Application Notes for Avaya Communication Server 1000E 7.5 and

Avaya Solution & Interoperability Test Lab
Application Notes for Avaya Communication Server 1000E
7.5 and Acme Packet 3820 Net-Net® Session Director 6.3.0
with CenturyLink SIP Trunk Service (Legacy Qwest) –
Issue 1.0
Abstract
These Application Notes describe the steps to configure Session Initiation Protocol (SIP)
Trunking between CenturyLink SIP Trunk Service (Legacy Qwest) using Sonus NBS version
7.3.5R6 and an Avaya SIP-enabled enterprise solution. The Avaya solution consists of Avaya
Communication Server 1000E, Acme Packet 3820 Net-Net® Session Director and various
Avaya endpoints.
CenturyLink is a member of the Avaya DevConnect Service Provider program. Information in
these Application Notes has been obtained through DevConnect compliance testing and
additional technical discussions. Testing was conducted via the DevConnect Program at the
Avaya Solution and Interoperability Test Lab.
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Table of Contents
1.
2.
Introduction ............................................................................................................................. 4
General Test Approach and Test Results ................................................................................ 4
2.1. Interoperability Compliance Testing ................................................................................ 5
2.2. Test Results ...................................................................................................................... 5
2.3. Support ............................................................................................................................. 6
3. Reference Configuration ......................................................................................................... 6
4. Equipment and Software Validated ........................................................................................ 7
5. Configure Avaya Communication Server 1000E ................................................................... 7
5.1. Administer an IP Telephony Node ................................................................................... 9
5.1.1. Obtain Node IP Address ........................................................................................... 9
5.1.2. Terminal Proxy Server (TPS) ................................................................................. 10
5.1.3. Quality of Service (QoS) ........................................................................................ 11
5.1.4. Voice Gateway and Codecs .................................................................................... 12
5.1.5. SIP Gateway............................................................................................................ 13
5.1.6. Synchronize Node Configuration ........................................................................... 16
5.2. Virtual Superloops.......................................................................................................... 18
5.3. Media Gateway .............................................................................................................. 18
5.4. Virtual D-Channel, Routes and Trunks .......................................................................... 22
5.4.1. Virtual D-Channel Configuration ........................................................................... 22
5.4.2. Routes and Trunks Configuration ........................................................................... 24
5.5. Dialing and Numbering Plans ........................................................................................ 26
5.5.1. Route List Block ..................................................................................................... 26
5.5.2. NARS Access Code ................................................................................................ 28
5.5.3. Numbering Plan Area Codes .................................................................................. 29
5.5.4. Special Numbers ..................................................................................................... 31
5.5.5. Incoming Digit Translation ..................................................................................... 32
5.6. Zones and Bandwidth ..................................................................................................... 33
5.7. Example CS1000E Telephone Users ............................................................................. 35
5.7.1. Example SIP Phone DN 7108, Codec Considerations............................................ 35
5.7.2. Example Digital Phone DN 7107 with Call Waiting .............................................. 36
5.7.3. Example Analog Port with DN 7106, Fax .............................................................. 37
5.8. Save Configuration ......................................................................................................... 38
6. Configure Acme Packet 3820 Net-Net® Session Director ................................................... 38
6.1. Acme Packet Command Line Interface Summary ......................................................... 40
6.2. System Configuration ..................................................................................................... 41
6.3. Physical and Network Interfaces .................................................................................... 42
6.4. Realm ............................................................................................................................. 44
6.5. SIP Configuration ........................................................................................................... 46
6.6. SIP Interface ................................................................................................................... 47
6.7. Session Agent ................................................................................................................. 48
6.8. Session Agent Group ...................................................................................................... 51
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6.9. SIP Manipulation............................................................................................................ 52
6.10.
Steering Pools ............................................................................................................. 58
6.11.
Local Policy ................................................................................................................ 59
7. CenturyLink SIP Trunk Service Configuration .................................................................... 61
8. Verification ........................................................................................................................... 61
8.1. Avaya Communication Server 1000E Verifications ...................................................... 61
8.1.1. IP Network Maintenance and Reports Commands ................................................. 61
8.1.2. System Maintenance Commands ............................................................................ 63
9. Conclusion ............................................................................................................................ 64
10. Additional References ........................................................................................................ 65
Appendix A: Acme Packet 3820 Configuration File .................................................................... 65
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1. Introduction
These Application Notes describe a sample configuration of Avaya Communication Server
1000E release 7.5 and Acme Packet 3820 Net-Net Session Director 6.2.0 (Acme Packet 3820)
integration with CenturyLink SIP Trunk Service (Legacy Qwest) using Sonus NBS version
7.3.5R6. CenturyLink can offer SIP trunk service using several different platform technologies in
the CenturyLink network. These Application Notes correspond to the SIP trunk service offered
using a Sonus platform in the network.
In the sample configuration, the Acme Packet 3820 is used as an edge device between Avaya
Customer Premise Equipment (CPE) and CenturyLink SIP Trunk. The Acme Packet 3820
performs SIP header manipulation and provides Network Address Translation (NAT)
functionality to convert the private Avaya CPE IP addressing to IP addressing appropriate for the
CenturyLink SIP Trunk access method.
CenturyLink SIP Trunk Service is positioned for customers that have an IP-PBX or IP-based
network equipment with SIP functionality, but need a form of IP transport and local services to
complete their solution.
CenturyLink SIP Trunk Service will enable delivery of origination and termination of local,
long-distance and toll-free traffic across a single broadband connection. A SIP signaling
interface will be enabled to the Customer Premises Equipment (CPE). CenturyLink SIP Trunk
Service will also offer remote DID capability for a customer wishing to offer local numbers to
their customers that can be aggregated in SIP format back to customer.
DevConnect Compliance Testing is conducted jointly by Avaya and DevConnect members. The
jointly-defined test plan focuses on exercising APIs and/or standards-based interfaces pertinent
to the interoperability of the tested products and their functionalities. DevConnect Compliance
Testing is not intended to substitute full product performance or feature testing performed by
DevConnect members, nor is it to be construed as an endorsement by Avaya of the suitability or
completeness of a DevConnect member’s solution.
2. General Test Approach and Test Results
The general test approach was to configure a simulated enterprise site using Avaya
Communication Server 1000E (CS1000E) and Acme Packet 3820 to connect to the public
Internet using a broadband connection. The enterprise site was configured to connect to
CenturyLink SIP Trunk Service. This configuration (shown in Figure 1) was used to exercise the
features and functionality listed in Section 2.1.
DevConnect Compliance Testing is conducted jointly by Avaya and DevConnect members. The
jointly-defined test plan focuses on exercising APIs and/or standards-based interfaces pertinent
to the interoperability of the tested products and their functionalities. DevConnect Compliance
Testing is not intended to substitute full product performance or feature testing performed by
DevConnect members, nor is it to be construed as an endorsement by Avaya of the suitability or
completeness of a DevConnect member’s solution.
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2.1. Interoperability Compliance Testing
To verify SIP trunking interoperability, the following features and functionality were covered
during the interoperability compliance test:
Incoming PSTN calls to various phone types. Phone types included UNIStim, SIP,
digital, and analog telephones at the enterprise. All inbound PSTN calls were routed to
the enterprise across the SIP trunk from the service provider.
Outgoing PSTN calls from various phone types. Phone types included UNIStim, SIP,
digital, and analog telephones at the enterprise. All outbound PSTN calls were routed
from the enterprise across the SIP trunk to the service provider.
Inbound and outbound PSTN calls to/from Avaya one-X Communicator (soft client).
Various call types including: local, long distance, international, outbound toll-free,
operator assisted calls, emergency calls (911) and local directory assistance (411).
Inbound toll-free calls.
Codecs G.729A, G.729B and G.711MU.
DTMF transmission using RFC 2833.
T.38 Fax.
Caller ID presentation and Caller ID restriction.
Voicemail navigation for inbound and outbound calls.
User features such as hold and resume, transfer, and conference.
Off-net call forwarding and Mobile-X (extension to cellular).
Items not supported or not tested included the following:
SIP REFER method is not supported by Avaya CS1000E.
Mid-Call features using Mobile-X were not tested.
2.2. Test Results
Interoperability testing of CenturyLink SIP Trunk Service was completed with successful results
for all test cases with the exception of the observations/limitations described below.
Calling Party Number (PSTN transfers): The calling party number displayed on the
PSTN phone is not updated to reflect the true connected party on calls that are
transferred to the PSTN. After the call transfer is complete, the calling party number
displays the number of the transferring party and not the actual connected party. The
PSTN phone display is ultimately controlled by the PSTN provider, thus this behavior is
not necessarily indicative of a limitation of the combined Avaya/CenturyLink SIP Trunk
Service solution. It is listed here simply as an observation.
History-Info header: The CenturyLink SIP Trunk Service does not support SIP History
Info Headers. Instead, the CenturyLink SIP Trunk Service requires that SIP Diversion
Header be sent for redirected calls. The CS1000E includes History-Info header in
messaging sent to Acme Packet 3820. A header manipulation rule was created in the
Acme Packet 3820 to add a valid Diversion header for redirected calls. See Section 6.9
and Appendix A.
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CenturyLink SIP Trunk Service (Legacy Qwest) passed compliance testing.
2.3. Support
For technical support on the CenturyLink SIP Trunk Service, contact CenturyLink using the
Customer Care links at www.centurylink.com.
3. Reference Configuration
Figure 1 illustrates the sample configuration used for the DevConnect compliance testing. The
configuration is comprised of the Avaya CPE location connected via a T1 Internet connection to
the CenturyLink SIP Trunks to East and West servers. The Avaya CPE location simulates a
customer site. At the edge of the Avaya CPE location, an Acme Packet 3820 provides NAT
functionality and SIP header manipulation. The Acme Packet 3820 receives traffic from
CenturyLink SIP Trunk Service on port 5060 and sends traffic to the CenturyLink SIP Trunk
Service using destination port 5060, using the UDP protocol. For security reasons, any actual
public IP addresses used in the configuration have been replaced with private IP addresses.
Similarly, any references to real routable PSTN numbers have also been changed to numbers that
cannot be routed by the PSTN.
Figure 1: Avaya Interoperability Test Lab Configuration
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4. Equipment and Software Validated
The following equipment and software were used for the sample configuration provided:
Avaya IP Telephony Solution Components
Component
Release
Avaya Communication Server 1000E running
Call Server: 7.50 .17 GA (CoRes)
on CP+DC server as co-resident configuration
Service Pack: 7.50.17_20120110
SSG Server: 7.50.17 GA
SLG Server: 7.50.17 GA
Communication Server 1000E Media
CSP Version: MGCC CD02
Gateway
MSP Version: MGCM AB01
APP Version: MGCA BA15
FPGA Version: MGCF AA19
BOOT Version: MGCB BA15
DSP1 Version: DSP4 AB01
BCSP Version: MGCC CD01
Acme Packet Net-Net Session Director 3820
6.3.0 MR-1
Avaya 1165E (UNIStim)
0626C8A
Avaya 1140E (SIP)
04.03.09.00
Avaya one-X® Communicator (SIP)
CS6.1.1.02
Avaya M3904 (Digital)
n/a
Avaya 6210 Analog Telephone
n/a
CenturyLink (Legacy Qwest) SIP Trunking Solution Components
Component
Release
Sonus Network Border Switch (NBS)
07.03.05 R006
Table 1: Equipment and Software Tested
The specific configuration above was used for the compatibility testing.
5. Configure Avaya Communication Server 1000E
This section describes the Avaya Communication Server 1000E configuration, focusing on the
routing of calls to CenturyLink over a SIP trunk. In the sample configuration, Avaya
Communication Server 1000E Release 7.5 was deployed as a co-resident system with the SIP
Signaling Server, and Call Server applications all running on the same CP+DC server platform.
