Application Notes for Avaya Aura® Communication Manager 6.0.1

Application Notes for Avaya Aura® Communication Manager 6.0.1
Avaya Solution & Interoperability Test Lab
Application Notes for Avaya Aura® Communication
Manager 6.0.1 and Avaya Aura® Session Border
Controller with Qwest iQ® SIP Trunk (version 6.5)
– Issue 1.0
Abstract
These Application Notes describe the steps to configure Session Initiation Protocol (SIP)
Trunking between Qwest iQ® SIP Trunk (version 6.5) and an Avaya SIP-enabled enterprise
solution. The Avaya solution consists of Avaya Aura® Communication Manager 6.0.1 and
Avaya Aura® Session Border Controller with various Avaya endpoints.
Information in these Application Notes has been obtained through DevConnect compliance
testing and additional technical discussions. Testing was conducted in the Avaya Solutions
and Interoperability Test Lab, utilizing Qwest SIP Trunk Services.
MEO; Reviewed:
SPOC 6/23/2011
Solution & Interoperability Test Lab Application Notes
©2011 Avaya Inc. All Rights Reserved.
1 of 39
CM601ASBC_QwSIP
1. Introduction
These Application Notes describe a sample configuration of Avaya Aura® Communication
Manager 6.0.1 and Avaya Aura® Session Border Controller integration with Qwest iQ® SIP
Trunk (version 6.5).
In the sample configuration, the Session Border Controller is used as an edge device between
Avaya Communication Manager and Qwest-SIP Trunk. The Session Border Controller
performs SIP header manipulation and provides Network Address Translation (NAT)
functionality to convert the private IP addressing to IP addressing appropriate for the QwestSIP Trunk access method.
The Communication Manager and Session Border Controller are directly connected with a
Communication Manager SIP trunk, Avaya Modular Messaging is also connected to the
Communication Manager through a SIP trunk.
Qwest SIP Trunk 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.
Qwest SIP Trunk 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). SIP Trunk 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.
2. General Test Approach and Test Results
The general test approach was to connect a simulated enterprise site to the Qwest SIP
Trunking service via the public Internet and exercise the features and functionality listed in
Section 2.1. The simulated enterprise site was comprised of Avaya Aura® Communication
Manager, the Avaya Aura® Session Border Controller, and various Avaya endpoints.
2.1. Interoperability Compliance Testing
The interoperability compliance testing focused on verifying inbound and outbound call flows
to / from Communication Manager 6.0.1 and Session Border Controller, and subsequent
redirection of inbound calls to Qwest-SIP Trunk.


Response to SIP OPTIONS queries.
Incoming PSTN calls to various phone types.
Phone types included H.323, digital, and analog telephones at the enterprise. Inbound
PSTN calls were routed to the enterprise across the SIP trunk from the service
provider. No SIP phones were tested because there was no SIP registrar in the
configuration.
MEO; Reviewed:
SPOC 6/23/2011
Solution & Interoperability Test Lab Application Notes
©2011 Avaya Inc. All Rights Reserved.
2 of 39
CM601ASBC_QwSIP













Outgoing PSTN calls from various phone types.
Phone types included H.323, digital, and analog telephones at the enterprise.
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).
Avaya one-X® Communicator supports two modes (Road Warrior and
Telecommuter). Each supported mode was tested. Avaya one-X® Communicator also
supports two Voice Over IP (VoIP) protocols: H.323 and SIP. Only the H.323 version
of Communicator was tested.
Various call types including: local, long distance, emergency, international, outbound
toll-free, operator (0) and 0+ dialing.
Codecs G.711MU, G.729A, and G.729AB were tested.
DTMF transmission using RFC 2833.
T.38 Fax
Caller ID presentation and Caller ID restriction.
Response to incomplete call attempts and trunk errors.
All trunks busy scenarios
Voicemail navigation for inbound and outbound calls.
User features such as hold and resume, internal call forwarding, transfer, and
conference.
Off-net call forwarding and mobility (extension to cellular).
Network re-direct using REFER
2.2. Support
2.2.1. Avaya
Avaya customers may obtain documentation and support for Avaya products by visiting
http://support.avaya.com. In the United States, (866) GO-AVAYA (866-462-8292) provides
access to overall sales and service support menus.
2.2.2. CenturyLinkTM
CenturyLink acquired Qwest in April 2011. Over time Qwest branded services and web sites
may be renamed by CenturyLink.
For technical support on the Qwest iQ SIP Trunk services, contact Customer Service at
http://www.qwest.com/business/products/products-and-services/voip-adv-voice/sip-trunk.html
Enter a phone number and click “Speak to us now” and Customer Service will call the entered
number, or select the “Email us” link to send an e-mail inquiry or click “Contact a rep” and fill
in the request information.
2.3. Test Results / Known Limitations
Interoperability testing of Qwest iQ SIP Trunk (version 6.5) was completed with successful
results for all test cases with the exception of the observations/limitations described below.
MEO; Reviewed:
SPOC 6/23/2011
Solution & Interoperability Test Lab Application Notes
©2011 Avaya Inc. All Rights Reserved.
3 of 39
CM601ASBC_QwSIP






No Error Indication if No Matching Codec Offered on Inbound Calls: If the
Communication Manager SIP trunk is improperly configured to have no matching codec
with the service provider and an inbound call is placed, the service provider only returns
a “488 Not Acceptable Here” response and the caller will hear a fast busy after 30
seconds. Codecs are normally agreed to upon turn-up so this condition should be
discovered at that time.
No Error Indication if No Matching Codec Offered on Outbound Calls: If the
Communication Manager SIP trunk is improperly configured to have no matching codec
with the service provider and an outbound call is placed, the service provider only
returns a “487 Request Terminated” response. The caller will hear a fast busy and the
called party will hear one ring before the call is terminated. Codecs are normally agreed
to upon turn-up so this condition should be discovered at that time.
No Support for G.729B: Qwest SIP Trunk does not support G.729B codec.
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/Qwest SIP Trunk
solution. It is listed here simply as an observation.
Asynchronous DTMF payload header values are not supported: Qwest SIP Trunk
does not support the use of a different DTMF payload header value in each direction of
a single call. This may occur if the media is re-directed from the Communication
Manager to an endpoint and the endpoint wishes to use a different DTMF payload
header value then was negotiated when the call was initially established. Qwest SIP
Trunk will send a re-INVITE to force the DTMF payload header value to be the same in
each direction. In response, Communication Manager will send a re-INVITE to force
the DTMF payload header value back to the original asynchronous values which allow
the DTMF payload header value to be the same end-to-end in the same direction (even
though the values are different in each direction). These re-INVITEs continue for
several minutes before one side gives up and tears down the call. This issue manifested
itself in two separate call scenarios during the compliance test described below. This
issue may occur in other call scenarios that were not tested.
- An inbound call from the PSTN to an enterprise Avaya phone that is
transferred back to the PSTN unattended will drop after several minutes.
This is because Qwest SIP Trunk uses a value of 100 for the DTMF payload
header value and the Communication Manager uses a value of 127 by default.
This scenario can be avoided by setting the “Telephone Event Payload Type”
on the trunk group, page 4 to 100.
- An inbound call from the PSTN to Avaya phone that is transferred back to
the PSTN using an attended transfer will drop after several minutes. This
is the same scenario as described above except for attended and the corrective
action is the same.
All Trunks Busy will ring from 7 – 40 seconds before fast busy: When all
Communication Manager trunk group members are busy, the caller will hear ringing for
MEO; Reviewed:
SPOC 6/23/2011
Solution & Interoperability Test Lab Application Notes
©2011 Avaya Inc. All Rights Reserved.
4 of 39
CM601ASBC_QwSIP
anywhere from 7 seconds to 40 seconds before finally hearing a fast busy. Qwest SIP
Trunk will send the call to the Avaya Communication Manager and it will erroneously
return a “403 Forbidden” instead of a “503 Service Unavailable”. The workaround for
this is to upgrade to one of the following Communication Manager loads: 5.2.1 SP9,
6.0.1 SP3, 6.2. Use of a 503 allows for a back-off time period and a retry by Qwest.

SIP Network REFER to an off-net extension is not supported: When
Communication Manager receives a PSTN call and tries to use a vector to automatically
re-direct using a SIP REFER to another PSTN extension, the call will drop. Qwest SIP
Trunk does not allow re-directs to/from non-Qwest PSTN numbers.

