONS msgs from either PBX or SBC are being responded correctly by the other entity
The steps will be based on the type of SBC being utilized.
Ensure the TWO SBCs are configured with individual trunks to ITSP
1. Each SBC is configured with a trunk to ITSP and associated authentiation/digest and registration information.
2. Invoke a command on
SBC to register the trunk with ITSP.
3. Verify that 200 OK is received from ITSP for both the trunks.
Passed
Passed
Passed
Passed
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Inbound PSTN calls pick correct trunk to SBC
Verify that PSTN to PBX inbound calls arrive on both the trunks when multiple calls are made
1. Dial an inbound call to the PBX.
2. Verify ringing is heard by calling and called parties.
3. Verify the trace shows a valid ringing indication message
4. Take called party phone off-hook.
5. Verify that a media path is established in both directions.
6. Hang up calling party
7. Verify the IP/PBX receives a Bye message.
8. Make a note of the
Trunk on which the call arrived to the SBC and
PBX.
9. Repeat the above steps 3 more times (total
4 calls).
10. Verify that calls to
PBX arrive on both the trunks.
11. Document Test
Results.
12. Save Trace.
Passed
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PBX calls are delivered to
PSTN on both the trunks
Calls from PBX to PSTN are delivered to
ITSP/PSTN utilizing both the configured trunks
Alien TN calls on 1st trunk Verify calls are successful with Alien TNs on 1st trunk
1. Dial an outbound call from the PBX.
2. Verify ringing is heard by calling and called parties.
3. Verify the trace shows a valid ringing indication message
4. Take called party phone off-hook.
5. Verify that a media path established in both directions.
6. Hang up Calling Party
7. Verify the IP/PBX sends a Bye message.
8. Make a note of the
Trunk on which the call was sent to ITSP.
9. Repeat the above steps 3 more times (total
4 calls).
10. Verify that calls from
PBX are sent out on both the trunks to ITSP.
11. Verify each call has
PAI sent per the trunk configuration
12. Document Test
Results.
13. Save Trace.
1. After Alien TN is set up on a Trunk1 in
CenturyLink Network
2. PBX User Places a Call to PSTN
3. PBX User receives ring back
4. PSTN receives ringing
5. PSTN receives caller id of the Alien TN
6. PSTN answers the call
7. 2 way audio is received
8. PBX Phone releases
Calls
9. PSTN receives a Bye
Passed
Passed
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Alien TN calls on 2nd trunk
Failover of 1st trunk WAN
- PSTN-PBX
Verify calls are successful with Alien TNs on 2nd trunk
Ensure that calls are delivered from PSTN to
PBX when the first trunk has failed on the WAN side
1. After Alien TN is set up on a Trunk2 in
CenturyLink Network
2. PBX User Places a Call to PSTN
3. PBX User receives ring back
4. PSTN receives ringing
5. PSTN receives caller id of the Alien TN
6. PSTN answers the call
7. 2 way audio is received
8. PBX Phone releases
Calls
9. PSTN receives a Bye
1. Down the WAN interface associated with
Trunk 1.
2. Make 3 calls from PSTN to PBX (one after the other)
3. Ensure that all 3 calls are delivered to PBX utilizing Trunk 2
4. PBX user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
Passed
Passed
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Failover of 1st trunk WAN
- PBX-PSTN
Ensure that calls are delivered from PBX to
PSTN when the first trunk has failed on the WAN side
Restore 1st trunk WAN:
PSTN-PBX
Ensure that calls are delivered from PSTN to
PBX when the first trunk has has been restored
1. Down the WAN interface associated with
Trunk 1.
2. Make 3 calls from PBX to PSTN (one after the other)
3. Ensure that all 3 calls are delivered to PSTN utilizing Trunk 2
4. PSTN user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
1. WAN interface associated with Trunk 1 is brought back into service.
2. Make 3 calls from PSTN to PBX (one after the other)
3. Ensure that at least one call is delivered to the PBX via Trunk 1
4. PBX user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
Passed
Passed
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Restore 1st trunk WAN:
PBX-PSTN
Failover of 2nd trunk
WAN: PSTN-PBX
Ensure that calls are delivered from PBX to
PSTN when the first trunk has has been restored
Ensure that calls are delivered from PSTN to
PBX when the second trunk has failed on the
WAN side
1. WAN interface associated with Trunk 1 is brought back into service.
2. Make 3 calls from PBX to PSTN (one after the other)
3. Ensure that at least one call is delivered to the
PSTN via Trunk 1
4. PSTN user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
1. Down the WAN interface associated with
Trunk 2.