This section focuses on the SIP Trunking configuration. Although sample screens are illustrated
to document the overall configuration, it is assumed that the basic configuration of the Call
Server and SIP Signaling Server applications has been completed, and that the Avaya
Communication Server 1000E is configured to support analog, digital, UNIStim, and SIP
telephones. For references on how to administer these functions of Avaya Communication
Server 1000E, see Section 10.
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Configuration will be shown using the web based Avaya Unified Communications Management
GUI. The Avaya Unified Communications Management GUI may be launched directly via
https://<ipaddress> where the relevant <ipaddress> in the sample configuration is 10.80.140.102.
The following screen shows an abridged log in screen. Log in with appropriate credentials.
The Avaya Unified Communications Management Elements page will be used for configuration.
Click on the Element Name corresponding to CS1000 in the Element Type column. In the
abridged screen below, the user would click on the Element Name EM on cs1k-cpdc.
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5.1. Administer an IP Telephony Node
This section describes how to configure an IP Telephony Node on the Communication Server
1000E.
5.1.1. Obtain Node IP Address
Expand System  IP Network on the left panel and select Nodes: Servers, Media Cards.
The IP Telephony Nodes page is displayed as shown below. Click <Node id> in the Node ID
column to view details of the node. In the sample configuration, Node ID 1005 was used.
The Node Details screen is displayed with additional details as shown below. Under the Node
Details heading at the top of the screen, make a note of the TLAN Node IPV4 address. In the
sample screen below, the Node IPV4 address is 10.80.140.103. This IP address will be needed
when configuring Acme Packet 3820 with a Session Agent for the CS1000E in Section 6.7.
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The following screen shows the Associated Signaling Servers & Cards heading at the bottom
of the screen, simply to document the configuration.
5.1.2. Terminal Proxy Server (TPS)
On the Node Details screen, scroll down in the top window and select the Terminal Proxy
Server (TPS) link as show below.
Check the UNIStim Line Terminal Proxy Server check box and then click the Save button
(not shown).
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5.1.3. Quality of Service (QoS)
On the Node Details screen, scroll down in the top window and select the Quality of Service
(QoS) link as shown below.
Set the Control packets and Voice packets values to the desired Diffserv settings required on
the internal network. The default Diffserv values are shown below. Click on the Save button.
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5.1.4. Voice Gateway and Codecs
On the Node Details screen, scroll down in the top window and select the Voice Gateway
(VGW) and Codecs link as shown below.
The following screen shows the General parameters used in the sample configuration.
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Use the scroll bar on the right to find the area with heading Voice Codecs. Note that Codec
G.711 is enabled by default. The following screen shows the G.711 parameters used in the
sample configuration.
For the Codec G.729, ensure that the Enabled box is checked, and the Voice Activity Detection
(VAD) box is un-checked. In the sample configuration, the CS1000E was configured to include
G.729A and G.711 in SDP Offers, in that order. During compliance testing, the G.729B codec
was also tested by checking the Voice Activity Detection (VAD) box.
5.1.5. SIP Gateway
The SIP Gateway is the SIP trunk between the CS1000E and Acme Packet 3820. On the Node
Details screen, scroll down in the top window and select the Gateway (SIPGw) link as show
below.
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On the Node ID: <id> – Virtual Trunk Gateway Configuration Details page, enter the
following values and use default values for remaining fields.
Sip domain name:
Local SIP port:
Gateway endpoint name:
Application node ID:
Enter the appropriate SIP domain for the customer network.
In the sample configuration, avayalab.com was used in the
Avaya Solutions and Interoperability Test lab environment.
Enter 5060.
Enter a descriptive name.
Enter <Node id>. In the sample configuration, Node 1005
was used matching the node show in Section 5.1.1.
The values defined for the sample configuration are shown below.
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Scroll down to the SIP Gateway Settings  Proxy or Redirect Server: section.
Under Proxy Server Route 1, enter the following and use default values for remaining fields.
Primary TLAN IP address: Enter the IP address of the Acme Packet Inside media
interface. In the sample configuration 10.64.19.150 was
used.
Port:
Enter 5060
Transport protocol:
Select TCP
The values defined for the sample configuration are shown below.
Scroll down and repeat these steps for the Proxy Server Route 2.
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Scroll down to the SIP URI Map section. The values defined for the sample configuration are
shown below. The Avaya CS1000E will put the “string” entered in the SIP URI Map in the
“phone-context=<string>” parameter in SIP headers such as the To and From headers. If the
value is configured to blank, the CS1000E will omit the “phone-context=” in the SIP header
altogether.
Scroll to the bottom of the page and click Save (not shown) to save SIP Gateway configuration
settings. This will return the interface to the Node Details screen.
5.1.6. Synchronize Node Configuration
On the Node Details screen click Save as shown below.
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Select Transfer Now on the Node Saved page as show below.
Once the transfer is complete, the Synchronize Configurations Files (NODE ID <id>) page is
displayed. Place a check mark next to the appropriate Hostname and click Start Sync. The
screen will automatically refresh until the synchronization is finished.
The Synchronization Status field will update from Sync required (as shown above) to
Synchronized (as shown below). After synchronization completes, place a check mark next to
the appropriate Hostname and click Restart Applications.
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5.2. Virtual Superloops
Expand System  Core Equipments on the left panel and select Superloops. In the sample
configuration, Superloop 4 is for the Media Gateway and Superloop 252 is the virtual Superloop
used by the IP phones and SIP trunks.
5.3. Media Gateway
Expand System  IP Network on the left panel and select Media Gateways. Click the link in
the Type column for the appropriate Media Gateway to be modified as shown below.
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The IPMG 4 0 Media Gateway Survivable(MGS) Configuration window appears. The
Telephony LAN (TLAN) IP Address under the DSP Daughterboard heading will be the IP
Address in the SDP portion of SIP messages, for calls requiring a gateway resource. For
example, for a call from a digital telephone to the PSTN via CenturyLink SIP Trunk, the IP
Address in the SDP in the INVITE message will be 10.80.140.104 in the sample configuration.
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Scroll down to the area of the screen containing VGW and IP phone codec profile and expand
it. The fax T.38 settings used for compliance testing is shown below.
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The Codec G.711 is enabled by default. Ensure that the Select box is checked for Codec G729A
and the VAD (Voice Activity Detection) box is un-checked. The Voice payload size of 20 can
be used with CenturyLink SIP Trunk Service for both G.729A and G.711. Click Save (not
shown) at the bottom of the window. Then click OK in the dialog box (not shown) to save the
IPMG configuration. During compliance testing, the G.729B codec was also tested by checking
the Voice Activity Detection (VAD) box. Scroll down and click Save and then click OK on the
new dialog box that appears to save the configuration.
After the configuration is saved, the Media Gateways page is displayed. Select the appropriate
Media Gateway and click Reboot to load the new configuration.
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5.4. Virtual D-Channel, Routes and Trunks
Avaya Communication Server 1000E Call Server utilizes a virtual D-channel and associated
Route and Trunks to communicate with the Signaling Server.
5.4.1. Virtual D-Channel Configuration
Expand Routes and Trunks on the left panel and select D-Channels. In the sample
configuration, there is a virtual D-Channel 15 associated with the Signaling Server.
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Select Edit to verify the configuration, as shown below. Verify DCIP has been selected for D
Channel Card Type field and the Interface type for D-Channel is set to Meridian Meridian
1(SL1). Under the Basic Options section, verify 128 is selected for the Output request Buffers
value.
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5.4.2. Routes and Trunks Configuration
In addition to configuring a virtual D-channel, a Route and associated Trunks must be
configured. Expand Routes and Trunks on the left panel and expand the customer number. In
the example screen that follows, it can be observed that Route 15 has 32 trunks in the sample
configuration.
Select Edit to verify the configuration, as shown below. As can be observed in the Incoming
and outgoing trunk (ICOG) parameter, incoming and outgoing calls are allowed. The Access
code for the trunk route (ACOD) will in general not be dialed, but the number that appears in
this field may be observed on Avaya CS1000E display phones if an incoming call on the trunk is
anonymous or marked for privacy.
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Further down in the Basic Configuration section verify the Node ID of signaling server of this
route (NODE) matches the node shown in Section 5.1. Also verify SIP (SIP) has been selected
for Protocol ID for the route (PCID) field. The Zone for codec selection and bandwidth
management (ZONE) parameter can be used to associate the route with a zone for configuration
of the audio codec preferences sent via the Session Description Protocol (SDP) in SIP
messaging. The D channel number (DCH) field must match the D-Channel number shown in
Section 5.4.1.
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Scroll down and expand the Basic Route Options section. Check the North American toll
scheme (NATL) and Incoming DID digit conversion on this route (IDC), input DCNO 0 for
both Day IDC Tree Number and Night IDC Tree Number as shown below. The DCNO is
created later on in Section 5.5.5.
5.5. Dialing and Numbering Plans
This section provides the configuration of the routing used in the sample configuration for
routing calls over the SIP Trunk between Avaya Communication Server 1000E and Acme Packet
3820 for calls destined for the CenturyLink SIP Trunk. The routing defined in this section is
simply an example and not intended to be prescriptive. Other routing policies may be appropriate
for different customer networks.
5.5.1. Route List Block
Expand Dialing and Numbering Plans on the left panel and select Electronic Switched
Network. Select Route List Block (RLB) on the Electronic Switched Network (ESN) page as
shown on the following page.
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The Route List Blocks screen is displayed. Enter an available route list index number in the
Please enter a route list index field and click to Add, or edit an existing entry by clicking the
corresponding Edit button. In the sample configuration, route list block index 15 is used. If
adding the route list index anew, scroll down to the Options area of the screen. If editing an
existing route list block index, select the Edit button next to the appropriate Data Entry Index as
shown below, and scroll down to the Options area of the screen.
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Under the Options section, select <Route id> in the Route Number field. In the sample
configuration route number 15 was used. Default values may be retained for remaining fields.
5.5.2. NARS Access Code
Expand Dialing and Numbering Plans on the left panel and select Electronic Switched
Network. Select ESN Access Codes and Parameters (ESN). Although not repeated below, this
link can be observed in the first screen in Section 5.5.1. In the NARS/BARS Access Code 1
field, enter the number the user will dial before the target PSTN number. In the sample
configuration, the single digit 9 was used.
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5.5.3. Numbering Plan Area Codes
Expand Dialing and Numbering Plans on the left panel and select Electronic Switched
Network. Scroll down and select Numbering Plan Area Code (NPA) under the appropriate
access code heading. In the sample configuration, this is Access Code 1, as shown below.
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Add a new NPA by entering it in the Please enter an area code box and click to Add or click
Edit to view or change an NPA that has been previously configured. In the screen below, it can
be observed that various dial strings such as 1303 and 1800 are configured.
In the screen below, the entry for 1303 is displayed. In the Route List Index, 15 is selected to use
the route list associated with the SIP Trunk to Acme Packet 3820 as shown in Section 5.4.2.
Default parameters may be retained for other parameters. Repeat this procedure for the dial
strings associated with other numbering plan area codes that should route to the SIP Trunk to
Acme Packet 3820.
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5.5.4. Special Numbers
In the testing associated with these Application Notes, special service numbers such as x11,
international calls, and operator assisted calls were also routed to Acme Packet 3820 and
ultimately to the CenturyLink SIP Trunk. Although not intended to be prescriptive, one approach
to such routing is summarized in this section.
Expand Dialing and Numbering Plans on the left panel and select Electronic Switched
Network. Scroll down and select Special Number (SPN) under the appropriate access code
heading (as can be observed in the first screen in Section 5.5.3).