SIP REFER with transfer (consultative or blind) is not supported in Qwest iQ®
SIP Trunk : When an extension receives a call from a PSTN number and attempts to
transfer (either consultative or blind) the call to another PSTN extension, the call will
initially connect and then will be dropped as soon as the transfer is completed on the
enterprise user’s side. This is addressed in a future Qwest iQ® SIP Trunk release,
meanwhile the work-around is to have the Network Call Redirection field to n on page
4 of the trunk group form, refer to section 5.7.
MEO; Reviewed:
SPOC 6/23/2011
Solution & Interoperability Test Lab Application Notes
©2011 Avaya Inc. All Rights Reserved.
5 of 39
CM601ASBC_QwSIP
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 Qwest iQ® SIP Trunk. The Avaya CPE location simulates a customer site. At the edge of the
Avaya CPE location, an Avaya Aura® Session Border Controller provides NAT functionality and
SIP header manipulation. The Session Border Controller receives traffic from Qwest iQ® SIP
Trunk on port 5060 and sends traffic to the Qwest iQ® SIP Trunk using destination port 5060,
using the UDP protocol.
Figure 1: Avaya Interoperability Test Lab Configuration
3.1. Interoperability Compliance Testing
A separate trunk was created between Communication Manager and the Session Border
Controller to carry the service provider traffic; this was done so that any trunk or codec setting
MEO; Reviewed:
SPOC 6/23/2011
Solution & Interoperability Test Lab Application Notes
©2011 Avaya Inc. All Rights Reserved.
6 of 39
CM601ASBC_QwSIP
required by the service provider could be applied only to this trunk and not affect other
enterprise traffic. In addition, this trunk carried both inbound and outbound traffic.
For inbound calls, the calls flow from the service provider to the Session Border Controller
then to Communication Manager. Communication Manager uses the configured dial patterns
and routing policies to determine the recipient and any further incoming call treatment, such as
incoming digit translations and class of service restrictions may be performed.
Outbound calls to the PSTN are first processed by Communication Manager and may be
subject to outbound features such as automatic route selection, digit manipulation and class of
service restrictions. Once Communication Manager selects the proper SIP trunk, the call is
routed to Session Border Controller. The Session Border Controller forwards the call to
Qwest SIP Trunk.
4. Equipment and Software Validated
The following equipment and software were used for the sample configuration provided:
Equipment:
Avaya S8510 Server (Communication
Manager)
G650 Gateway
TN2312BP (IPSI)
TN2602AP (MedPro)
TN799DP (CLAN)
TN2224B (Digital Line Card)
TN793B (Analog Line Card)
Avaya S8800 Server (Session Border
Controller)
Avaya Modular Messaging (Application
Server)
Avaya Modular Messaging (Storage Server)
Avaya 9600-Series Telephones (H.323)
Avaya One-X Communicator (H.323)
Avaya 2400-Series and 6400-Series Digital
Telephones
Software:
Avaya Aura® Communication
Manager Release 6.0.1 load 510.1
HW36 FW 51
HW28 FW55
HW16 FW38
HW12
HW6
Avaya Aura® Session Border
Controller Release 6.0 SBC Template
SBCT 6.0.0.1.5
Avaya Modular Messaging (MAS) 5.2
Service Pack 5 Patch 1
Avaya Modular Messaging (MSS) 5.2,
Build 5.2-11.0
Release 030909 - H.323 - 4625
Release 3.0 – H.323 -9630
Release 6.0 - H.323 - 9608, 9621
Release 6.0.1.16-SP1-25226
N/A
5. Configure Avaya Aura® Communication Manager
This section describes the procedure for configuring Communication Manager for Qwest SIP
Trunking. A SIP trunk is established between Communication Manager and the Avaya Aura®
Session Border Controller for use by signaling traffic to and from Qwest SIP Trunk. It is
MEO; Reviewed:
SPOC 6/23/2011
Solution & Interoperability Test Lab Application Notes
©2011 Avaya Inc. All Rights Reserved.
7 of 39
CM601ASBC_QwSIP
assumed the general installation of Communication Manager and Avaya G650 Media
Gateway has been previously completed and is not discussed here.
The Communication Manager configuration was performed using the System Access Terminal
(SAT). Some screens in this section have been abridged and highlighted for brevity and
clarity in presentation. Note that the IP addresses and phone numbers shown throughout these
Application Notes have been edited so that the actual public IP addresses of the network
elements and public PSTN numbers are not revealed.
5.1. Licensing and Capacity
Use the display system-parameters customer-options command to verify that the
Maximum Administered SIP Trunks value on Page 2 is sufficient to support the desired
number of simultaneous SIP calls across all SIP trunks at the enterprise including any trunks
to the service provider. The example shows that 24000 SIP trunks are available and 257 are in
use. The license file installed on the system controls the maximum values for these attributes.
If a required feature is not enabled or there is insufficient capacity, contact an authorized
Avaya sales representative to add additional capacity.
display system-parameters customer-options
OPTIONAL FEATURES
Page
IP PORT CAPACITIES
Maximum Administered H.323 Trunks:
Maximum Concurrently Registered IP Stations:
Maximum Administered Remote Office Trunks:
Maximum Concurrently Registered Remote Office Stations:
Maximum Concurrently Registered IP eCons:
Max Concur Registered Unauthenticated H.323 Stations:
Maximum Video Capable Stations:
Maximum Video Capable IP Softphones:
Maximum Administered SIP Trunks:
Maximum Administered Ad-hoc Video Conferencing Ports:
Maximum Number of DS1 Boards with Echo Cancellation:
Maximum TN2501 VAL Boards:
Maximum Media Gateway VAL Sources:
Maximum TN2602 Boards with 80 VoIP Channels:
Maximum TN2602 Boards with 320 VoIP Channels:
Maximum Number of Expanded Meet-me Conference Ports:
12000
18000
12000
18000
414
100
18000
18000
24000
24000
522
128
250
128
128
300
2 of
11
USED
0
6
0
0
0
0
0
0
257
0
0
0
0
0
2
0
On Page 3 of the System-Parameters Customer-Options form, verify that ARS is enabled.
MEO; Reviewed:
SPOC 6/23/2011
Solution & Interoperability Test Lab Application Notes
©2011 Avaya Inc. All Rights Reserved.
8 of 39
CM601ASBC_QwSIP
display system-parameters customer-options
OPTIONAL FEATURES
Abbreviated Dialing Enhanced List?
Access Security Gateway (ASG)?
Analog Trunk Incoming Call ID?
A/D Grp/Sys List Dialing Start at 01?
Answer Supervision by Call Classifier?
ARS?
ARS/AAR Partitioning?
ARS/AAR Dialing without FAC?
ASAI Link Core Capabilities?
ASAI Link Plus Capabilities?
Async. Transfer Mode (ATM) PNC?
Async. Transfer Mode (ATM) Trunking?
ATM WAN Spare Processor?
ATMS?
Attendant Vectoring?
y
n
y
y
y
y
y
n
n
n
n
n
n
y
Page
3 of
Audible Message Waiting?
Authorization Codes?
CAS Branch?
CAS Main?
Change COR by FAC?
Computer Telephony Adjunct Links?
Cvg Of Calls Redirected Off-net?
DCS (Basic)?
DCS Call Coverage?
DCS with Rerouting?
11
y
y
n
n
n
y
y
y
y
y
Digital Loss Plan Modification? y
DS1 MSP? y
DS1 Echo Cancellation? y
On Page 4 of the System-Parameters Customer-Options form, verify that IP Trunks, IP
Stations, and ISDN-PRI features are enabled. If the use of SIP REFER messaging will be
required for the call flows, verify that the ISDN/SIP Network Call Redirection feature is
enabled.
display system-parameters customer-options
OPTIONAL FEATURES
Page
Emergency Access to Attendant?