2. Make 3 calls from PSTN to PBX (one after the other)
3. Ensure that all 3 calls are delivered to PBX utilizing Trunk 1
4. PBX user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
Passed
Passed
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Failover of 2nd trunk
WAN: PBX-PSTN
Restore 2nd trunk WAN:
PSTN-PBX
Ensure that calls are delivered from PBX-PSTN when the second trunk has failed on the WAN side
Ensure that calls are delivered from PSTN to
PBX when the second trunk has has been restored
1. Down the WAN interface associated with
Trunk 2.
2. Make 3 calls from PBX to PSTN (one after the other)
3. Ensure that all 3 calls are delivered to PBX utilizing Trunk 1
4. PSTN user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
1. WAN interface associated with Trunk 2 is brought back into service.
2. Make 3 calls from PSTN to PBX (one after the other)
3. Ensure that at least one call is delivered to the PBX via Trunk 2
4. PSTN user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
Passed
Passed
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Restore 2nd trunk WAN:
PBX-PSTN
Failover of 1st trunk LAN -
PBX to PSTN
Ensure that calls are delivered from PBX to
PSTN when the second trunk has has been restored
Ensure that calls are delivered from PBX to
PSTN when the first trunk has failed on the LAN side
1. WAN interface associated with Trunk 2 is brought back into service.
2. Make 3 calls from PBX to PSTN (one after the other)
3. Ensure that at least one call is delivered to the PBX via Trunk 2
4. PBX user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
1. Down the LAN interface associated with Trunk 1.
2. Make 3 calls from PBX to PSTN (one after the other)
3. Ensure that all 3 calls are delivered to PSTN utilizing Trunk 2
4. PSTN user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
Passed
Passed
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Failover of 1st trunk LAN -
PSTN to PBX
Ensure that calls are delivered from PSTN to
PBX when the first trunk has failed on the LAN side
Restore 1st trunk LAN -
PBX to PSTN
Ensure that calls are delivered from PBX to
PSTN when the first trunk has has been restored
1. Down the LAN interface associated with Trunk 1.
2. Make 3 calls from PSTN to PBX (one after the other)
3. Ensure that all 3 calls are delivered to PBX utilizing Trunk 2
4. PBX user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
1. LAN interface associated with Trunk 1 is brought back into service.
2. Make 3 calls from PBX to PSTN (one after the other)
3. Ensure that at least one call is delivered to the
PSTN via Trunk 2
4. PSTN user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
Passed
Passed
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Restore 1st trunk LAN -
PSTN to PBX
Failover of 2nd trunk LAN
- PBX to PSTN
Ensure that calls are delivered from PSTN to
PBX when the first trunk has has been restored
Ensure that calls are delivered from PBX to
PSTN when the second trunk has failed on the
LAN side
1. LAN interface associated with Trunk 1 is brought back into service.
2. Make 3 calls from PSTN to PBX (one after the other)
3. Ensure that at least one call is delivered to the PBX via Trunk 2
4. PBX user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
1. Down the LAN interface associated with Trunk 2.
2. Make 3 calls from PBX to PSTN (one after the other)
3. Ensure that all 3 calls are delivered to PSTN utilizing Trunk 1
4. PSTN user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
Passed
Passed
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Failover of 2nd trunk LAN
- PSTN to PBX
Ensure that calls are delivered from PSTN to
PBX when the second trunk has failed on the
LAN side
Restore 2nd trunk LAN -
PBX to PSTN
Ensure that calls are delivered from PBX to
PSTN when the second trunk has has been restored
1. Down the LAN interface associated with Trunk 2.
2. Make 3 calls from PSTN to PBX (one after the other)
3. Ensure that all 3 calls are delivered to PBX utilizing Trunk 1
4. PBX user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
1. LAN interface associated with Trunk 2 is brought back into service.
2. Make 3 calls from PBX to PSTN (one after the other)
3. Ensure that at least one call is delivered to the
PSTN via Trunk 2
4. PSTN user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
Passed
Passed
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.