Add a new number by entering it in the Please enter a Special Number box and click to Add or
click Edit to view or change a special number that has been previously configured. In the screen
below, it can be observed that various dial strings such as 0, 011, 411 and 911 calls are listed.
Route list index 15 has been selected in the same manner as shown for the NPAs in the prior
section.
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5.5.5. Incoming Digit Translation
In general, the incoming digit translation can be used to manipulate the digits received for an
incoming call. Expand Dialing and Numbering Plans on the left panel and select Incoming
Digit Translation. Click on the Edit IDC button as shown on the following screen.
Click on the New DCNO to create the digit translation mechanism or if editing an existing one,
select the Edit DCNO button next to the appropriate Digit Conversion Tree Number. In this
example, Digit Conversion Tree Number (DCNO) 0 has been created as shown below.
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Detail configuration of the DCNO is shown below. The Incoming Digits can be added to map to
the Converted Digits which would be the CS1000E system phones DN. This DCNO has been
assigned to route 15 as shown in Section 5.4.2.
In the following configuration, the incoming call from PSTN with the prefix 303-555-71xx will
be translated to CS1000E DN 71xx. The PSTN with the prefix 614-555-01xx will be translated
to CS1000E DN 51xx. The DID 303-555-7799 is translated to 5000 for Voicemail accessing
purpose.
5.6. Zones and Bandwidth
Zone configuration can be used to control codec selection and for bandwidth management. To
configure, expand System  IP Network on the left panel and select Zones as shown below.
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Select Bandwidth Zones. In the sample lab configuration, two zones are configured. In
production environments, it is likely that more zones will be required. Select the zone associated
with the virtual trunk to Acme Packet 3820 and click Edit as shown below. In the sample
configuration, this is Zone number 99.
In the resultant screen shown below, select Zone Basic Property and Bandwidth
Management.
The following screen shows the Zone 99 configuration. Note that Best Bandwidth (BB) is
selected for the zone strategy parameters so that codec G.729A is preferred over codec G.711MU
for calls with CenturyLink SIP Trunk.
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5.7. Example CS1000E Telephone Users
This section is not intended to be prescriptive, but simply illustrates a sampling of the telephone
users in the sample configuration.
5.7.1. Example SIP Phone DN 7108, Codec Considerations
The following screen shows basic information for a SIP phone in the configuration. The
telephone is configured as Directory Number 7108. Note that the telephone is in Zone 1 and is
associated with Node 1005 (see Section 5.1). A call between this telephone and another
telephone in Zone 1 will use a best quality strategy (see Section 5.6) and therefore can use
G.711MU. If this same telephone calls out to the PSTN via the CenturyLink SIP Trunk, the call
would use a best bandwidth strategy, and the call would use G.729A.
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5.7.2. Example Digital Phone DN 7107 with Call Waiting
The following screen shows basic information for a digital phone in the configuration. The
telephone is configured as Directory Number 7107.
The following screen shows basic key information for the telephone. It can be observed that the
telephone can support call waiting with tone. Although not shown in detail below, to use call
waiting with tone, assign a key CWT – Call Waiting, set the feature SWA – Call waiting from
a Station to Allowed, and set the feature WTA – Warning Tone to Allowed.
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5.7.3. Example Analog Port with DN 7106, Fax
The following screen shows basic information for an analog port in the configuration that may be
used with a telephone or fax machine. The port is configured as Directory Number 7106.
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5.8. Save Configuration
Expand Tools  Backup and Restore on the left panel and select Call Server. Select Backup
(not shown) and click Submit to save configuration changes as shown below.
6. Configure Acme Packet 3820 Net-Net® Session Director
This section describes the configuration of the Acme Packet 3820 necessary for interoperability
with CenturyLink and CS1000E. The Acme Packet 3820 is configured via the Acme Packet
Command Line Interface (ACLI). This section assumes the reader is familiar with accessing and
configuring the Acme Packet 3820.
A pictorial view of this configuration is shown below. It shows the internal components needed
for the compliance test. Each of these components is defined in the Acme Packet 3820
configuration file contained in Appendix A. However, this section does not cover standard
Acme Packet 3820 configurations that are not directly related to the interoperability test. The
details of these configuration elements can be found in Appendix A.
This section will not attempt to describe each component in its entirety but instead will highlight
critical fields in each component which relates to the functionality in these Application Notes
and the direct connection to CenturyLink and CS1000E. These same fields are highlighted in
Appendix A. The remaining fields are generally the default/standard value used by the Acme
Packet 3820 for that field. For additional details on the administration of the Acme Packet 3820,
see Reference [7].
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6.1. Acme Packet Command Line Interface Summary
The Acme Packet 3820 is configured using the Acme Packet Command Line Interface (ACLI).
The following are the generic ACLI steps for configuring various elements.
1. Access the console port of the Acme Packet 3820 using a PC and a terminal emulation
program such as HyperTerminal (use the RJ-45 to DB9 adapter as packaged with the
3820 for cable connection). Use the following settings for the serial port on the PC.
Bits per second: 115200
Data bits: 8
Parity : None
Stop bits: 1
Flow control: None
2. Log in to the Acme Packet 3820 with the user password.
3. Enable the Superuser mode by entering the enable command and then the superuser
password. The command prompt will change to include a “#” instead of a “>” while in
Superuser mode. This level of system access (i.e. at the “acmesystem#” prompt) will be
referred to as the main level of the ACLI. Specific sub-levels of the ACLI will then be
accessed to configure specific elements and specific parameters of those elements.
4. In Superuser mode, enter the configure terminal command. The configure terminal
command is used to access the system level where all operating and system elements may
be configured. This level of system access will be referred to as the configuration level.
5. Enter the name of an element to be configured (e.g., system).
6. Enter the name of a sub-element, if any (e.g., phy-interface).
7. Enter the name of an element parameter followed by its value (e.g., name M00).
8. Enter done to save changes to the element. Use of the done command causes the system
to save and display the settings for the current element.
9. Enter exit as many times as necessary to return to the configuration level.
10. Repeat Steps 5 - 9 to configure all the elements.
11. Enter exit to return to the main level.
12. Type save-config to save the entire configuration.
13. Type activate-config to activate the entire configuration.
After accessing different levels of the ACLI to configure elements and parameters, it is necessary
to return to the main level in order to run certain tasks such as saving the configuration,
activating the configuration, and rebooting the system.
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6.2. System Configuration
The system configuration defines system-wide parameters for the Acme Packet 3820.
The key system configuration (system-config) field is:
default-gateway: The IP address of the default gateway for the management network
(10.80.150.0/24) from Figure 1. In this case, the default gateway is 10.80.150.1.
system-config
hostname
description
location
mib-system-contact
mib-system-name
< text removed for brevity >
call-trace
internal-trace
log-filter
default-gateway
restart
exceptions
telnet-timeout
console-timeout
remote-control
cli-audit-trail
link-redundancy-state
source-routing
cli-more
terminal-height
debug-timeout
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disabled
disabled
all
10.80.150.1
enabled
0
0
enabled
enabled
disabled
disabled
disabled
24
0
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6.3. Physical and Network Interfaces
As part of the compliance test, the Ethernet interface slot 0 / port 0 of the Acme Packet 3820 was
connected to the external untrusted network. Ethernet slot 1 / port 0 was connected to the internal
corporate LAN. A network interface was defined for each physical interface to assign it a
routable IP address.
The key physical interface (phy-interface) fields are:
name: A descriptive string used to reference the Ethernet interface.
operation-type: Media indicates both signaling and media packets are sent on this
interface.
slot / port: The identifier of the specific Ethernet interface used.
phy-interface
name
operation-type
port
slot
virtual-mac
admin-state
auto-negotiation
duplex-mode
speed
overload-protection
last-modified-by
last-modified-date
phy-interface
name
operation-type
port
slot
virtual-mac
admin-state
auto-negotiation
duplex-mode
speed
overload-protection
last-modified-by
last-modified-date
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M00
Media
0
0
enabled
enabled
FULL
100
disabled
admin@console
2011-11-01 09:59:56
M10
Media
0
1
enabled
enabled
FULL
100
disabled
admin@console
2011-11-01 10:00:38
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The key network interface (network-interface) fields are:
name: The name of the physical interface (defined previously) that is associated with this
network interface.
description: A descriptive name to help identify the interface.
ip-address: The IP address on the interface connected to the network on which the
CenturyLink SIP Trunk Service resides. In the compliance test, the IP address 10.2.2.92
was assigned to the public interface and 10.64.19.150 was assigned to the private
interface.
netmask: Subnet mask for the IP subnet.
gateway: The subnet gateway address.
hip-ip-list: The list of virtual IP addresses assigned to the Acme Packet 3820 on this
interface. If a single virtual IP address is used, this value would be the same as the value
entered for the ip-address field above.
icmp-address: The list of IP addresses to which the Acme Packet 3820 will answer
ICMP requests on this interface.
network-interface
name
sub-port-id
description
hostname
ip-address
pri-utility-addr
sec-utility-addr
netmask
gateway
sec-gateway
gw-heartbeat
state
heartbeat
retry-count
retry-timeout
health-score
dns-ip-primary
dns-ip-backup1
dns-ip-backup2
dns-domain
dns-timeout
hip-ip-list
ftp-address
icmp-address
snmp-address
telnet-address
ssh-address
last-modified-by
last-modified-date
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M00
0
PUBLIC
10.2.2.92
255.255.255.128
10.2.2.1
disabled
0
0
1
0
11
10.2.2.92
admin@10.80.150.38
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The settings for the private side network interface are shown below.
network-interface
name
sub-port-id
description
hostname
ip-address
pri-utility-addr
sec-utility-addr
netmask
gateway
sec-gateway
gw-heartbeat
state
heartbeat
retry-count
retry-timeout
health-score
dns-ip-primary
dns-ip-backup1
dns-ip-backup2
dns-domain
dns-timeout
hip-ip-list
ftp-address
icmp-address
snmp-address
telnet-address
ssh-address
last-modified-by
last-modified-date
M10
0
PRIVATE
10.64.19.150
255.255.255.0
10.64.19.1
disabled
0
0
1
0
11
10.64.19.150
10.64.19.150
admin@10.80.150.38
2011-11-01 12:16:22
6.4. Realm
A realm represents a group of related Acme Packet 3820 components. Two realms were defined
for the compliance test. The peer realm was defined for the external network and the core realm
was defined for the internal network.
The key realm (realm-config) fields are:
identifier: A string used as a realm reference. This will be used in the configuration of
other components.
network interfaces: The network interfaces located in this realm.
In-manipulationid: For the core realm CS1K_To_CL was used. This name refers to a
set of sip-manipulations (defined in Section 6.9) that is performed on inbound traffic to
the Acme Packet 3820.
out-manipulationid: For the peer realm NatIP was used and for the core realm
AddDomain was used. These names refer to a set of sip-manipulations (defined in
Section 6.9) that are performed on outbound traffic from the Acme Packet 3820. These
sip-manipulations are specified in each realm. Thus, these sip-manipulations are applied
to outbound traffic from the public side (peer) of the Acme Packet 3820 as well as to
outbound traffic from the private side (core) of the Acme Packet 3820.