Enable 'dadmin' Login?
Enhanced Conferencing?
Enhanced EC500?
Enterprise Survivable Server?
Enterprise Wide Licensing?
ESS Administration?
Extended Cvg/Fwd Admin?
External Device Alarm Admin?
Five Port Networks Max Per MCC?
Flexible Billing?
Forced Entry of Account Codes?
Global Call Classification?
Hospitality (Basic)?
Hospitality (G3V3 Enhancements)?
IP Trunks?
IP Stations? y
y
y
y
y
n
n
y
y
y
n
n
y
y
y
y
y
4 of
11
ISDN Feature Plus?
ISDN/SIP Network Call Redirection?
ISDN-BRI Trunks?
ISDN-PRI?
Local Survivable Processor?
Malicious Call Trace?
Media Encryption Over IP?
Mode Code for Centralized Voice Mail?
n
y
y
y
n
y
n
n
Multifrequency Signaling?
Multimedia Call Handling (Basic)?
Multimedia Call Handling (Enhanced)?
Multimedia IP SIP Trunking?
y
y
y
y
IP Attendant Consoles? y
On Page 6 of the System-Parameters Customer-Options form, verify that any required call
center features are enabled. In the sample configuration, vectoring was used to refer calls to
alternate destinations using SIP NCR (Network Call Redirect). Vector variables were used to
include User-User Information (UUI) with the referred calls.
5.2. System Features
Use the change system-parameters feature command to set the Trunk-to-Trunk Transfer
field to all to allow incoming calls from the PSTN to be transferred to another PSTN endpoint.
MEO; Reviewed:
SPOC 6/23/2011
Solution & Interoperability Test Lab Application Notes
©2011 Avaya Inc. All Rights Reserved.
9 of 39
CM601ASBC_QwSIP
If for security reasons, incoming calls should not be allowed to transfer back to the PSTN then
leave the field set to none.
change system-parameters features
Page
FEATURE-RELATED SYSTEM PARAMETERS
Self Station Display Enabled? n
Trunk-to-Trunk Transfer: all
Automatic Callback with Called Party Queuing? n
Automatic Callback - No Answer Timeout Interval (rings): 3
Call Park Timeout Interval (minutes): 10
Off-Premises Tone Detect Timeout Interval (seconds): 20
AAR/ARS Dial Tone Required? y
1 of
19
On Page 9, verify that a text string has been defined to replace the Calling Party Number
(CPN) for restricted or unavailable calls. This text string is entered in the two fields
highlighted below. The compliance test used the value of anonymous for both.
change system-parameters features
FEATURE-RELATED SYSTEM PARAMETERS
Page
9 of
19
CPN/ANI/ICLID PARAMETERS
CPN/ANI/ICLID Replacement for Restricted Calls: anonymous
CPN/ANI/ICLID Replacement for Unavailable Calls: anonymous
DISPLAY TEXT
Identity When Bridging: principal
User Guidance Display? n
Extension only label for Team button on 96xx H.323 terminals? n
INTERNATIONAL CALL ROUTING PARAMETERS
Local Country Code:
International Access Code:
ENBLOC DIALING PARAMETERS
Enable Enbloc Dialing without ARS FAC? n
CALLER ID ON CALL WAITING PARAMETERS
Caller ID on Call Waiting Delay Timer (msec): 200
5.3. IP Node Names
Use the change node-names ip command to verify that node names have been previously
defined for the IP addresses of the clan of the Avaya S8510 Server running Communication
Manager and for the Session Border Controller. These node names will be needed for
defining the service provider signaling group in Section 5.6.
MEO; Reviewed:
SPOC 6/23/2011
Solution & Interoperability Test Lab Application Notes
©2011 Avaya Inc. All Rights Reserved.
10 of 39
CM601ASBC_QwSIP
change node-names ip
Page
1 of
2
IP NODE NAMES
Name
AuraSBC-Inside
Gateway1
Gateway254
MM
MedPro1A03
MedPro1A04
clan
default
procr
procr6
IP Address
205.3.0.1
205.3.0.1
205.3.0.254
205.3.0.56
205.3.0.222
205.3.0.223
205.3.0.221
0.0.0.0
205.3.0.200
::
5.4. Codecs
Use the change ip-codec-set command to define a list of codecs to use for calls between the
enterprise and the service provider. For the compliance test, codecs G.729A and G.711mu
( 10 of 10
administered node-names were displayed )
were
ip-codec-set
1. To
enter G.729A
and G.711MU in the
Use tested
'list using
node-names'
command
touse
seethese
all codecs,
the administered
node-names
Use 'change
node-names
xxx'
node-name 'xxx'
add acan
node-name
Audio
Codec column
of theiptable
in to
thechange
order ofa preference.
Defaultorvalues
be used for
all other fields. Silence suppression is normally set to n and packet size is standard at 20ms.
change ip-codec-set 1
Page
1 of
2
Page
2 of
2
IP Codec Set
Codec Set: 1
Audio
Silence
Frames
Packet
Codec
Suppression Per Pkt Size(ms)
1: G.711MU
n
2
20
2: G.729A
n
2
20
3:
4:
On5:
Page 2, set the Fax Mode to T.38-standard.
6:
7:
change ip-codec-set 1
IP Codec Set
Allow Direct-IP Multimedia? n
FAX
Modem
TDD/TTY
Clear-channel
Mode
t.38-standard
off
US
n
Redundancy
0
0
3
0
5.5. IP Network Region
Create a separate IP network region for the service provider trunk. This allows for separate
codec or quality of service settings to be used (if necessary) for calls between the enterprise
and the service provider versus calls within the enterprise or elsewhere. For the compliance
test, IP-network-region 1 was chosen for the service provider trunk. Use the change ipnetwork-region 1 command to configure region 1 with the following parameters:
MEO; Reviewed:
SPOC 6/23/2011
Solution & Interoperability Test Lab Application Notes
©2011 Avaya Inc. All Rights Reserved.
11 of 39
CM601ASBC_QwSIP





Set the Authoritative Domain field to match the SIP domain of the enterprise. In this
configuration, the domain name is avayalab.com. This name appears in the “From”
header of SIP messages originating from this IP region.
Enter a descriptive name in the Name field.
Enable IP-IP Direct Audio (shuffling) to allow audio traffic to be sent directly
between IP endpoints without using media resources in the Avaya Media Gateway.
Set both Intra-region and Inter-region IP-IP Direct Audio to yes. This is the default
setting. Shuffling can be further restricted at the trunk level on the Signaling Group
form.
Set the Codec Set field to the IP codec set defined in Section 5.4.
Default values can be used for all other fields.
change ip-network-region 1
Page
1 of
20
IP NETWORK REGION
Region: 1
Location:
Authoritative Domain: avayalab.com
Name: Enterprise
MEDIA PARAMETERS
Intra-region IP-IP Direct Audio: yes
Codec Set: 1
Inter-region IP-IP Direct Audio: yes
UDP Port Min: 2048
IP Audio Hairpinning? y
UDP Port Max: 8001
DIFFSERV/TOS PARAMETERS
Call Control PHB Value: 34
Audio PHB Value: 46
Video PHB Value: 26
802.1P/Q PARAMETERS
Call Control 802.1p Priority: 7
Audio 802.1p Priority: 6
Video 802.1p Priority: 5
AUDIO RESOURCE RESERVATION PARAMETERS
H.323 IP ENDPOINTS
RSVP Enabled? n
H.323 Link Bounce Recovery? y
Idle Traffic Interval (sec): 20
Keep-Alive Interval (sec): 5
Keep-Alive Count: 5
On Page 4, define the IP codec set to be used for traffic in region 1. Enter the desired IP codec
set in the codec set column of the row with destination region (dst rgn) 1. Default values may
be used for all other fields. The example below shows the settings used for the compliance
test. It indicates that codec set 1 will be used for calls in region 1 (the service provider region)
and region 1 (the enterprise side). Creating this table entry for ip network region 1 will
automatically create a complementary table entry on the ip network region 1 form for
destination region 1.
change ip-network-region 1
Source Region: 1
Inter Network Region Connection Management
dst codec direct
WAN-BW-limits
Video
Intervening
rgn set
WAN Units
Total Norm Prio Shr Regions
1
1
2
3
1
y
NoLimit
4
5
MEO; Reviewed:
SPOC 6/23/2011
Page
Solution & Interoperability Test Lab Application Notes
©2011 Avaya Inc. All Rights Reserved.