Restore 2nd trunk LAN -
PSTN to PBX
Ensure that calls are delivered from PSTN to
PBX when the second trunk has has been restored
1. LAN interface associated with Trunk 2 is brought back into service.
2. Make 3 calls from PSTN to PBX (one after the other)
3. Ensure that at least one call is delivered to the PBX via Trunk 2
4. PBX user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
Passed
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Troubleshooting Tools
If you find that you are not able to complete calls or have problems with the test cases, there are a few tools available for Windows
Server, Lync Server, and the Oracle SBC like logging and tracing which may be of assistance. In this section we will provide a list of tools which you can use to aid in troubleshooting any issues you may encounter.
Since we are concerned with communication between the Lync Server mediation server and the SBC we will focus on the troubleshooting tools to use between those devices if calls are not working or tests are not passing.
Microsoft Network Monitor (NetMon)
NetMon is a network protocol analyzer which is freely downloadable from Microsoft. It can be found at www.microsoft.com/downloads . NetMon could be installed on the Lync Server mediation server, the Lync Server Standard Edition server, or Enterprise Edition front end server.
Wireshark
Wireshark is also a network protocol analyzer which is freely downloadable from www.wireshark.org
. Wireshark could be installed on the Lync Server mediation server, the Lync Server Standard Edition server, or MCS Enterprise Edition front end server.
Eventviewer
There are several locations in the event viewer where you can find valuable information to aid in troubleshooting issues with your deployment.
With the requirement that there is a completely functioning Lync Server with Enterprise Voice deployment in place, there are only a few areas in which one would use the Event Viewer for troubleshooting:
The Enterprise Voice client;
The Lync Server Front End server;
A Lync Server Standard Edition Server; and
A Lync Server Mediation Server.
On the Oracle SBC 4600 Series
The Oracle SBC provides a rich set of statistical counters available from the ACLI, as well as log file output with configurable detail.
The follow sections detail enabling, adjusting and accessing those interfaces.
Resetting the statistical counters, enabling logging and restarting the log files
.
At the SBC Console: oraclesbc1# reset sipd oraclesbc1# notify sipd debug oraclesbc1# enabled SIP Debugging oraclesbc1# notify all rotate-logs
Examining the log files
Note: You will FTP to the management interface of the SBC with the username user and user mode password (the default is
“acme”).
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C:\Documents and Settings\user>ftp 192.168.5.24
Connected to 192.168.85.55.
220 oraclesbc1FTP server (VxWorks 6.4) ready.
User (192.168.85.55:(none)): user
331 Password required for user.
Password: acme
230 User user logged in. ftp> cd /ramdrv/logs
250 CWD command successful. ftp> get sipmsg.log
200 PORT command successful.
150 Opening ASCII mode data connection for '/ramdrv/logs/sipmsg.log' (3353 bytes).
226 Transfer complete. ftp: 3447 bytes received in 0.00Seconds 3447000.00Kbytes/sec. ftp> get log.sipd
200 PORT command successful.
150 Opening ASCII mode data connection for '/ramdrv/logs/log.sipd' (204681 bytes).
226 Transfer complete. ftp: 206823 bytes received in 0.11Seconds 1897.46Kbytes/sec. ftp> bye
221 Goodbye.
You may now examine the log files with the text editor of your choice.
Through the Web GUI
You can also check the display results of filtered SIP session data from the Oracle Enterprise Session Border Controller, and provides traces in a common log format for local viewing or for exporting to your PC. Please check the “Monitor and Trace” section
(page 145) of the Web GUI User Guide available at http://docs.oracle.com/cd/E56581_01/index.htm
Telnet
Since we are working within an architecture which uses bound TCP listening ports for functionality, the simplest form of troubleshooting can be seeing if the devices are listening on a particular port, as well as confirming that the there is nothing blocking them such as firewalls. Ensure that you have a TELNET client available on a workstation as well as on the Lync Server mediation server.