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realm-config
identifier
description
addr-prefix
network-interfaces
mm-in-realm
mm-in-network
mm-same-ip
mm-in-system
peer
0.0.0.0
M00:0
enabled
enabled
enabled
enabled
< text removed for brevity >
out-translationid
in-manipulationid
out-manipulationid
manipulation-string
manipulation-pattern
class-profile
average-rate-limit
NatIP
0
< text removed for brevity >
realm-config
identifier
description
addr-prefix
network-interfaces
mm-in-realm
mm-in-network
mm-same-ip
mm-in-system
core
0.0.0.0
M10:0
enabled
enabled
enabled
enabled
< text removed for brevity >
out-translationid
in-manipulationid
out-manipulationid
manipulation-string
manipulation-pattern
class-profile
average-rate-limit
CS1K_To_CL
AddDomain
0
< text removed for brevity >
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6.5. SIP Configuration
The SIP configuration (sip-config) defines the global system-wide SIP parameters, including SIP
timers, SIP options, which realm to send requests to if not specified elsewhere, and enabling the
SD to collect statistics on requests other than REGISTERs and INVITEs.
The key SIP configuration (sip-config) fields are:
state: enabled
home-realm-id: The name of the realm on the private side of the Acme Packet 3820.
egress-realm-id: The name of the realm on the private side of the Acme Packet 3820.
options: max-udp=length=0 This option was used to prevent errors about the packet
size being too large.
sip-config
state
operation-mode
dialog-transparency
home-realm-id
egress-realm-id
nat-mode
registrar-domain
registrar-host
registrar-port
register-service-route
init-timer
max-timer
trans-expire
invite-expire
enabled
dialog
enabled
core
core
None
0
always
500
4000
32
180
< text removed for brevity >
options
refer-src-routing
add-ucid-header
proxy-sub-events
max-udp-length=0
disabled
disabled
< text removed for brevity >
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6.6. SIP Interface
The SIP interface (sip-interface) defines the receiving characteristics of the SIP interfaces on the
Acme Packet 3820. Two SIP interfaces were defined; one for each realm.
The key SIP interface (sip-interface) fields are:
realm-id: The name of the realm to which this interface is assigned.
sip-port
o address: The IP address assigned to this sip-interface.
o port: The port assigned to this sip-interface. Port 5060 is used for both UDP and
TCP.
o transport-protocol: The transport method used for this interface.
o allow-anonymous: Defines from whom SIP requests will be allowed. On the peer
side, the value of agents-only is used. Thus, SIP requests will only be accepted
from session agents (as defined in Section 6.7) on this interface. On the core side,
the value of all is used. Thus, SIP requests will be accepted from anyone on this
interface.
sip-interface
state
realm-id
description
sip-port
address
port
transport-protocol
tls-profile
allow-anonymous
ims-aka-profile
carriers
trans-expire
invite-expire
enabled
peer
10.2.2.92
5060
UDP
agents-only
0
0
< text removed for brevity >
sip-interface
state
realm-id
description
sip-port
address
port
transport-protocol
tls-profile
allow-anonymous
ims-aka-profile
carriers
trans-expire
invite-expire
enabled
core
10.64.19.150
5060
TCP
all
0
0
< text removed for brevity >
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6.7. Session Agent
A session agent defines the characteristics of a signaling peer to the Acme Packet 3820 such as
CS1000E and CenturyLink SIP Trunk Service.
The key session agent (session-agent) fields are:
hostname: Fully qualified domain name or IP address of this SIP peer.
ip-address: The IP address of this SIP peer.
port: The port used by the peer for SIP traffic.
app-protocol: SIP
transport-method: UDP
realm-id: The realm id where this peer resides.
description: A descriptive name for the peer.
ping-method: OPTIONS;hops=70 This setting defines that the SIP OPTIONS message
will be sent to the peer to verify that the SIP connection is functional. In addition, this
parameter causes the Acme Packet 3820 to set the SIP “Max-Forward” field to 70 in
outbound SIP OPTIONS pings generated by the Acme Packet 3820 to this session agent.
ping-interval: Specifies the interval (in seconds) between each ping attempt.
The settings for the session agent used for CenturyLink East Inbound/Outbound peer:
session-agent
hostname
ip-address
port
state
app-protocol
app-type
transport-method
realm-id
egress-realm-id
description
carriers
allow-next-hop-lp
constraints
max-sessions
172.16.3.8
172.16.3.8
5060
enabled
SIP
UDP
peer
enabled
disabled
0
< text removed for brevity >
response-map
ping-method
ping-interval
OPTIONS;hops=70
60
< text removed for brevity >
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The settings for the session agent used for CenturyLink East Remote DID peer:
session-agent
hostname
ip-address
port
state
app-protocol
app-type
transport-method
realm-id
egress-realm-id
description
carriers
allow-next-hop-lp
constraints
max-sessions
172.16.3.9
172.16.3.9
5060
enabled
SIP
UDP
peer
enabled
disabled
0
< text removed for brevity >
response-map
ping-method
ping-interval
OPTIONS;hops=70
60
< text removed for brevity >
The settings for the session agent used for CenturyLink West Inbound/Outbound peer:
session-agent
hostname
ip-address
port
state
app-protocol
app-type
transport-method
realm-id
egress-realm-id
description
carriers
allow-next-hop-lp
constraints
max-sessions
172.16.2.8
172.16.2.8
5060
enabled
SIP
UDP
peer
enabled
disabled
0
< text removed for brevity >
response-map
ping-method
ping-interval
OPTIONS;hops=70
60
< text removed for brevity >
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The settings for the session agent used for CenturyLink West Remote DID peer:
session-agent
hostname
ip-address
port
state
app-protocol
app-type
transport-method
realm-id
egress-realm-id
description
carriers
allow-next-hop-lp
constraints
max-sessions
172.16.2.9
172.16.2.9
5060
enabled
SIP
UDP
peer
enabled
disabled
0
< text removed for brevity >
response-map
ping-method
ping-interval
OPTIONS;hops=70
60
< text removed for brevity >
The settings for the session agent used for CS1000E:
session-agent
hostname
ip-address
port
state
app-protocol
app-type
transport-method
realm-id
egress-realm-id
description
carriers
allow-next-hop-lp
constraints
max-sessions
10.80.140.103
10.80.140.103
5060
enabled
SIP
TCP
core
enabled
disabled
0
< text removed for brevity >
response-map
ping-method
ping-interval
OPTIONS;hops=70
60
< text removed for brevity >
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6.8. Session Agent Group
Session agents can be configured in a session agent group (SAG), so multiple session agents can
be assigned to a route policy for fail-over or load balancing purposes. For compliance testing
CenturyLink had four session agents assigned. Two of them were used for remote DIDs and
were allocated for inbound only, while the other two were used for both inbound and outbound
traffic. Only the two session agents allocated for outbound traffic were added to the SAG.
The key session agent group (session-group) fields are:
group-name: A descriptive string used to reference the session agent group.
state: enabled
app-protocol: SIP
strategy: Hunt This strategy will route to the secondary session agent only if the
primary fails. An alternative is to use a strategy of RoundRobin. This strategy will
alternatively select between session agents.
dest: The list of session agents to be added to the group by hostname. For compliance
testing 172.16.3.8 and 172.16.2.8 were used.
sag-recursion: enabled This allows Acme Packet 3820 to select a different session
agent in the SAG if a failure occurs to the first session agent.
session-group
group-name
description
state
app-protocol
strategy
dest
CL-OUT
enabled
SIP
Hunt
172.16.3.8
172.16.2.8
trunk-group
sag-recursion
stop-sag-recurse
last-modified-by
last-modified-date
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enabled
401,407
admin@10.80.150.38
2012-06-18 10:27:19
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6.9. SIP Manipulation
SIP manipulations are rules used to modify the SIP messages (if necessary) for interoperability.
In Section 6.4, it was defined that the set of sip-manipulations named NatIP would be performed
on outbound traffic in the peer realm and AddDomain would be performed on outbound traffic
in the core realm. The sip-manipulation named CS1K_To_CL would be performed on inbound
traffic from the core realm. For the complete configuration of these rules refer to Appendix A.
The key SIP manipulation (sip-manipulation) fields are:
name: The name of this set of SIP header rules.
header-rule
o name: The name of this individual header rule.
o header-name: The SIP header to be modified.
o action: The action to be performed on the header.
o comparison-type: The type of comparison performed when determining a match.
o msg-type: The type of message to which this rule applies.
o element-rule
 name: The name of this individual element rule.
 type: Defines the particular element in the header to be modified.
 action: The action to be performed on the element.
 match-val-type: Element matching criteria on the data type (if any) in
order to perform the defined action.
 comparison-type: The type of comparison performed when determining a
match.
 match-value: Element matching criteria on the data value (if any) in order
to perform the defined action.
 new-value: New value for the element (if any).
In the configuration file in Appendix A, the NatIP sip manipulation has many modifications (or
header-rules) defined. These header manipulations were added to hide the private IP address and
enterprise domain name which appear in the “To”, “From”, “Request-URI”, “Diversion” and
“PAI” SIP headers for outbound calls. As well as remove unwanted headers going to the SIP
service provider.
Similarly the AddDomain sip manipulation was used towards CS1000E to hide the public IP
addresses and to add the enterprise domain to the “From” and “PAI” SIP headers.
The CS1K_To_CL sip manipulation was used to add a “Diversion” header from the “HistoryInfo” header for redirected calls from CS1000E. This was added to the inbound traffic to the
Acme Packet 3820 so that it could be further modified by the NatIP sip manipulation to remove
the “History-Info” header and to hide the enterprise domain name.
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The example below shows the natFROM header-rule in the NatIP sip manipulation. It
specifies that the “From” header in SIP request messages will be manipulated based on the
element rule defined. The element rule natHost will match any value in the host part of the URI
and replace it with the value of $LOCAL_IP. The value of $LOCAL_IP is the outside IP
address of the Acme Packet 3820. See Appendix A for the complete NatIP sip manipulation
used during compliance testing.
sip-manipulation
name
description
split-headers
join-headers
header-rule
name
header-name
action
comparison-type
msg-type
methods
match-value
new-value
element-rule
name
parameter-name
type
action
match-val-type
comparison-type
match-value
new-value
NatIP
natFROM
From
manipulate
case-sensitive
request
natHost
uri-host
replace
any
case-sensitive
$LOCAL_IP
< text removed for brevity >
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The NatIP sip manipulation was also used to remove the “History-Info” and “x-nt-e164-clid”
headers and unwanted MIMEs in the body of the message.
header-rule
name
removeHist
header-name
History-Info
action
delete
comparison-type
case-sensitive
msg-type
any
methods
match-value
new-value
header-rule
name
removeXNTe164
header-name
X-nt-e164-clid
action
delete
comparison-type
case-sensitive
msg-type
any
methods
match-value
new-value
header-rule
name
removeMultiMIME
header-name
Content-Type
action
manipulate
comparison-type
case-sensitive
msg-type
any
methods
match-value
new-value
element-rule
name
nt_mcdn
parameter-name
application/x-ntmcdn-frag-hex;version-ssLinux-7.50.17;base=x2611
type
mime
action
delete-element
match-val-type
any
comparison-type
case-sensitive
match-value
new-value
element-rule
name
nt_esn5
parameter-name
application/x-ntesn5-frag-hex
type
mime
action
delete-element
match-val-type
any
comparison-type
case-sensitive
match-value
new-value
element-rule
name
nt_epid
parameter-name
application/x-ntepid-frag-hex;version=ssLinux-7.50.17;base=x2611
type
mime
action
delete-element
match-val-type
any
comparison-type
case-sensitive
match-value
new-value
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The example below shows the FromDomain header-rule in the AddDomain sip manipulation.