Dyn
CAC
4 of
I
G
A
R
n
A
G
L
all
20
M
t
c
e
t
12 of 39
CM601ASBC_QwSIP
5.6. Signaling Group
Use the add signaling-group command to create a signaling group between Communication
Manager and the Session Border Controller for use by the service provider trunk. This
signaling group is used for inbound and outbound calls between the service provider and the
enterprise. For the compliance test, signaling group 3 was used for this purpose and was
configured using the parameters highlighted below.












Set the Group Type field to sip.
Set the Transport Method to the recommended default value of tls (Transport Layer
Security). For ease of troubleshooting during testing, part of the compliance test was
conducted with the Transport Method set to tcp. The transport method specified here
is used between the Communication Manager and the Session Border Controller.
Set the Near-end Listen Port and Far-end Listen Port to a valid unused port instead
of the default well-known port value. (For TLS, the well-known port value is 5061 and
for TCP the well-known port value is 5060). The compliance test was conducted with
the Near-end Listen Port and Far-end Listen Port set to 5060.
Set the Near-end Node Name to clan. This node name maps to the IP address of the
clan in the G650 gateway as defined in the node-names ip screen shot in section 5.3.
Set the Far-end Node Name to AuraSBC-inside. This node name maps to the IP
address of the Session Border Controller Inside interface as defined in the nodenames-ip screen shot in section 5.3.
Set the Far-end Network Region to the IP network region for the service provider in
Section 5.5.
Set the Far-end Domain to the domain of the enterprise (usually an IP Address).
Set Direct IP-IP Audio Connections to y. This field will enable media shuffling on
the SIP trunk allowing Communication Manager to redirect media traffic directly
between the SIP trunk and the enterprise endpoint.
Set Initial IP-IP Direct Media to n. This field will enable media shuffling on the SIP
trunk allowing Communication Manager to redirect media traffic directly between the
SIP trunk and the enterprise endpoint. Both Direct and Initial IP-IP Direct Media need
to be set as indicated for Early Media to be Enabled.
Set the DTMF over IP field to rtp-payload. This value enables Communication
Manager to send DTMF transmissions using RFC 2833.
Set the Alternate Route Timer to 15. This defines the number of seconds the that
Communication Manager will wait for a response (other than 100 Trying) to an
outbound INVITE before selecting another route. If an alternate route is not defined,
then the call is cancelled after this interval.
Default values may be used for all other fields.
MEO; Reviewed:
SPOC 6/23/2011
Solution & Interoperability Test Lab Application Notes
©2011 Avaya Inc. All Rights Reserved.
13 of 39
CM601ASBC_QwSIP
change signaling-group 1
Page
1 of
1
SIGNALING GROUP
Group Number: 1
Group Type: sip
IMS Enabled? n
Transport Method: tcp
Q-SIP? n
IP Video? n
Peer Detection Enabled? y Peer Server: Others
Near-end Node Name: clan
Near-end Listen Port: 5060
SIP Enabled LSP? n
Enforce SIPS URI for SRTP? y
Far-end Node Name: AuraSBC-Inside
Far-end Listen Port: 5060
Far-end Network Region: 1
Far-end Domain: 67.148.000.000
Incoming Dialog Loopbacks: eliminate
DTMF over IP: rtp-payload
Session Establishment Timer(min): 3
Enable Layer 3 Test? y
H.323 Station Outgoing Direct Media? n
Bypass If IP Threshold Exceeded?
RFC 3389 Comfort Noise?
Direct IP-IP Audio Connections?
IP Audio Hairpinning?
Initial IP-IP Direct Media?
Alternate Route Timer(sec):
n
n
y
n
n
15
5.7. Trunk Group
Use the add trunk-group command to create a trunk group for the signaling group created in
Section 5.6. For the compliance test, trunk group 3 was configured using the parameters
highlighted below.








Set the Group Type field to sip.
Enter a descriptive name for the Group Name.
Enter an available trunk access code (TAC) that is consistent with the existing dial plan
in the TAC field.
Set the Service Type field to public-ntwrk.
Set Member Assignment Method to auto.
Set the Signaling Group to the signaling group shown in the previous step.
Set the Number of Members field to the number of trunk members in the SIP trunk
group. This value determines how many simultaneous SIP calls can be supported by
this trunk.
Default values were used for all other fields.
change trunk-group 1
Page
1 of
21
TRUNK GROUP
Group Number:
Group Name:
Direction:
Dial Access?
Queue Length:
Service Type:
MEO; Reviewed:
SPOC 6/23/2011
1
OUTSIDE CALL
two-way
n
0
public-ntwrk
Group Type: sip
CDR Reports: y
COR: 1
TN: 1
TAC: *101
Outgoing Display? n
Night Service:
Auth Code? n
Member Assignment Method: auto
Signaling Group: 1
Number of Members: 20
Solution & Interoperability Test Lab Application Notes
©2011 Avaya Inc. All Rights Reserved.
14 of 39
CM601ASBC_QwSIP
On Page 2, the Redirect On OPTIM Failure value is the amount of time (in milliseconds)
that Communication Manager will wait for a response (other than 100 Trying) to a pending
INVITE sent to an EC500 remote endpoint before selecting another route. If another route is
not defined, then the call is cancelled after this interval. This time interval should be set to a
value comparable to the Alternate Route Timer on the signaling group form described in
Section 5.6.
Verify that the Preferred Minimum Session Refresh Interval is set to a value acceptable to
the service provider. This value defines the interval that re-INVITEs must be sent to keep the
active session alive. For the compliance test, the value of 900 seconds was used.
change trunk-group 1
Group Type: sip
Page
2 of
21
TRUNK PARAMETERS
Unicode Name: auto
Redirect On OPTIM Failure: 20000
SCCAN? n
Digital Loss Group: 18
Preferred Minimum Session Refresh Interval(sec): 900
Disconnect Supervision - In? y
Out? y
XOIP
Treatment:
auto
Delay
Call
When
Accessed
Via Header
IGAR? n
On Page 4, set the
Network
Call Redirection
field
to n.Setup
Set the
Send
Diversion
field to y. This field provides additional information to the network if the call has been redirected. This is needed to support call forwarding of inbound calls back to the PSTN and
some Extension to Cellular (EC500) call scenarios.
Set the Telephone Event Payload Type to 100, the value preferred by Qwest SIP Trunk.
change trunk-group 1
Page
4 of
21
PROTOCOL VARIATIONS
Mark Users as Phone?
Prepend '+' to Calling Number?
Send Transferring Party Information?
Network Call Redirection?
Send Diversion Header?
Support Request History?
Telephone Event Payload Type:
Convert 180 to 183 for Early Media?
Always Use re-INVITE for Display Updates?
Identity for Calling Party Display:
Enable Q-SIP?
y
n
y
n
y
y
100
n
n
P-Asserted-Identity
n
5.8. Outbound Routing
In these Application Notes, the Automatic Route Selection (ARS) feature is used to route
outbound calls via the SIP trunk to the service provider. In the sample configuration, the
single digit 9 is used as the ARS access code. Enterprise callers will dial 9 to reach an
“outside line”. This common configuration is illustrated below with little elaboration. Use the
MEO; Reviewed:
SPOC 6/23/2011
Solution & Interoperability Test Lab Application Notes
©2011 Avaya Inc. All Rights Reserved.