The Lync Server mediation server will listen on TCP port 5067 by default for SIP signaling. In our example we are listening on 5060 on the PSTN facing NIC. From the Standard Edition pool or Enterprise Edition pool the Mediation Server would be listening on port
5061. Tests may include:
Client to pool server:
telnet <servername> 5061
Pool server to Mediation Server:
telnet <servername> 5061
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On the Lync Server
Lync Server Logging Tool
The Skype for Business Logging Tool provides internal traces and messaging between different Skype for Business elements like
Front-end, Mediation server, Lync Clients, etc. File name is OCSReskit.msi. Once installed, it can be accessed from any one of the
Lync Server servers by running Start/Microsoft Skype for Business/Lync Server Logging Tool.
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Appendix A
Accessing the ACLI
Access to the ACLI is provided by:
The serial console connection;
TELNET, which is enabled by default but may be disabled; and
SSH, this must be explicitly configured.
Initial connectivity will be through the serial console port. At a minimum, this is how to configure the management (eth0) interface on the SBC.
ACLI Basics
There are two password protected modes of operation within the ACLI, User mode and Superuser mode.
When you establish a connection to the SBC, the prompt for the User mode password appears. The default password is acme.
User mode consists of a restricted set of basic monitoring commands and is identified by the greater than sign (>) in the system prompt after the target name. You cannot perform configuration and maintenance from this mode.
The Superuser mode allows for access to all system commands for operation, maintenance, and administration. This mode is identified by the pound sign (#) in the prompt after the target name. To enter the Superuser mode, issue the enable command in the User mode.
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From the Superuser mode, you can perform monitoring and administrative tasks; however you cannot configure any elements. To return to User mode, issue the exit command.
You must enter the Configuration mode to configure elements. For example, you can access the configuration branches and configuration elements for signaling and media configurations. To enter the Configuration mode, issue the
configure terminal command in the Superuser mode.
Configuration mode is identified by the word configure in parenthesis followed by the pound sign (#) in the prompt after the target name, for example, oraclesbc1(configure)#. To return to the Superuser mode, issue the
exit command.
In the configuration mode, there are six configuration branches:
bootparam;
ntp-sync;
media-manager;
session-router;
system; and
security.
The ntp-sync and bootparams branches are flat branches (i.e., they do not have elements inside the branches). The rest of the branches have several elements under each of the branches.
The bootparam branch provides access to SBC boot parameters. Key boot parameters include:
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boot device
– The global management port, usually eth0
file name
– The boot path and the image file.
inet on ethernet
– The IP address and subnet mask (in hex) of the management port of the SD.
host inet
–The IP address of external server where image file resides.
user and ftp password
– Used to boot from the external FTP server.
gateway inet
– The gateway IP address for reaching the external server, if the server is located in a different network.
The ntp-sync branch provides access to ntp server configuration commands for synchronizing the SBC time and date.
The security branch provides access to security configuration.
The system branch provides access to basic configuration elements as system-config, snmp-community, redundancy, physical interfaces, network interfaces, etc.
The session-router branch provides access to signaling and routing related elements, including
H323-config, sip-config, iwf-config, local-policy, sip-manipulation, session-agent, etc.
The media-manager branch provides access to media-related elements, including realms, steering pools, dns-config, mediamanager, and so forth.
You will use media-manager, session-router, and system branches for most of your working configuration.
Configuration Elements
The configuration branches contain the configuration elements. Each configurable object is referred to as an element. Each element consists of a number of configurable parameters.
Some elements are single-instance elements, meaning that there is only one of that type of the element - for example, the global system configuration and redundancy configuration.
Some elements are multiple-instance elements. There may be one or more of the elements of any given type. For example, physical and network interfaces.
Some elements (both single and multiple instance) have sub-elements. For example:
SIP-ports - are children of the sip-interface element
peers
– are children of the redundancy element
destinations
– are children of the peer element
Creating an Element
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1. To create a single-instance element, you go to the appropriate level in the ACLI path and enter its parameters. There is no need to specify a unique identifier property because a single-instance element is a global element and there is only one instance of this element.