It specifies that the “From” header in SIP request messages will be manipulated based on the
element rule defined. The element rule From will match any value in the host part of the URI
and replace it with the value of avayalab.com. The value of avayalab.com is the domain name
used in the enterprise. This value should match the Domain set in the CS1000E SIP Gateway
(Section 5.1.5).
sip-manipulation
name
description
split-headers
join-headers
header-rule
name
header-name
action
comparison-type
msg-type
methods
match-value
new-value
element-rule
name
parameter-name
type
action
match-val-type
comparison-type
match-value
new-value
AddDomain
FromDomain
From
manipulate
case-sensitive
request
From
uri-host
replace
any
case-sensitive
avayalab.com
< text removed for brevity >
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The example below shows the CS1K_To_CL sip manipulation. This manipulation uses the PAI
and History Info headers to determine the type of call redirection and builds a Diversion header
accordingly. The header rule AddDiversion1 specifies that if the P-Asserted-Identity header
does not have a phone number within the range 303-555-7100 to 303-5557199 (the DID range
specified by CenturyLink) and does not have a “reason” parameter in the History-Info header, a
static Diversion header will be created. The header rules AddDiversion2, AddDiversion3 and
AddDiversion4 will create a Diversion header based on all 3 redirection reasons with the user
and host gathered from the History-Info header.
sip-manipulation
name
CS1K_To_CL
description
CS1K_to_CL
split-headers
join-headers
header-rule
name
PAIRegex
header-name
P-Asserted-Identity
action
store
comparison-type
pattern-rule
msg-type
any
methods
INVITE
match-value
new-value
element-rule
name
chkUser
parameter-name
type
header-value
action
store
match-val-type
any
comparison-type
pattern-rule
match-value
(.*)(30355571)(.*)
new-value
header-rule
name
HistRegex
header-name
History-Info
action
store
comparison-type
pattern-rule
msg-type
any
methods
match-value
new-value
element-rule
name
GetReason1
parameter-name
type
header-value
action
store
match-val-type
any
comparison-type
pattern-rule
match-value
(.*)(reason)(.*)
new-value
element-rule
name
GetReason2
parameter-name
type
header-value
action
none
match-val-type
any
comparison-type
pattern-rule
match-value
(.*)(Moved)(.*)
new-value
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element-rule
name
GetReason3
parameter-name
type
header-value
action
none
match-val-type
any
comparison-type
pattern-rule
match-value
(.*)(Busy)(.*)
new-value
element-rule
name
GetReason4
parameter-name
type
header-value
action
none
match-val-type
any
comparison-type
pattern-rule
match-value
(.*)(Unavailable)(.*)
new-value
element-rule
name
GetUser
parameter-name
type
uri-user
action
store
match-val-type
any
comparison-type
pattern-rule
match-value
new-value
element-rule
name
GetHost
parameter-name
type
uri-host
action
store
match-val-type
any
comparison-type
pattern-rule
match-value
new-value
header-rule
name
AddDiversion1
header-name
Diversion
action
add
comparison-type
boolean
msg-type
request
methods
INVITE
match-value
(!$PAIRegex[0].$chkUser)&!$HistRegex[0].$GetReason1
new-value
"<sip:3035557104@avayalab.com;user=phone>"
header-rule
name
AddDiversion2
header-name
Diversion
action
add
comparison-type
boolean
msg-type
request
methods
INVITE
match-value
$HistRegex[0].$GetReason2
new-value
<sip:+$HistRegex[0].$GetUser.$0+@+$HistRegex[0].$GetHost.$0+>;privacy=off;reason=unconditional;
screen=no
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header-rule
name
AddDiversion3
header-name
Diversion
action
add
comparison-type
boolean
msg-type
request
methods
INVITE
match-value
$HistRegex[0].$GetReason3
new-value
<sip:+$HistRegex[0].$GetUser.$0+@+$HistRegex[0].$GetHost.$0+>;privacy=off;reason=user\busy;screen=no
header-rule
name
AddDiversion4
header-name
Diversion
action
add
comparison-type
boolean
msg-type
request
methods
INVITE
match-value
$HistRegex[0].$GetReason4
new-value
<sip:+$HistRegex[0].$GetUser.$0+@+$HistRegex[0].$GetHost.$0+>;privacy=off;reason=no\answer;screen=no
6.10. Steering Pools
Steering pools define the range of ports to be used for the RTP voice stream. Two steering pools
were defined; one for each realm.
The key steering pool (steering-pool) fields are:
ip-address: The address of the interface on the Acme Packet 3820.
start-port: An even number of the port that begins the range.
end-port: An odd number of the port that ends the range.
realm-id: The realm to which this steering pool is assigned
steering-pool
ip-address
start-port
end-port
realm-id
network-interface
last-modified-by
last-modified-date
steering-pool
ip-address
start-port
end-port
realm-id
network-interface
last-modified-by
last-modified-date
DDT; Reviewed:
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10.2.2.92
49152
65535
peer
admin@console
2012-06-06 15:07:34
10.64.19.150
49152
65535
core
admin@console
2012-06-06 15:08:02
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6.11. Local Policy
Local policy controls the routing of SIP calls from one realm to another.
The key local policy (local-policy) fields are:
from-address: A policy filter indicating the originating IP address to which this policy
applies. An asterisk (*) indicates any IP address.
to-address: A policy filter indicating the terminating IP address to which this policy
applies. An asterisk (*) indicates any IP address.
source-realm: A policy filter indicating the matching realm in order for the policy rules
to be applied.
policy-attribute:
o next-hop: The IP address where the message should be sent when the policy rules
match.
o realm: The realm associated with the next-hop IP address.
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In this case, the first policy provides a simple routing rule indicating that messages originating
from the peer realm are to be sent to the core realm via IP address 10.80.140.103 (CS1000E at
the enterprise). The second policy indicates that messages originating from the core realm are to
be sent to the peer realm via the Session Agent Group CL-OUT created in Section 6.8.
local-policy
from-address
*
to-address
*
source-realm
peer
description
activate-time
N/A
< text removed for brevity >
policy-attribute
next-hop
realm
action
10.80.140.103
core
none
< text removed for brevity >
local-policy
from-address
*
to-address
*
source-realm
core
description
activate-time
N/A
< text removed for brevity >
policy-attribute
next-hop
realm
SAG:CL-OUT
peer
< text removed for brevity >
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7. CenturyLink SIP Trunk Service Configuration
To use CenturyLink SIP Trunk Service, a customer must request the service from CenturyLink
using their sales processes. This process can be initiated by contacting CenturyLink via the
corporate web site at www.centurylink.com and requesting information via the online sales links
or telephone numbers
8. Verification
This section provides verification steps that may be performed in the field to verify that the
solution is configured properly.
8.1. Avaya Communication Server 1000E Verifications
This section illustrates sample verifications that may be performed using the Avaya CS1000E
Element Manager GUI.
8.1.1. IP Network Maintenance and Reports Commands
From Element Manager, navigate to System  IP Network  Maintenance and Reports as
shown below. In the resultant screen on the right, click the Gen CMD button.
The General Commands page is displayed as shown below.
A variety of commands are available by selecting an appropriate Group and Command from the
drop-down menus, and selecting Run.
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To check the status of the SIP Gateway to Acme Packet 3820 in the sample configuration, select
Sip from the Group menu and SIPGwShow from the Command menu. Click Run. The example
output below shows that Acme Packet (10.64.19.150, port 5060, TCP) has SIPNPM Status
Active.
The following screen shows a means to view registered SIP telephones. The screen shows the
output of the Command sigSetShowAll in Group SipLine. At the time this screen was
captured, the SIP telephone with DN 7108 was involved in an active call with the CenturyLink
SIP Trunk Service.
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The following screen shows a means to view IP UNIStim telephones. The screen shows the
output of the Command isetShow in Group Iset. At the time this screen was captured, the
UNIStim telephone with IP address 10.80.150.111 was involved in an active call with the
CenturyLink SIP Trunk Service.
8.1.2. System Maintenance Commands
A variety of system maintenance commands are available by navigating to System 
Maintenance using Element Manager. The user can navigate the maintenance commands using
either the Select by Overlay approach or the Select by Functionality approach.
The following screen shows an example where Select by Overlay has been chosen. The various
overlays are listed, and the LD 96 – D-Channel is selected.
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On the preceding screen, if D-Channel Diagnostics is selected on the right, a screen such as the
following is displayed. D-Channel number 15, which is used in the sample configuration, is
established EST and active ACTV.
8.2. Acme Packet 3820 Net-Net ® Session Director Verifications
Below is a list of commands used to verify the configuration of the Acme Packet 3820.
show running-config – Displays the current config
verify-config – Displays any errors or warnings in the configuration.
show prom-info all – Displays the all prom information including serial number,
hardware revision, manufacturing date, part numbers and more
show sipd sessions all – Will display all of the active SIP sessions that are
currently traversing the SBC, including the To, From, Call-ID.
show support-info - Outputs all of the system level info, including hardware
specifics, licensing info, current call volume, etc.
show health - For a redundant system will give a status of synchronized
processes and an overview of failover history
show sipd invite - Will display a chart of all recent SIP requests and responses
display-alarms - Alarm log output of recent and current alarms
show logfile sipmsg.log - Will output the contents of the sipmsg.log without
having to FTP this file off the SBC
9. Conclusion
These Application Notes describe the configuration necessary to connect Avaya Communication
Server 1000E and Acme Packet 3820 Net-Net Session Director to the CenturyLink SIP Trunk
Service (Legacy Qwest). The CenturyLink SIP Trunk Service is a SIP-based Voice over IP
solution for customers ranging from small businesses to large enterprises. The CenturyLink SIP
Trunk Service provides businesses a flexible, cost-saving alternative to traditional hardwired
telephony trunks.
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10. Additional References
This section references the documentation relevant to these Application Notes. Additional
Avaya product documentation is available at http://support.avaya.com.
[1] Avaya Communication Server 1000E Installation and Commissioning, November 2010,
Document Number NN43041-310.
[2] Feature Listing Reference Avaya Communication Server 1000, November 2010, Document
Number NN43001-111, 05.01.
[3] RFC 3261 SIP: Session Initiation Protocol, http://www.ietf.org/
[4] Signaling Server IP Line Applications Fundamentals Avaya Communication Server 1000,
Document Number NN43001-125, 03.09 October 2011
[5] Linux Platform Base and Applications Installation and Commissioning Avaya Communication
Server 1000, Document Number NN43001-315, 05.18 January 2012
[6] SIP Software for Avaya 1100 Series IP Deskphones-Administration , Document Number
NN43170-600, Standard 04.02 December 2011
[7] Acme Packet, “Net-Net 4000 S-CX6.3.0 ACLI Configuration Guide”, 400-0061-62, Nov 2009
[8] Acme Packet, “Net-Net 3800 Series And Net-Net 4500 SSM2 Installation Guide”, 400-011420, Apr 2010
[9] Acme Packet, “Net-Net 3820 Hardware Installation Guide”, 400-0134-10, Mar 2011
Appendix A: Acme Packet 3820 Configuration File
Included below is the Acme Packet 3820 configuration used during the compliance testing. The
contents of the configuration can be shown by using the ACLI command show running-config
at the Acme Packet 3820.