15 of 39
CM601ASBC_QwSIP
change dialplan analysis command to define a dialed string beginning with 9 of length 1 as a
feature access code (fac).
change dialplan analysis
Page
DIAL PLAN ANALYSIS TABLE
Location: all
Dialed
String
1
10
2
3
7
7
8
9
*
*10
#
Total
Length
3
4
4
4
3
4
4
1
3
4
3
Call
Type
fac
ext
ext
ext
fac
ext
ext
fac
fac
dac
fac
Dialed
String
Total Call
Length Type
Dialed
String
1 of
12
Percent Full: 2
Total Call
Length Type
Use the change feature-access-codes command to configure 9 as the Auto Route Selection
(ARS) – Access Code 1.
change feature-access-codes
Page
1 of
FEATURE ACCESS CODE (FAC)
Abbreviated Dialing List1 Access Code: 137
Abbreviated Dialing List2 Access Code:
Abbreviated Dialing List3 Access Code: 160
Abbreviated Dial - Prgm Group List Access Code:
Announcement Access Code: 115
Answer Back Access Code: 116
Attendant Access Code:
Auto Alternate Routing (AAR) Access Code: *88
Auto Route Selection (ARS) - Access Code 1: 9
Access Code 2:
Automatic Callback Activation: 120
Deactivation: 121
Call Forwarding Activation Busy/DA: 122
All: 123
Deactivation: 124
Call Forwarding Enhanced Status: 112
Act: 113
Deactivation: 114
Call Park Access Code: 125
Call Pickup Access Code: 126
CAS Remote Hold/Answer Hold-Unhold Access Code:
CDR Account Code Access Code:
Change COR Access Code:
Change Coverage Access Code: 143
10
Use the change ars analysis command to configure the routing of dialed digits following the
first digit 9. The example below shows a subset of the dialed strings tested as part of the
compliance test. See Section 2.1 for the complete list of call types tested. All dialed strings
are mapped to route pattern 1 which contains the SIP trunk to the service provider (as defined
next).
MEO; Reviewed:
SPOC 6/23/2011
Solution & Interoperability Test Lab Application Notes
©2011 Avaya Inc. All Rights Reserved.
16 of 39
CM601ASBC_QwSIP
change ars analysis 0
Page
ARS DIGIT ANALYSIS TABLE
Location: all
Dialed
String
Total
Min Max
1
1
8
8
11
11
2
2
9
17
10
18
8
8
18
18
16
24
17
25
18
18
6
6
16
16
14
22
15
23
0
0
0
00
01
011
101xxxx0
101xxxx0
101xxxx01
101xxxx011
101xxxx1
10xxx0
10xxx0
10xxx01
10xxx011
Route
Pattern
1
1
1
deny
deny
1
deny
deny
deny
deny
deny
deny
deny
deny
deny
Call
Type
op
op
op
op
iop
intl
op
op
iop
intl
fnpa
op
op
iop
intl
1 of
2
Percent Full: 1
Node
Num
ANI
Reqd
n
n
n
n
n
n
n
n
n
n
n
n
n
n
n
The route pattern defines which trunk group will be used for the call and performs any
necessary digit manipulation. Use the change route-pattern command to configure the
parameters for the service provider trunk route pattern in the following manner. The example
below shows the values used for route pattern 2 during the compliance test.





Pattern Name: Enter a descriptive name.
Grp No: Enter the outbound trunk group for the SIP service provider. For the
compliance test, trunk group 3 was used.
FRL: Set the Facility Restriction Level (FRL) field to a level that allows access to this
trunk for all users that require it. The value of 0 is the least restrictive level.
Pfx Mrk: 1 The prefix mark (Pfx Mrk) of one will prefix any FNPA 10-digit number
with a 1 and leave numbers of any other length unchanged. This will ensure 1 + 10
digits are sent to the service provider for long distance North American Numbering
Plan (NANP) numbers. All HNPA 10 digit numbers are left unchanged.
LAR: next This is the routing preference for Look Ahead Routing.
change route-pattern 1
Page
Pattern Number: 1
Pattern Name: toAuraSBC
SCCAN? n
Secure SIP? n
Grp FRL NPA Pfx Hop Toll No. Inserted
No
Mrk Lmt List Del Digits
Dgts
1: 1
0
1
2: 3
0
1
3:
4:
5:
6:
BCC VALUE TSC CA-TSC
0 1 2 M 4 W
Request
1: y y y y
2: y y y y
3: y y y y
4: y y y y
5: y y y y
6:Reviewed:
y y y y
MEO;
SPOC 6/23/2011
y
y
y
y
y
y
n
n
n
n
n
n
n
n
n
n
n
n
1 of
3
DCS/
QSIG
Intw
n
n
n
n
n
n
IXC
user
user
user
user
user
user
ITC BCIE Service/Feature PARM
No. Numbering LAR
Dgts Format
Subaddress
rest
next
rest
none
rest
none
rest
none
rest
none
rest
none
Solution & Interoperability
Test Lab Application Notes
17 of 39
©2011 Avaya Inc. All Rights Reserved.
CM601ASBC_QwSIP
5.9. Vector Directory Numbers (VDNs) and Vectors for SIP NCR
This section describes the basic commands used to configure Vector Directory Numbers (VDNs)
and corresponding vectors. These vectors contain steps that invoke the Communication Manager
SIP Network Call Redirection (NCR) functionality. These Application Notes provide rudimentary
vector definitions to demonstrate and test the SIP NCR and UUI functionalities. In general, call
centers will use vector functionality that is more complex and tailored to individual needs. Call
centers may also use customer hosts running applications used in conjunction with Application
Enablement Services to define call routing and provide associated UUI. The definition and
documentation of those complex applications and associated vectors are beyond the scope of these
Application Notes. In Section 2.3, Test Results / Known Limitations, Qwest SIP Trunk does not
support SIP-NCR to an off-net PSTN number and therefore we suggest that Network Call
Redirection (trunk-group page 4) be turned off to support blind transfers and conference calls.
SIP NCR is allowed for internal call redirection to another internal extension and this section is
describing how to configure that. However, Network Call Redirection would have to be turned
on and would then affect call transfers.
5.9.1. Post-Answer Redirection to a PSTN Destination
This section provides an example configuration of a vector that will use post-answer redirection to
a PSTN destination. A corresponding detailed verification is provided in Section 8.1. In this
example, the inbound toll-free call is routed to VDN 3999 shown in the following screen. The
originally dialed service provider Toll Free number may be mapped to VDN 3999 by the incoming
call handling treatment for the inbound trunk group (described in section 5.10 below).
display vdn 3999
Page
1 of
3
VECTOR DIRECTORY NUMBER
Extension:
Name*:
Destination:
Attendant Vectoring?
Meet-me Conferencing?
Allow VDN Override?
COR:
TN*: 1
Measured:
3999 Acceptable
is associatedService
with vector
2, which
is
Level
(sec):
3999
Qwest Call Center
Vector Number
n
n
n
1
2
internal
VDN
shown
below. Vector 2 plays an announcement
20
and collects 5 digits (step 3) to answer the call. After the digit collection, the route-to number
(step 5)VDN
includes
~r3036267690
where the number 303-626-7690 is an internal destination. This
of Origin
Annc. Extension*:
step causes a REFER
message
to
1st Skill*: be sent where the Refer-To header includes 13035557690 as the
user portion.
2nd Skill*:
3rd Skill*:
* Follows VDN Override Rules
MEO; Reviewed:
SPOC 6/23/2011
Solution & Interoperability Test Lab Application Notes
©2011 Avaya Inc. All Rights Reserved.