2. When creating a multiple-instance element, you must specify a unique identifier for each instance of the element.
3. It is important to check the parameters of the element you are configuring before committing the changes. You do this by issuing the
show command before issuing the done command. The parameters that you did not configure are filled with either default values or left empty.
4. On completion, you must issue the
done command. The done command causes the configuration to be echoed to the screen and commits the changes to the volatile memory. It is a good idea to review this output to ensure that your configurations are correct.
5. Issue the
exit command to exit the selected element.
Note that the configurations at this point are not permanently saved yet. If the SBC reboots, your configurations will be lost.
Editing an Element
The procedure of editing an element is similar to creating an element, except that you must select the element that you will edit before editing it.
1. Enter the element that you will edit at the correct level of the ACLI path.
2. Select the element that you will edit, and view it before editing it.
The
select command loads the element to the volatile memory for editing. The show command allows you to view the element to ensure that it is the right one that you want to edit.
3. Once you are sure that the element you selected is the right one for editing, edit the parameter one by one. The new value you provide will overwrite the old value.
4. It is important to check the properties of the element you are configuring before committing it to the volatile memory. You do this by issuing the
show command before issuing the done command.
5. On completion, you must issue the
done command.
6. Issue the
exit command to exit the selected element.
Note that the configurations at this point are not permanently saved yet. If the SBC reboots, your configurations will be lost.
Deleting an Element
The no command deletes an element from the configuration in editing.
To delete a single-instance element,
1. Enter the
no command from within the path for that specific element
2. Issue the
exit command.
To delete a multiple-instance element,
1. Enter the
no command from within the path for that particular element.
The key field prompt, such as <name>:<sub-port-id>, appears.
2. Use the <Enter> key to display a list of the existing configured elements.
3. Enter the number corresponding to the element you wish to delete.
4. Issue the
select command to view the list of elements to confirm that the element was removed.
Note that the configuration changes at this point are not permanently saved yet. If the SBC reboots, your configurations will be lost.
Configuration Versions
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At any time, three versions of the configuration can exist on the SBC: the edited configuration, the saved configuration, and the running configuration.
The edited configuration
– this is the version that you are making changes to. This version of the configuration is stored in the SBC
’s volatile memory and will be lost on a reboot.
To view the editing configuration, issue the
show configuration command.
The saved configuration
– on issuing the save-config command, the edited configuration is copied into the nonvolatile memory on the SBC and becomes the saved configuration. Because the saved configuration has not been activated yet, the changes in the configuration will not take effect. On reboot, the last activated configuration (i.e., the last running configuration) will be loaded, not the saved configuration.
The running configuration is the saved then activated configuration. On issuing the
activate-config command, the saved configuration is copied from the non-volatile memory to the volatile memory. The saved configuration is activated and becomes the running configuration. Although most of the configurations can take effect once being activated without reboot, some configurations require a reboot for the changes to take effect.
To view the running configuration, issue command show
running-config.
Saving the Configuration
The
save-config command stores the edited configuration persistently.
Because the saved configuration has not been activated yet, changes in configuration will not take effect. On reboot, the last activated configuration (i.e., the last running configuration) will be loaded. At this stage, the saved configuration is different from the running configuration.
Because the saved configuration is stored in non-volatile memory, it can be accessed and activated at later time.
Upon issuing the
save-config command, the SBC displays a reminder on screen stating that you must use the activate-
config command if you want the configurations to be updated. oraclesbc1 # save-config
Save-Config received, processing. waiting 1200 for request to finish
Request to 'SAVE-CONFIG' has Finished,
Save complete
Currently active and saved configurations do not match!
To sync & activate, run 'activate-config' or 'reboot activate'. oraclesbc1 #
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Activating the Configuration
On issuing the
activate-config command, the saved configuration is copied from the non-volatile memory to the volatile memory. The saved configuration is activated and becomes the running configuration.
Some configuration changes are service affecting when activated. For these configurations, the SBC warns that the change could have an impact on service with the configuration elements that will potentially be service affecting. You may decide whether or not to continue with applying these changes immediately or to apply them at a later time. oraclesbc1# activate-config
Activate-Config received, processing. waiting 120000 for request to finish
Request to 'ACTIVATE-CONFIG' has Finished,
Activate Complete oraclesbc1#
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