ACMESYSTEM# show running-config
local-policy
from-address
*
to-address
*
source-realm
peer
description
activate-time
deactivate-time
state
policy-priority
last-modified-by
last-modified-date
policy-attribute
next-hop
realm
action
terminate-recursion
carrier
start-time
end-time
days-of-week
cost
app-protocol
state
methods
media-profiles
DDT; Reviewed:
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N/A
N/A
enabled
none
admin@10.80.150.50
2012-06-28 16:05:37
10.80.140.103
core
none
disabled
0000
2400
U-S
0
SIP
enabled
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lookup
next-key
eloc-str-lkup
eloc-str-match
single
disabled
local-policy
from-address
*
to-address
*
source-realm
core
description
activate-time
N/A
deactivate-time
N/A
state
enabled
policy-priority
none
last-modified-by
admin@10.80.150.38
last-modified-date
2011-11-03 17:39:11
policy-attribute
next-hop
SAG:CL-OUT
realm
peer
action
none
terminate-recursion
disabled
carrier
start-time
0000
end-time
2400
days-of-week
U-S
cost
0
app-protocol
SIP
state
enabled
methods
media-profiles
lookup
single
next-key
eloc-str-lkup
disabled
eloc-str-match
media-manager
state
enabled
latching
enabled
flow-time-limit
86400
initial-guard-timer
300
subsq-guard-timer
300
tcp-flow-time-limit
86400
tcp-initial-guard-timer
300
tcp-subsq-guard-timer
300
tcp-number-of-ports-per-flow
2
hnt-rtcp
disabled
algd-log-level
NOTICE
mbcd-log-level
NOTICE
red-flow-port
1985
red-mgcp-port
1986
red-max-trans
10000
red-sync-start-time
5000
red-sync-comp-time
1000
media-policing
enabled
max-signaling-bandwidth
10000000
max-untrusted-signaling
100
min-untrusted-signaling
30
app-signaling-bandwidth
0
tolerance-window
30
rtcp-rate-limit
0
trap-on-demote-to-deny
disabled
syslog-on-demote-to-deny
disabled
syslog-on-demote-to-untrusted disabled
anonymous-sdp
disabled
arp-msg-bandwidth
32000
fragment-msg-bandwidth
0
rfc2833-timestamp
disabled
default-2833-duration
100
rfc2833-end-pkts-only-for-non-sig enabled
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translate-non-rfc2833-event
media-supervision-traps
dnsalg-server-failover
last-modified-by
last-modified-date
network-interface
name
sub-port-id
description
hostname
ip-address
pri-utility-addr
sec-utility-addr
netmask
gateway
sec-gateway
gw-heartbeat
state
heartbeat
retry-count
retry-timeout
health-score
dns-ip-primary
dns-ip-backup1
dns-ip-backup2
dns-domain
dns-timeout
hip-ip-list
ftp-address
icmp-address
snmp-address
telnet-address
ssh-address
signaling-mtu
last-modified-by
last-modified-date
network-interface
name
sub-port-id
description
hostname
ip-address
pri-utility-addr
sec-utility-addr
netmask
gateway
sec-gateway
gw-heartbeat
state
heartbeat
retry-count
retry-timeout
health-score
dns-ip-primary
dns-ip-backup1
dns-ip-backup2
dns-domain
dns-timeout
hip-ip-list
ftp-address
icmp-address
snmp-address
telnet-address
ssh-address
signaling-mtu
last-modified-by
last-modified-date
phy-interface
name
operation-type
DDT; Reviewed:
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disabled
disabled
disabled
admin@10.80.150.38
2011-11-01 12:25:41
M00
0
PUBLIC
10.2.2.92
255.255.255.128
10.2.2.1
disabled
0
0
1
0
11
10.2.2.92
10.2.2.92
0
admin@10.80.150.50
2012-06-06 14:40:39
M10
0
PRIVATE
10.64.19.150
255.255.255.0
10.64.19.1
disabled
0
0
1
0
10.80.150.201
avayalab.com
11
10.64.19.150
10.64.19.150
0
admin@10.80.150.50
2012-06-06 14:42:37
M00
Media
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port
slot
virtual-mac
admin-state
auto-negotiation
duplex-mode
speed
overload-protection
last-modified-by
last-modified-date
phy-interface
name
operation-type
port
slot
virtual-mac
admin-state
auto-negotiation
duplex-mode
speed
overload-protection
last-modified-by
last-modified-date
realm-config
identifier
description
addr-prefix
network-interfaces
mm-in-realm
mm-in-network
mm-same-ip
mm-in-system
bw-cac-non-mm
msm-release
qos-enable
generate-UDP-checksum
max-bandwidth
fallback-bandwidth
max-priority-bandwidth
max-latency
max-jitter
max-packet-loss
observ-window-size
parent-realm
dns-realm
media-policy
media-sec-policy
srtp-msm-passthrough
in-translationid
out-translationid
in-manipulationid
out-manipulationid
manipulation-string
manipulation-pattern
class-profile
average-rate-limit
access-control-trust-level
invalid-signal-threshold
maximum-signal-threshold
untrusted-signal-threshold
nat-trust-threshold
deny-period
cac-failure-threshold
untrust-cac-failure-threshold
ext-policy-svr
diam-e2-address-realm
symmetric-latching
pai-strip
trunk-context
DDT; Reviewed:
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0
0
enabled
enabled
FULL
100
disabled
admin@console
2011-11-01 09:59:56
M10
Media
0
1
enabled
enabled
FULL
100
disabled
admin@console
2011-11-01 10:00:38
peer
0.0.0.0
M00:0
enabled
enabled
enabled
enabled
disabled
disabled
disabled
disabled
0
0
0
0
0
0
0
disabled
NatIP
0
none
0
0
0
0
30
0
0
disabled
disabled
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early-media-allow
enforcement-profile
additional-prefixes
restricted-latching
restriction-mask
accounting-enable
user-cac-mode
user-cac-bandwidth
user-cac-sessions
icmp-detect-multiplier
icmp-advertisement-interval
icmp-target-ip
monthly-minutes
net-management-control
delay-media-update
refer-call-transfer
refer-notify-provisional
dyn-refer-term
codec-policy
codec-manip-in-realm
constraint-name
call-recording-server-id
xnq-state
hairpin-id
stun-enable
stun-server-ip
stun-server-port
stun-changed-ip
stun-changed-port
match-media-profiles
qos-constraint
sip-profile
sip-isup-profile
block-rtcp
hide-egress-media-update
last-modified-by
last-modified-date
realm-config
identifier
description
addr-prefix
network-interfaces
mm-in-realm
mm-in-network
mm-same-ip
mm-in-system
bw-cac-non-mm
msm-release
qos-enable
generate-UDP-checksum
max-bandwidth
fallback-bandwidth
max-priority-bandwidth
max-latency
max-jitter
max-packet-loss
observ-window-size
parent-realm
dns-realm
media-policy
media-sec-policy
srtp-msm-passthrough
in-translationid
out-translationid
in-manipulationid
out-manipulationid
manipulation-string
manipulation-pattern
class-profile
DDT; Reviewed:
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none
32
enabled
none
0
0
0
0
0
disabled
disabled
disabled
none
disabled
disabled
xnq-unknown
0
disabled
0.0.0.0
3478
0.0.0.0
3479
disabled
disabled
admin@10.80.150.38
2011-11-01 13:03:09
core
0.0.0.0
M10:0
enabled
enabled
enabled
enabled
disabled
disabled
disabled
disabled
0
0
0
0
0
0
0
disabled
CS1K_To_CL
AddDomain
Solution & Interoperability Test Lab Application Notes
©2012 Avaya Inc. All Rights Reserved.
69 of 85
CLCS1K75AP
average-rate-limit
access-control-trust-level
invalid-signal-threshold
maximum-signal-threshold
untrusted-signal-threshold
nat-trust-threshold
deny-period
cac-failure-threshold
untrust-cac-failure-threshold
ext-policy-svr
diam-e2-address-realm
symmetric-latching
pai-strip
trunk-context
early-media-allow
enforcement-profile
additional-prefixes
restricted-latching
restriction-mask
accounting-enable
user-cac-mode
user-cac-bandwidth
user-cac-sessions
icmp-detect-multiplier
icmp-advertisement-interval
icmp-target-ip
monthly-minutes
net-management-control
delay-media-update
refer-call-transfer
refer-notify-provisional
dyn-refer-term
codec-policy
codec-manip-in-realm
constraint-name
call-recording-server-id
xnq-state
hairpin-id
stun-enable
stun-server-ip
stun-server-port
stun-changed-ip
stun-changed-port
match-media-profiles
qos-constraint
sip-profile
sip-isup-profile
block-rtcp
hide-egress-media-update
last-modified-by
last-modified-date
session-agent
hostname
ip-address
port
state
app-protocol
app-type
transport-method
realm-id
egress-realm-id
description
carriers
allow-next-hop-lp
constraints
max-sessions
max-inbound-sessions
max-outbound-sessions
max-burst-rate
max-inbound-burst-rate
DDT; Reviewed:
SPOC 9/12/2012
0
none
0
0
0
0
30
0
0
disabled
disabled
none
32
enabled
none
0
0
0
0
0
disabled
disabled
disabled
none
disabled
disabled
xnq-unknown
0
disabled
0.0.0.0
3478
0.0.0.0
3479
disabled
disabled
admin@10.80.150.50
2012-06-21 12:20:52
10.80.140.103
10.80.140.103
5060
enabled
SIP
UDP
core
enabled
disabled
0
0
0
0
0
Solution & Interoperability Test Lab Application Notes
©2012 Avaya Inc. All Rights Reserved.
70 of 85
CLCS1K75AP
max-outbound-burst-rate
max-sustain-rate
max-inbound-sustain-rate
max-outbound-sustain-rate
min-seizures
min-asr
time-to-resume
ttr-no-response
in-service-period
burst-rate-window
sustain-rate-window
req-uri-carrier-mode
proxy-mode
redirect-action
loose-routing
send-media-session
response-map
ping-method
ping-interval
ping-send-mode
ping-all-addresses
ping-in-service-response-codes
out-service-response-codes
load-balance-dns-query
media-profiles
in-translationid
out-translationid
trust-me
request-uri-headers
stop-recurse
local-response-map
ping-to-user-part
ping-from-user-part
li-trust-me
in-manipulationid
out-manipulationid
manipulation-string
manipulation-pattern
p-asserted-id
trunk-group
max-register-sustain-rate
early-media-allow
invalidate-registrations
rfc2833-mode
rfc2833-payload
codec-policy
enforcement-profile
refer-call-transfer
refer-notify-provisional
reuse-connections
tcp-keepalive
tcp-reconn-interval
max-register-burst-rate
register-burst-window
sip-profile
sip-isup-profile
kpml-interworking
last-modified-by
last-modified-date
session-agent
hostname
ip-address
port
state
app-protocol
app-type
transport-method
realm-id
egress-realm-id
description
DDT; Reviewed:
SPOC 9/12/2012
0
0
0
0
5
0
0
0
0
0
0
None
Proxy
enabled
enabled
OPTIONS;hops=70
60
keep-alive
disabled
hunt
disabled
disabled
0
disabled
none
0
disabled
none
NONE
none
0
0
0
inherit
admin@10.80.150.50
2012-06-28 16:04:46
172.16.2.8
172.16.2.8
5060
enabled
SIP
UDP
peer
Solution & Interoperability Test Lab Application Notes
©2012 Avaya Inc. All Rights Reserved.
71 of 85
CLCS1K75AP
carriers
allow-next-hop-lp
constraints
max-sessions
max-inbound-sessions
max-outbound-sessions
max-burst-rate
max-inbound-burst-rate
max-outbound-burst-rate
max-sustain-rate
max-inbound-sustain-rate
max-outbound-sustain-rate
min-seizures
min-asr
time-to-resume
ttr-no-response
in-service-period
burst-rate-window
sustain-rate-window
req-uri-carrier-mode
proxy-mode
redirect-action
loose-routing
send-media-session
response-map
ping-method
ping-interval
ping-send-mode
ping-all-addresses
ping-in-service-response-codes
out-service-response-codes
load-balance-dns-query
media-profiles
in-translationid
out-translationid
trust-me
request-uri-headers
stop-recurse
local-response-map
ping-to-user-part
ping-from-user-part
li-trust-me
in-manipulationid
out-manipulationid
manipulation-string
manipulation-pattern
p-asserted-id
trunk-group
max-register-sustain-rate
early-media-allow
invalidate-registrations
rfc2833-mode
rfc2833-payload
codec-policy
enforcement-profile
refer-call-transfer
refer-notify-provisional
reuse-connections
tcp-keepalive
tcp-reconn-interval
max-register-burst-rate
register-burst-window
sip-profile
sip-isup-profile
kpml-interworking
last-modified-by
last-modified-date
session-agent
hostname
ip-address
DDT; Reviewed:
SPOC 9/12/2012
enabled
disabled
0
0
0
0
0
0
0
0
0
5
0
0
0
0
0
0
None
enabled
enabled
OPTIONS;hops=70
60
keep-alive
disabled
hunt
disabled
disabled
0
disabled
none
0
disabled
none
NONE
none
0
0
0
inherit
admin@10.80.150.38
2011-11-01 12:39:40
172.16.2.9
172.16.2.9
Solution & Interoperability Test Lab Application Notes
©2012 Avaya Inc. All Rights Reserved.