18 of 39
CM601ASBC_QwSIP
display vector 2
Page
1 of
6
CALL VECTOR
Number: 2
Name: PreAns Redirect
Multimedia? n
Attendant Vectoring? n
Meet-me Conf? n
Lock? n
Basic? y
EAS? y
G3V4 Enhanced? y
ANI/II-Digits? y
ASAI Routing? y
Prompting? y
LAI? y G3V4 Adv Route? y
CINFO? y
BSR? y
Holidays? y
Variables? y
3.0 Enhanced? y
01 wait-time
2
secs hearing ringback
02 #
Collect 5 digits - which answers the call
03 collect
5
digits after announcement 3998
for none
04 #
Refer the call to the PSTN destination
05 route-to
number ~r3035557690
with cov n if unconditionally
06 #
If Refer fails, play announcement and disconnect
07 disconnect
after announcement 3997
08
5.10. Incoming Call Handling Treatment for Incoming Calls
In general, the “incoming call handling treatment” for a trunk group can be used to manipulate the
digits received for an incoming call if necessary. The toll-free number sent by Qwest SIP Trunk
can be mapped to an extension using the incoming call handling treatment of the receiving trunk
group. As an example, the following screen illustrates a conversion of toll-free number
8778620755 to extension 3003.
change inc-call-handling-trmt trunk-group 1
INCOMING CALL HANDLING TREATMENT
Service/
Number
Number
Del Insert
Feature
Len
Digits
public-ntwrk
10 8775550755
10 3003
public-ntwrk
10 3035557686
10 1000
public-ntwrk
10 3035557687
10 1001
public-ntwrk
10 3035557688
10 2000
public-ntwrk
10 3035557689
10 3000
public-ntwrk
10 3035557690
10 3001
public-ntwrk
10 3035557691
10 3002
public-ntwrk
10 3035557692
10 3003
public-ntwrk
10 3035557693
10 3999
public-ntwrk
10 3035557694
10 3005
public-ntwrk
10 6515555198
10 3000
public-ntwrk
10 6785559410
10 2000
public-ntwrk
10 8775550751
10 3003
Page
1 of
30
5.11. Modular Messaging Hunt Group
Although not specifically related to Qwest SIP Trunk, this section shows the hunt group used
for access to Avaya Modular Messaging. In the sample configuration, users with voice mail
have a coverage path containing hunt group 99. Users can dial extension 7999 to reach
Modular Messaging (e.g., for message retrieval). The following screen shows Page 1 of huntgroup 99.
MEO; Reviewed:
SPOC 6/23/2011
Solution & Interoperability Test Lab Application Notes
©2011 Avaya Inc. All Rights Reserved.
19 of 39
CM601ASBC_QwSIP
display hunt-group 99
Page
1 of
60
HUNT GROUP
Group Number:
Group Name:
Group Extension:
Group Type:
TN:
COR:
Security Code:
ISDN/SIP Caller Display:
99
MM
7999
ucd-mia
1
1
ACD?
Queue?
Vector?
Coverage Path:
Night Service Destination:
MM Early Answer?
Local Agent Preference?
n
n
n
n
n
mbr-name
The following screen shows Page 2 of hunt-group 99, which routes to the AAR access code *88
and Voice Mail Number 7999.
display hunt-group 99
Page
2 of
60
HUNT GROUP
Message Center: sip-adjunct
Voice Mail Number
Voice Mail Handle
7999
MM
Routing Digits
(e.g., AAR/ARS Access Code)
*88
5.12. AAR Routing to Avaya Modular Messaging
Although not specifically related to Qwest SIP Trunk, this section shows the AAR routing for the
number used in the hunt group in the previous section. The bold row shows that calls to the
number 7999, which is the Modular Messaging Group Extension for hunt group 99, will use
Route Pattern 2.
change aar analysis 7
Page
AAR DIGIT ANALYSIS TABLE
Location: all
Dialed
String
Total
Min Max
7999
4
4
Route
Pattern
2
Call
Type
unku
Node
Num
1 of
2
Percent Full: 1
ANI
Reqd
n
6. Avaya Aura® Session Border Controller Element
Manager Configuration
6.1. Initial Installation
This section displays basic configuration of the Session Border Controller from the installation
wizard. The initial configuration was completed by installing the Virtual Server Platform
(VSP) 6.0.1.0.5 and then logging into the web interface of the server (cdom) address. This
screen will verify the state of the dom-0 and cdom platforms (the State column below) and the
Application State of the Session Border Controller after it has been installed.
MEO; Reviewed:
SPOC 6/23/2011
Solution & Interoperability Test Lab Application Notes
©2011 Avaya Inc. All Rights Reserved.
20 of 39
CM601ASBC_QwSIP
After Installation of VSP is complete open the web interface to the CDOM and go to Virtual
Machine Management Solution Template
Select the location of the template:
Select the correct template for the server hardware:
MEO; Reviewed:
SPOC 6/23/2011
Solution & Interoperability Test Lab Application Notes
©2011 Avaya Inc. All Rights Reserved.
21 of 39
CM601ASBC_QwSIP
Select “Install” after verifying:
At this point, the installer will open a log that will show the progress of the installation. When
the installer gets to “Wait for User to Complete Data Entry”, another window will open for
input (ensure the browser has pop-ups enabled).
MEO; Reviewed:
SPOC 6/23/2011
Solution & Interoperability Test Lab Application Notes
©2011 Avaya Inc. All Rights Reserved.
22 of 39
CM601ASBC_QwSIP
In this version the domain that is listed is NOT optional and must be populated or installation
will fail, please refer to the release notes for further information.
The installer then prompts for passwords for the different accounts:
MEO; Reviewed:
SPOC 6/23/2011
Solution & Interoperability Test Lab Application Notes
©2011 Avaya Inc. All Rights Reserved.
23 of 39
CM601ASBC_QwSIP
Specify VPN preferences:
Next is the service provider input screen. For the Service Provider “Generic” was chosen.
MEO; Reviewed:
SPOC 6/23/2011
Solution & Interoperability Test Lab Application Notes
©2011 Avaya Inc. All Rights Reserved.
24 of 39
CM601ASBC_QwSIP
A summary of the input is displayed, confirm choices by clicking “Next Step”:
The progress page will re-appear as it continues the installation.
MEO; Reviewed:
SPOC 6/23/2011
Solution & Interoperability Test Lab Application Notes
©2011 Avaya Inc. All Rights Reserved.
25 of 39
CM601ASBC_QwSIP
Once the installation is completed, verify all application statuses:
MEO; Reviewed:
SPOC 6/23/2011
Solution & Interoperability Test Lab Application Notes
©2011 Avaya Inc. All Rights Reserved.
26 of 39
CM601ASBC_QwSIP
Please refer to Session Border Controller installation manuals and appropriate release notes for
further assistance. Also note, this version of the Session Border Controller is the first to
require the license file be loaded via WebLM.
6.2. Configuration File
The GUI installation wizard prompted for IP addresses, port numbers, domains, etc. and then
created the initial configuration file for the Session Border Controller. This configuration file
is listed below and has been appended, but the configured fields are shown.
#
# Copyright (c) 2004-2010 Acme Packet Inc.
# All Rights Reserved.
#
# File: /cxc/cxc.cfg
# Date: 15:15:21 Thu 2011-04-07
#
config cluster
config box 1
set hostname AA-SBC.avayalab.com
set timezone America/Denver
set name AA-SBC.avayalab.com
set identifier 00:ca:fe:41:41:30
config interface eth0
config ip inside
set ip-address static 205.3.000.000/24
config ssh
return
config snmp
set trap-target 205.3.000.000 162
return
config web
return
config web-service
set protocol https 8443
set authentication certificate "vsp\tls\certificate ws-cert"
return
config sip
set udp-port 5060 "" "" any 0
set tcp-port 5060 "" "" any 0
set tls-port 5061 "" "" TLS 0 "vsp\tls\certificate aasbc.p12"
return
config routing
config route Default
set gateway 205.5.000.000
return
config route Static0
set destination network 192.11.13.4/30
set gateway 205.3.0.0
return
config route Static1
set admin disabled
return
MEO; Reviewed:
SPOC 6/23/2011
Solution & Interoperability Test Lab Application Notes
©2011 Avaya Inc. All Rights Reserved.
27 of 39
CM601ASBC_QwSIP
return
return
return
config interface eth2
config ip outside
set ip-address static 205.168.000.000/25
config sip
set udp-port 5060 "" "" any 0
return
config media-ports
return
config routing
config route Default
set admin disabled
return
config route external-sip-media-1
set destination network 67.148.000.000/28
set gateway 205.168.1.1
return
return
config kernel-filter
config allow-rule allow-sip-udp-from-peer-1
set destination-port 5060
set source-address/mask 67.148.000.000/28
set protocol udp
return
config deny-rule deny-all-sip
set destination-port 5060
return
return
return
return
config cli
set prompt AA-SBC.avayalab.com
return
return
return
config vsp
set admin enabled
config default-session-config
config media
set anchor enabled
set rtp-stats enabled
return
config sip-directive
set directive allow
return
config log-alert
set apply-to-methods-for-filtered-logs
return
config third-party-call-control
set admin enabled
set handle-refer-locally disabled
return
return
config tls
MEO; Reviewed:
SPOC 6/23/2011
Solution & Interoperability Test Lab Application Notes
©2011 Avaya Inc. All Rights Reserved.
28 of 39
CM601ASBC_QwSIP
config default-ca
set ca-file /cxc/certs/sipca.pem
return
config certificate ws-cert
set certificate-file /cxc/certs/ws.cert
return
config certificate aasbc.p12
set certificate-file /cxc/certs/aasbc.p12
set passphrase-tag aasbc-cert-tag
return
return
config session-config-pool
config entry ToTelco
config to-uri-specification
set host next-hop
set user-param keep
return
config from-uri-specification
set host local-ip
set user-param keep
return
config request-uri-specification
set host next-hop
set user-param keep
return
config p-asserted-identity-uri-specification
set host local-ip
set user-param keep
return
config forking-settings
set outbound-arbiter-rule weighted-round-robin
return
config header-settings
return
return
config entry ToPBX
config to-uri-specification
set host next-hop-domain
return
config request-uri-specification
set host next-hop-domain
return
config header-settings
return
return
config entry Discard
config sip-directive
return
return
return
config dial-plan
config route Default
set priority 500
set location-match-preferred exclusive
set session-config vsp\session-config-pool\entry Discard
return
MEO; Reviewed:
SPOC 6/23/2011
Solution & Interoperability Test Lab Application Notes
©2011 Avaya Inc. All Rights Reserved.
29 of 39
CM601ASBC_QwSIP
config source-route FromTelco
set peer server "vsp\enterprise\servers\sip-gateway PBX"
set source-match server "vsp\enterprise\servers\sip-gateway Telco"
return
config source-route FromPBX
set peer server "vsp\enterprise\servers\sip-gateway Telco"
set source-match server "vsp\enterprise\servers\sip-gateway PBX"
return
return
config enterprise
config servers
config sip-gateway PBX
set domain avayalab.com
set failover-detection ping
set outbound-session-config-pool-entry vsp\session-config-pool\entry
ToPBX
config server-pool
config server PBX1
set host 205.3.000.000
set transport TCP
return
return
return
config sip-gateway Telco
set failover-detection ping
set outbound-session-config-pool-entry vsp\session-config-pool\entry
ToTelco
config server-pool
config server Telco1
set host 67.148.000.000
return
return
return
return
return
config dns
config resolver
config server 205.3.000.000
return
return
return
config settings
set read-header-max 8191
return
return
config external-services
return
config preferences
config gui-preferences
set enum-strings DatabaseName spotlite
set enum-strings SIPSourceHeader Refer-to
return
return
config permissions read-only
set config view
set actions disabled
MEO; Reviewed:
SPOC 6/23/2011
Solution & Interoperability Test Lab Application Notes
©2011 Avaya Inc. All Rights Reserved.
30 of 39
CM601ASBC_QwSIP
return
config users
config user admin
set password
0x0062cbb6cbe2cad687396c595c8460dba0ea4dc0a39c09dc77475f6be3
set permissions access\permissions superuser
return
config user cust
set password
0x00c7e90a870789f8cfcbbeb59233fac1a36075cbd8013a9a5acceaa387
set permissions access\permissions read-only
return
config user init
set password
0x0094cf691d6888e6b02f413213093a7585c31fa1624682b766a60f24de
set permissions access\permissions superuser
return
config user craft
set password
0x00eeb7c06047a5f92e29fff123b971cd57ec0dd0768d7bf072c911bc70
set permissions access\permissions superuser
return
config user dadmin
set password
0x00107d0dc0bb643d5f4c1043175f8ecf01924b7812bd530c54ac1e97be
set permissions access\permissions read-only
return
return
return
config features
return
7. Qwest iQ SIP Trunking Configuration
To use the Qwest iQ SIP Trunk Service, a customer must request service. The process can be
started by accessing the corporate web site at www.qwest.com and requesting information via the
online sales links or telephone numbers.
8. Verification Steps
This section provides verification steps that may be performed in the field to verify that the
solution is configured properly. This section also provides a list of useful troubleshooting
commands that can be used to troubleshoot the solution.
8.1. Avaya Aura® Session Border Controller Verification
This section illustrates verifications using the Session Border Controller and Wireshark to
illustrate key SIP messaging.
This section contains verification steps that may be performed using the Session Border
Controller. The status of the virtual machines can be checked via the System Platform Console
Domain of the server. The following screen, available via the Virtual Machine Management link
in the console domain, shows the “Running” State of the Session Border Controller.
MEO; Reviewed:
SPOC 6/23/2011
Solution & Interoperability Test Lab Application Notes
©2011 Avaya Inc. All Rights Reserved.
31 of 39
CM601ASBC_QwSIP
Click on the wrench icon to the left of the name “sbc” to access the element manager user interface
of the Session Border Controller.
8.1.1. Verify Connectivity to Qwest SIP Trunk
Using Wireshark, verify that entity links from the Session Border Controller (205.168.x.x) to
Qwest SIP Trunk (67.148.x.x) are communicating with SIP OPTION messages and 200 OK
responses.
8.1.2. Verify Connectivity to Communication Manager
Verify that the signaling group / trunk group between the Communication Manager and
Session Border Controller are up by using the status signaling group # and status trunkgroup commands.
status signaling-group 1
STATUS SIGNALING GROUP
Group ID: 1
Group Type: sip
.
Group State: in-service
MEO; Reviewed:
SPOC 6/23/2011
Solution & Interoperability Test Lab Application Notes
©2011 Avaya Inc. All Rights Reserved.
32 of 39
CM601ASBC_QwSIP
status trunk 1
Page
1
TRUNK GROUP STATUS
Member
Port
Service State
Mtce Connected Ports
Busy
0001/001 T00001
in-service/idle
no
0001/002 T00002
in-service/idle
no
0001/003 T00013
in-service/idle
no
0001/004 T00014
in-service/idle
no
0001/005 T00015
in-service/idle
no
0001/006 T00016
in-service/idle
no
0001/007 T00017
in-service/idle
no
0001/008 T00018
in-service/idle
no
0001/009 T00019
in-service/idle
no
0001/010
T00020 Border
in-service/idle
no Logs
8.1.3.
Session
Controller Call
0001/011 T00021
in-service/idle
no
Call
Log Ladder
Diagrams
visually displaysnothe call flow between the service provider, the
0001/012
T00022
in-service/idle
0001/013
T00023
in-service/idle
no Manager. To view this diagram, log into the
Session
Border
Controller
and Communication
0001/014
T00024
in-service/idle
no navigate to Call Logs, in Search Type: Enter
web
interface
on the Session
Border Controller,
All Sessions or Filter appropriately Session Diagram (for the appropriate call).
The following screen shows a portion of the ladder diagram for an inbound call. Note that the
activity for both the inside private and outside public side of the Session Border Controller can
be seen.
MEO; Reviewed:
SPOC 6/23/2011
Solution & Interoperability Test Lab Application Notes
©2011 Avaya Inc. All Rights Reserved.
33 of 39
CM601ASBC_QwSIP
Scroll down to continue the ladder diagram. The following screen shows the portion of the ladder
diagram for a call that is answered by a Communication Manager and released on the
PSTN/Qwest SIP Trunk side.
MEO; Reviewed:
SPOC 6/23/2011
Solution & Interoperability Test Lab Application Notes
©2011 Avaya Inc. All Rights Reserved.
34 of 39
CM601ASBC_QwSIP
At the top right of the screen, the session may be saved as a text or XML file. If the session is
saved as an XML file, using the Save as XML link, the xml file can be provided to support
personnel that can open the session on another Session Border Controller for analysis.
8.2. Communication Manager and Wireshark Verifications
8.2.1. Example Incoming Call from PSTN via Qwest SIP Trunk
DID and incoming toll-free calls arrive from Qwest SIP Trunk at the Session Border
Controller, which sends the call to Communication Manager on trunk 1 signaling group 1.