72 of 85
CLCS1K75AP
port
state
app-protocol
app-type
transport-method
realm-id
egress-realm-id
description
carriers
allow-next-hop-lp
constraints
max-sessions
max-inbound-sessions
max-outbound-sessions
max-burst-rate
max-inbound-burst-rate
max-outbound-burst-rate
max-sustain-rate
max-inbound-sustain-rate
max-outbound-sustain-rate
min-seizures
min-asr
time-to-resume
ttr-no-response
in-service-period
burst-rate-window
sustain-rate-window
req-uri-carrier-mode
proxy-mode
redirect-action
loose-routing
send-media-session
response-map
ping-method
ping-interval
ping-send-mode
ping-all-addresses
ping-in-service-response-codes
out-service-response-codes
load-balance-dns-query
media-profiles
in-translationid
out-translationid
trust-me
request-uri-headers
stop-recurse
local-response-map
ping-to-user-part
ping-from-user-part
li-trust-me
in-manipulationid
out-manipulationid
manipulation-string
manipulation-pattern
p-asserted-id
trunk-group
max-register-sustain-rate
early-media-allow
invalidate-registrations
rfc2833-mode
rfc2833-payload
codec-policy
enforcement-profile
refer-call-transfer
refer-notify-provisional
reuse-connections
tcp-keepalive
tcp-reconn-interval
max-register-burst-rate
register-burst-window
DDT; Reviewed:
SPOC 9/12/2012
5060
enabled
SIP
UDP
peer
enabled
disabled
0
0
0
0
0
0
0
0
0
5
0
0
0
0
0
0
None
enabled
enabled
OPTIONS;hops=70
60
keep-alive
disabled
hunt
disabled
disabled
0
disabled
none
0
disabled
none
NONE
none
0
0
0
Solution & Interoperability Test Lab Application Notes
©2012 Avaya Inc. All Rights Reserved.
73 of 85
CLCS1K75AP
sip-profile
sip-isup-profile
kpml-interworking
last-modified-by
last-modified-date
session-agent
hostname
ip-address
port
state
app-protocol
app-type
transport-method
realm-id
egress-realm-id
description
carriers
allow-next-hop-lp
constraints
max-sessions
max-inbound-sessions
max-outbound-sessions
max-burst-rate
max-inbound-burst-rate
max-outbound-burst-rate
max-sustain-rate
max-inbound-sustain-rate
max-outbound-sustain-rate
min-seizures
min-asr
time-to-resume
ttr-no-response
in-service-period
burst-rate-window
sustain-rate-window
req-uri-carrier-mode
proxy-mode
redirect-action
loose-routing
send-media-session
response-map
ping-method
ping-interval
ping-send-mode
ping-all-addresses
ping-in-service-response-codes
out-service-response-codes
load-balance-dns-query
media-profiles
in-translationid
out-translationid
trust-me
request-uri-headers
stop-recurse
local-response-map
ping-to-user-part
ping-from-user-part
li-trust-me
in-manipulationid
out-manipulationid
manipulation-string
manipulation-pattern
p-asserted-id
trunk-group
max-register-sustain-rate
early-media-allow
invalidate-registrations
rfc2833-mode
rfc2833-payload
codec-policy
DDT; Reviewed:
SPOC 9/12/2012
inherit
admin@10.80.150.38
2011-11-01 12:39:46
172.16.3.8
172.16.3.8
5060
enabled
SIP
UDP
peer
enabled
disabled
0
0
0
0
0
0
0
0
0
5
0
0
0
0
0
0
None
enabled
enabled
OPTIONS;hops=70
60
keep-alive
disabled
hunt
disabled
disabled
0
disabled
none
0
Solution & Interoperability Test Lab Application Notes
©2012 Avaya Inc. All Rights Reserved.
74 of 85
CLCS1K75AP
enforcement-profile
refer-call-transfer
refer-notify-provisional
reuse-connections
tcp-keepalive
tcp-reconn-interval
max-register-burst-rate
register-burst-window
sip-profile
sip-isup-profile
kpml-interworking
last-modified-by
last-modified-date
session-agent
hostname
ip-address
port
state
app-protocol
app-type
transport-method
realm-id
egress-realm-id
description
carriers
allow-next-hop-lp
constraints
max-sessions
max-inbound-sessions
max-outbound-sessions
max-burst-rate
max-inbound-burst-rate
max-outbound-burst-rate
max-sustain-rate
max-inbound-sustain-rate
max-outbound-sustain-rate
min-seizures
min-asr
time-to-resume
ttr-no-response
in-service-period
burst-rate-window
sustain-rate-window
req-uri-carrier-mode
proxy-mode
redirect-action
loose-routing
send-media-session
response-map
ping-method
ping-interval
ping-send-mode
ping-all-addresses
ping-in-service-response-codes
out-service-response-codes
load-balance-dns-query
media-profiles
in-translationid
out-translationid
trust-me
request-uri-headers
stop-recurse
local-response-map
ping-to-user-part
ping-from-user-part
li-trust-me
in-manipulationid
out-manipulationid
manipulation-string
manipulation-pattern
DDT; Reviewed:
SPOC 9/12/2012
disabled
none
NONE
none
0
0
0
inherit
admin@10.80.150.50
2012-06-18 10:23:25
172.16.3.9
172.16.3.9
5060
enabled
SIP
UDP
peer
enabled
disabled
0
0
0
0
0
0
0
0
0
5
0
0
0
0
0
0
None
enabled
enabled
OPTIONS;hops=70
60
keep-alive
disabled
hunt
disabled
disabled
Solution & Interoperability Test Lab Application Notes
©2012 Avaya Inc. All Rights Reserved.
75 of 85
CLCS1K75AP
p-asserted-id
trunk-group
max-register-sustain-rate
early-media-allow
invalidate-registrations
rfc2833-mode
rfc2833-payload
codec-policy
enforcement-profile
refer-call-transfer
refer-notify-provisional
reuse-connections
tcp-keepalive
tcp-reconn-interval
max-register-burst-rate
register-burst-window
sip-profile
sip-isup-profile
kpml-interworking
last-modified-by
last-modified-date
session-group
group-name
description
state
app-protocol
strategy
dest
0
disabled
none
0
disabled
none
NONE
none
0
0
0
inherit
admin@10.80.150.50
2012-06-18 10:23:57
CL-OUT
enabled
SIP
Hunt
172.16.3.8
172.16.2.8
trunk-group
sag-recursion
stop-sag-recurse
last-modified-by
last-modified-date
sip-config
state
operation-mode
dialog-transparency
home-realm-id
egress-realm-id
nat-mode
registrar-domain
registrar-host
registrar-port
register-service-route
init-timer
max-timer
trans-expire
invite-expire
inactive-dynamic-conn
enforcement-profile
pac-method
pac-interval
pac-strategy
pac-load-weight
pac-session-weight
pac-route-weight
pac-callid-lifetime
pac-user-lifetime
red-sip-port
red-max-trans
red-sync-start-time
red-sync-comp-time
add-reason-header
sip-message-len
enum-sag-match
extra-method-stats
registration-cache-limit
register-use-to-for-lp
DDT; Reviewed:
SPOC 9/12/2012
enabled
401,407
admin@10.80.150.50
2012-06-18 10:27:19
enabled
dialog
enabled
core
core
None
0
always
500
4000
32
180
32
10
PropDist
1
1
1
600
3600
1988
10000
5000
1000
disabled
4096
disabled
disabled
0
disabled
Solution & Interoperability Test Lab Application Notes
©2012 Avaya Inc. All Rights Reserved.
76 of 85
CLCS1K75AP
options
refer-src-routing
add-ucid-header
proxy-sub-events
allow-pani-for-trusted-only
pass-gruu-contact
sag-lookup-on-redirect
set-disconnect-time-on-bye
last-modified-by
last-modified-date
sip-interface
state
realm-id
description
sip-port
address
port
transport-protocol
tls-profile
multi-home-addrs
allow-anonymous
ims-aka-profile
carriers
trans-expire
invite-expire
max-redirect-contacts
proxy-mode
redirect-action
contact-mode
nat-traversal
nat-interval
tcp-nat-interval
registration-caching
min-reg-expire
registration-interval
route-to-registrar
secured-network
teluri-scheme
uri-fqdn-domain
trust-mode
max-nat-interval
nat-int-increment
nat-test-increment
sip-dynamic-hnt
stop-recurse
port-map-start
port-map-end
in-manipulationid
out-manipulationid
manipulation-string
manipulation-pattern
sip-ims-feature
subscribe-reg-event
operator-identifier
anonymous-priority
max-incoming-conns
per-src-ip-max-incoming-conns
inactive-conn-timeout
untrusted-conn-timeout
network-id
ext-policy-server
default-location-string
charging-vector-mode
charging-function-address-mode
ccf-address
ecf-address
term-tgrp-mode
implicit-service-route
rfc2833-payload
rfc2833-mode
DDT; Reviewed:
SPOC 9/12/2012
max-udp-length=0
disabled
disabled
disabled
disabled
disabled
disabled
admin@10.80.150.38
2011-11-21 17:43:22
enabled
peer
10.2.2.92
5060
UDP
agents-only
0
0
0
none
none
30
90
disabled
300
3600
disabled
disabled
disabled
all
3600
10
30
disabled
401,407
0
0
disabled
disabled
none
0
0
0
0
pass
pass
none
disabled
101
transparent
Solution & Interoperability Test Lab Application Notes
©2012 Avaya Inc. All Rights Reserved.
77 of 85
CLCS1K75AP
constraint-name
response-map
local-response-map
ims-aka-feature
enforcement-profile
route-unauthorized-calls
tcp-keepalive
add-sdp-invite
add-sdp-profiles
sip-profile
sip-isup-profile
tcp-conn-dereg
register-keep-alive
kpml-interworking
tunnel-name
last-modified-by
last-modified-date
sip-interface
state
realm-id
description
sip-port
address
port
transport-protocol
tls-profile
multi-home-addrs
allow-anonymous
ims-aka-profile
carriers
trans-expire
invite-expire
max-redirect-contacts
proxy-mode
redirect-action
contact-mode
nat-traversal
nat-interval
tcp-nat-interval
registration-caching
min-reg-expire
registration-interval
route-to-registrar
secured-network
teluri-scheme
uri-fqdn-domain
trust-mode
max-nat-interval
nat-int-increment
nat-test-increment
sip-dynamic-hnt
stop-recurse
port-map-start
port-map-end
in-manipulationid
out-manipulationid
manipulation-string
manipulation-pattern
sip-ims-feature
subscribe-reg-event
operator-identifier
anonymous-priority
max-incoming-conns
per-src-ip-max-incoming-conns
inactive-conn-timeout
untrusted-conn-timeout
network-id
ext-policy-server
default-location-string
charging-vector-mode
DDT; Reviewed:
SPOC 9/12/2012
disabled
none
disabled
0
none
disabled
admin@10.80.150.50
2012-06-06 15:06:55
enabled
core
10.64.19.150
5060
TCP
all
0
0
0
none
none
30
90
disabled
300
3600
disabled
disabled
disabled
all
3600
10
30
disabled
401,407
0
0
disabled
disabled
none
0
0
0
0
pass
Solution & Interoperability Test Lab Application Notes
©2012 Avaya Inc. All Rights Reserved.
78 of 85
CLCS1K75AP
charging-function-address-mode
ccf-address
ecf-address
term-tgrp-mode
implicit-service-route
rfc2833-payload
rfc2833-mode
constraint-name
response-map
local-response-map
ims-aka-feature
enforcement-profile
route-unauthorized-calls
tcp-keepalive
add-sdp-invite
add-sdp-profiles
sip-profile
sip-isup-profile
tcp-conn-dereg
register-keep-alive
kpml-interworking
tunnel-name
last-modified-by
last-modified-date
sip-manipulation
name
description
split-headers
join-headers
header-rule
name
header-name
action
comparison-type
msg-type
methods
match-value
new-value
element-rule
name
parameter-name
type
action
match-val-type
comparison-type
match-value
new-value
header-rule
name
header-name
action
comparison-type
msg-type
methods
match-value
new-value
element-rule
name
parameter-name
type
action
match-val-type
comparison-type
match-value
new-value
header-rule
name
header-name
action
comparison-type
DDT; Reviewed:
SPOC 9/12/2012
pass
none
disabled
101
transparent
disabled
none
disabled
0
none
disabled
admin@10.80.150.50
2012-06-18 10:34:11
NatIP
natFROM
From
manipulate
case-sensitive
request
natHost
uri-host
replace
any
case-sensitive
$LOCAL_IP
natTO
To
manipulate
case-sensitive
request
natHost
uri-host
replace
any
case-sensitive
$REMOTE_IP
natPAI
P-Asserted-Identity
manipulate
case-sensitive
Solution & Interoperability Test Lab Application Notes
©2012 Avaya Inc. All Rights Reserved.
79 of 85
CLCS1K75AP
msg-type
methods
match-value
new-value
element-rule
name
parameter-name
type
action
match-val-type
comparison-type
match-value
new-value
header-rule
name
header-name
action
comparison-type
msg-type
methods
match-value
new-value
element-rule
name
parameter-name
type
action
match-val-type
comparison-type
match-value
new-value
header-rule
name
header-name
action
comparison-type
msg-type
methods
match-value
new-value
element-rule
name
parameter-name
type
action
match-val-type
comparison-type
match-value
new-value
header-rule
name
header-name
action
comparison-type
msg-type
methods
match-value
new-value
header-rule
name
header-name
action
comparison-type
msg-type
methods
match-value
new-value
header-rule
name
header-name
DDT; Reviewed:
SPOC 9/12/2012
any
natHost
uri-host
replace
any
case-sensitive
$LOCAL_IP
natRequest
Request-URI
manipulate
case-sensitive
request
natHost
uri-host
replace
any
case-sensitive
$REMOTE_IP
natDiversion
Diversion
manipulate
case-sensitive
request
natHost
uri-host
replace
any
case-sensitive
$LOCAL_IP
removeHist
History-Info
delete
case-sensitive
any
removeRPI
Remote-Party-ID
delete
case-sensitive
any
removeXNTe164
X-nt-e164-clid
Solution & Interoperability Test Lab Application Notes
©2012 Avaya Inc. All Rights Reserved.
80 of 85
CLCS1K75AP
action
delete
comparison-type
case-sensitive
msg-type
any
methods
match-value
new-value
header-rule
name
removeMultiMIME
header-name
Content-Type
action
manipulate
comparison-type
case-sensitive
msg-type
any
methods
match-value
new-value
element-rule
name
nt_mcdn
parameter-name
application/x-nt-mcdn-frag-hex;versionssLinux-7.50.17;base=x2611
type
mime
action
delete-element
match-val-type
any
comparison-type
case-sensitive
match-value
new-value
element-rule
name
nt_esn5
parameter-name
application/x-nt-esn5-frag-hex
type
mime
action
delete-element
match-val-type
any
comparison-type
case-sensitive
match-value
new-value
element-rule
name
nt_epid
parameter-name
application/x-nt-epid-fraghex;version=ssLinux-7.50.17;base=x2611
type
mime
action
delete-element
match-val-type
any
comparison-type
case-sensitive
match-value
new-value
last-modified-by
admin@10.80.150.50
last-modified-date
2012-06-28 16:56:45
sip-manipulation
name
AddDomain
description
split-headers
join-headers
header-rule
name
FromDomain
header-name
From
action
manipulate
comparison-type
case-sensitive
msg-type
request
methods
match-value
new-value
element-rule
name
From
parameter-name
type
uri-host
action
replace
match-val-type
any
comparison-type
case-sensitive
match-value
new-value
avayalab.com
header-rule
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CLCS1K75AP
name
header-name
action
comparison-type
msg-type
methods
match-value
new-value
element-rule
name
parameter-name
type
action
match-val-type
comparison-type
match-value
new-value
header-rule
name
header-name
action
comparison-type
msg-type
methods
match-value
new-value
element-rule
name
parameter-name
type
action
match-val-type
comparison-type
match-value
new-value
last-modified-by
last-modified-date
sip-manipulation
name
description
split-headers
join-headers
header-rule
name
header-name
action
comparison-type
msg-type
methods
match-value
new-value
element-rule
name
parameter-name
type
action
match-val-type
comparison-type
match-value
new-value
header-rule
name
header-name
action
comparison-type
msg-type
methods
match-value
new-value
element-rule
DDT; Reviewed:
SPOC 9/12/2012
PaiDomain
P-Asserted-Identity
manipulate
case-sensitive
request
Pai
uri-host
replace
any
case-sensitive
avayalab.com
natTO
To
manipulate
case-sensitive
request
To
uri-host
replace
any
case-sensitive
$REMOTE_IP
admin@10.80.150.50
2012-06-21 12:09:39
CS1K_To_CL
CS1K_to_CL
PAIRegex
P-Asserted-Identity
store
pattern-rule
any
INVITE
chkUser
header-value
store
any
pattern-rule
(.*)(30355571)(.*)
HistRegex
History-Info
store
pattern-rule
any
Solution & Interoperability Test Lab Application Notes
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CLCS1K75AP
name
parameter-name
type
action
match-val-type
comparison-type
match-value
new-value
element-rule
name
parameter-name
type
action
match-val-type
comparison-type
match-value
new-value
element-rule
name
parameter-name
type
action
match-val-type
comparison-type
match-value
new-value
element-rule
name
parameter-name
type
action
match-val-type
comparison-type
match-value
new-value
element-rule
name
parameter-name
type
action
match-val-type
comparison-type
match-value
new-value
element-rule
name
parameter-name
type
action
match-val-type
comparison-type
match-value
new-value
header-rule
name
header-name
action
comparison-type
msg-type
methods
match-value
new-value
header-rule
name
header-name
action
comparison-type
msg-type
methods
match-value
DDT; Reviewed:
SPOC 9/12/2012
GetReason1
header-value
store
any
pattern-rule
(.*)(reason)(.*)
GetReason2
header-value
none
any
pattern-rule
(.*)(Moved)(.*)
GetReason3
header-value
none
any
pattern-rule
(.*)(Busy)(.*)
GetReason4
header-value
none
any
pattern-rule
(.*)(Unavailable)(.*)
GetUser
uri-user
store
any
pattern-rule
GetHost
uri-host
store
any
pattern-rule
AddDiversion1
Diversion
add
boolean
request
INVITE
(!$PAIRegex[0].$chkUser)&!$HistRegex[0].$GetReason1
"<sip:3035557104@avayalab.com;user=phone>"
AddDiversion2
Diversion
add
boolean
request
INVITE
$HistRegex[0].$GetReason2
Solution & Interoperability Test Lab Application Notes
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CLCS1K75AP
new-value
<sip:+$HistRegex[0].$GetUser.$0+@+$HistRegex[0].$GetHost.$0+>;privacy=off;reason=unconditional;sc
reen=no
header-rule
name
AddDiversion3
header-name
Diversion
action
add
comparison-type
boolean
msg-type
request
methods
INVITE
match-value
$HistRegex[0].$GetReason3
new-value
<sip:+$HistRegex[0].$GetUser.$0+@+$HistRegex[0].$GetHost.$0+>;privacy=off;reason=user\busy;screen=no
header-rule
name
AddDiversion4
header-name
Diversion
action
add
comparison-type
boolean
msg-type
request
methods
INVITE
match-value
$HistRegex[0].$GetReason4
new-value
<sip:+$HistRegex[0].$GetUser.$0+@+$HistRegex[0].$GetHost.$0+>;privacy=off;reason=no\answer;screen=no
last-modified-by
admin@10.80.150.50
last-modified-date
2012-06-28 17:37:53
steering-pool
ip-address
10.2.2.92
start-port
49152
end-port
65535
realm-id
peer
network-interface
last-modified-by
admin@10.80.150.50
last-modified-date
2012-06-06 15:07:34
steering-pool
ip-address
10.64.19.150
start-port
49152
end-port
65535
realm-id
core
network-interface
last-modified-by
admin@10.80.150.50
last-modified-date
2012-06-06 15:08:02
system-config
hostname
description
location
mib-system-contact
mib-system-name
mib-system-location
snmp-enabled
enabled
enable-snmp-auth-traps
disabled
enable-snmp-syslog-notify
disabled
enable-snmp-monitor-traps
disabled
enable-env-monitor-traps
disabled
snmp-syslog-his-table-length
1
snmp-syslog-level
WARNING
system-log-level
WARNING
process-log-level
NOTICE
process-log-ip-address
0.0.0.0
process-log-port
0
collect
sample-interval
5
push-interval
15
boot-state
disabled
start-time
now
end-time
never
red-collect-state
disabled
red-max-trans
1000
red-sync-start-time
5000
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CLCS1K75AP
red-sync-comp-time
1000
push-success-trap-state
disabled
call-trace
disabled
internal-trace
disabled
log-filter
all
default-gateway
10.80.150.1
restart
enabled
exceptions
telnet-timeout
0
console-timeout
0
remote-control
enabled
cli-audit-trail
enabled
link-redundancy-state
disabled
source-routing
disabled
cli-more
disabled
terminal-height
24
debug-timeout
0
trap-event-lifetime
0
default-v6-gateway
::
ipv6-signaling-mtu
1500
ipv4-signaling-mtu
1500
cleanup-time-of-day
00:00
snmp-engine-id-suffix
snmp-agent-mode
v1v2
last-modified-by
admin@console
last-modified-date
2011-11-01 10:30:52
task done
ACMESYSTEM#
©2012
Avaya Inc. All Rights Reserved.
Avaya and the Avaya Logo are trademarks of Avaya Inc. All trademarks identified by ® and
™ are registered trademarks or trademarks, respectively, of Avaya Inc. All other trademarks
are the property of their respective owners. The information provided in these Application
Notes is subject to change without notice. The configurations, technical data, and
recommendations provided in these Application Notes are believed to be accurate and
dependable, but are presented without express or implied warranty. Users are responsible for
their application of any products specified in these Application Notes.
Please e-mail any questions or comments pertaining to these Application Notes along with the
full title name and filename, located in the lower right corner, directly to the Avaya
DevConnect Program at devconnect@avaya.com.
DDT; Reviewed:
SPOC 9/12/2012
Solution & Interoperability Test Lab Application Notes
©2012 Avaya Inc. All Rights Reserved.
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CLCS1K75AP
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