The following abridged Communication Manager “list trace” trace output shows a call incoming
on trunk group1. The PSTN telephone dialed 303-555-7694. The “inc-call-handling-trmt trunkgroup 1” maps the incoming number to an extension of a Communication Manager telephone
(x3005). Extension 3005 is an IP Telephone with IP Address 205.3.123.333 in Region 1. Initially,
the G650 Media Gateway (205.3.123.111) is used, but as can be seen in the final trace output,
once the call is answered, the final RTP media path is “ip-direct” from the IP Telephone
(205.3.123.333) to the “inside” of the Session Border Controller (205.3.123.222).
In Communication Manager Release 6, the tracing prints the Communication Manager release
version at the start of the trace, and intersperses the SIP messaging with the Communication
Manager processing.
MEO; Reviewed:
SPOC 6/23/2011
Solution & Interoperability Test Lab Application Notes
©2011 Avaya Inc. All Rights Reserved.
35 of 39
CM601ASBC_QwSIP
list trace tac *101
Page
1
LIST TRACE
time
data
16:24:22 TRACE STARTED 04/07/2011 CM Release String cold-00.1.510.1
16:25:07 SIP<INVITE sip:3035557694@avayalab.com:5060 SIP/2.0
16:25:07
active trunk-group 1 member 1 cid 0x208
16:25:07 SIP>SIP/2.0 180 Ringing
16:25:07
dial 3005
16:25:07
ring station
3005 cid 0x208
16:25:07
G711MU ss:off ps:20
rgn:1 [205.3.123.222]:7306
rgn:1 [205.3.123.111]:7672
16:25:07
G711MU ss:off ps:20
rgn:1 [205.3.123.333]:23512
rgn:1 [205.3.123.222]:7648
16:25:07
xoip options: fax:PT modem:off tty:US uid:0x50001
xoip ip: [205.3.3.222]:7648
16:25:07 SIP<PRACK sip:3035557694@avayalab.com:5060 SIP/2.0
16:25:09 SIP>SIP/2.0 200 OK
16:25:09
active station
3005 cid 0x208
16:25:09 SIP<ACK sip:3035557694@avayalab.com:5060 SIP/2.0
16:25:09 SIP>INVITE sip:3035551910@67.148.000.000:5060;transport=tc
16:25:09 SIP>p;maddr=205.3.3.208 SIP/2.0
16:25:09 SIP<SIP/2.0 100 Trying
16:25:09 SIP<SIP/2.0 200 OK
16:25:09 SIP>ACK sip:3035551910@67.148.000.000:5060;transport=tcp;m
16:25:09 SIP>addr=205.3.3.208 SIP/2.0
16:25:09
G711MU ss:off ps:20
rgn:1 [205.3.3.208]:23512
rgn:1 [205.3.3.235]:7306
16:25:09
G711MU
ss:off
The
following screen
shows
Pageps:20
2 of the output of the “status trunk 1/1” command pertaining
rgn:1 [205.3.3.235]:7306
the same call. Note
the
signaling
using
port 5060 between Communication Manager and the
rgn:1 [205.3.3.208]:23512
to
Session Border Controller. Note the media is “ip-direct” from the IP Telephone (205.3.3.235) to
the inside IP Address of the Session Border Controller (205.3.3.208) using G.711.
status trunk 1/1
Page
2 of
3
CALL CONTROL SIGNALING
Near-end Signaling Loc: 01A0217
Signaling
IP Address
Near-end: 205.3.3.221
Far-end: 205.3.3.208
H.245 Near:
H.245 Far:
H.245 Signaling Loc:
Port
: 5060
: 5060
H.245 Tunneled in Q.931? no
Audio Connection Type: ip-direct
Near-end Audio Loc:
Audio
IP Address
Near-end: 205.3.3.235
Far-end: 205.3.3.208
Authentication Type: None
Codec Type: G.711MU
Port
: 7306
: 23512
Video Near:
Video Far:
Video Port:
Video Near-end Codec:
Video Far-end Codec:
MEO; Reviewed:
SPOC 6/23/2011
Solution & Interoperability Test Lab Application Notes
©2011 Avaya Inc. All Rights Reserved.
36 of 39
CM601ASBC_QwSIP
status trunk 1/1
Page
3 of
3
SRC PORT TO DEST PORT TALKPATH
src port: T00001
T00001:TX:205.3.3.208:23512/g711u/20ms
S00010:RX:205.3.3.235:7306/g711u/20ms
Verification Steps:
1. Verify that entity links from the Session Border Controller (205.168.000.000) to Qwest
SIP Trunk (67.148.000.000) are up and communicating with SIP OPTION messages
and 200 OK responses.
2. Verify that the signaling group / trunk group between the Communication Manager
and Session Border Controller are up by using status signaling group # and status
trunk-group #.
3. Verify that endpoints at the enterprise site can place calls to the PSTN and that the call
remains active for more than 35 seconds. This time period is included to verify that
proper routing of the SIP messaging has satisfied SIP protocol timers.
4. Verify that endpoints at the enterprise site can receive calls from the PSTN and that the
call can remain active for more than 35 seconds.
5. Verify that the user on the PSTN can end an active call by hanging up.
6. Verify that an endpoint at the enterprise site can end an active call by hanging up.
Troubleshooting:
1. Communication Manager:
 list trace station <extension number> - Traces calls to and from a specific
station.
 list trace tac <trunk access code number> - Traces calls over a specific trunk
group.
 status station <extension number> - Displays signaling and media information
for an active call on a specific station.
 status trunk <trunk access code number> - Displays trunk group information.
 status trunk <trunk access code number/channel number> - Displays signaling
and media information for an active trunk channel.
2. Session Border Controller
 Virtual Server Platform status page
 Session Border Controller initial status page
 Call logs from the ladder diagrams
9. Conclusion
These Application Notes describe the configuration necessary to connect Avaya Aura®
Communication Manager and Avaya Aura® Session Border Controller to Qwest SIP
Trunking. Qwest SIP Trunking passed compliance testing. Please refer to Section 2.2 for any
observations or workarounds.
MEO; Reviewed:
SPOC 6/23/2011
Solution & Interoperability Test Lab Application Notes
©2011 Avaya Inc. All Rights Reserved.
37 of 39
CM601ASBC_QwSIP
10. References
This section references the documentation relevant to these Application Notes. Additional
Avaya product documentation is available at http://support.avaya.com.
[1] Administering Avaya Aura® Communication Manager, May 2011, Document Number 03603628.
[2] Avaya Aura® Communication Manager Feature Description and Implementation, August
2010, Document Number 555-245-205.
[3] Installing Avaya Aura® SBC System Administration Guide, 2010.
[4] Administering Avaya Aura® Session Services Configuration Guide, 2010.
[5] Avaya 1600 Series IP Deskphones Administrator Guide Release 1.2.x, February 2010,
Document Number 16-601443.
[6] 4600 Series IP Telephone LAN Administrator Guide, October 2007, Document Number
555-233-507.
[7] Avaya one-X® Deskphone H323 Administrator Guide Release 6.1, May 2011, Document
Number 16-300698.
[8] Avaya one-X® Communicator Getting Started, November 2009.
[9] RFC 3261 SIP: Session Initiation Protocol, http://www.ietf.org/
[10] RFC 2833 RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals,
http://www.ietf.org/
[11] RFC 4244, An Extension to the Session Initiation Protocol (SIP) for Request History
Information, http://www.ietf.org/
MEO; Reviewed:
SPOC 6/23/2011
Solution & Interoperability Test Lab Application Notes
©2011 Avaya Inc. All Rights Reserved.
38 of 39
CM601ASBC_QwSIP
©2011
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.
MEO; Reviewed:
SPOC 6/23/2011
Solution & Interoperability Test Lab Application Notes
©2011 Avaya Inc. All Rights Reserved.
39 of 39
CM601ASBC_QwSIP
Was this manual useful for you? yes no
Thank you for your participation!

* Your assessment is very important for improving the work of artificial intelligence, which forms the content of this project

Download PDF

advertising