Application Notes for Configuring Avaya IP Office 8.1 with Completel

Application Notes for Configuring Avaya IP Office 8.1 with Completel
Avaya Solution & Interoperability Test Lab
Application Notes for Configuring Avaya IP Office 8.1 with
Completel UCM Service – Issue 1.0
Abstract
These Application Notes describe the procedures for configuring Session Initiation Protocol
(SIP) trunking between Completel UCM Service and Avaya IP Office.
The Completel UCM Service provides PSTN access via a SIP trunk connected to the
Completel Voice Over Internet Protocol (VoIP) network as an alternative to legacy Analogue
or digital trunks. Completel are 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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1. Introduction
These Application Notes describe the procedures for configuring Session Initiation Protocol
(SIP) trunking between Completel UCM Service and Avaya IP Office. Completel UCM provides
PSTN access via a SIP trunk connected to the Completel network as an alternative to legacy
Analogue or Digital trunks. This approach generally results in lower cost for customers.
2. General Test Approach and Test Results
The general test approach was to configure a simulated enterprise site using Avaya IP Office to
connect to Completel UCM. 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.
2.1. Interoperability Compliance Testing
Avaya IP Office was connected to Completel UCM. To verify SIP trunking interoperability the
following features and functionality were exercised during the interoperability compliance test:
• Incoming PSTN calls to various phone types including H.323, Digital and Analogue
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 including H.323, Digital, and Analogue
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 an IP Office Softphone client
• Various call types including: local, long distance, international, toll free (outbound)and
directory assistance
• Codecs G.711A and G.729A
• Caller ID presentation and Caller ID restriction
• DTMF transmission using RFC 2833
• Voicemail navigation for inbound and outbound calls
• User features such as hold and resume, transfer, and conference
• Off-net call forwarding and twinning
Note: T.38 fax is normally tested but this is not currently functioning on IP Office due to the
codec change from the network being in an UPDATE message rather than a re-INVITE. Fax will
only work using G.711.
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2.2. Test Results
Interoperability testing of the sample configuration was completed with successful results for
Completel UCM with the following observations:
• When incoming test calls were not answered, it was five minutes before IP Office sent a
“503 Service unavailable”. This can be reduced depending on customer requirements.
• When outgoing test calls were not answered, it was only one minute before the network
sent a “BYE”. This is shorter than is usually observed.
• Test calls to busy destinations were answered and an announcement played. The more
usual failure for a busy call is 486 “Busy Here” which prevents any unwanted charging of
the caller
• DTMF test calls did not function correctly to all destinations. These were reliable only to
test phones at the Service Provider’s site. The advice from the Service provider is that
RFC 2833 is not implemented on all interconnects.
• Toll Free access was not available for incoming calls and was not tested.
• T.38 was not correctly negotiated on outbound fax test calls, these only worked when
G.711 was the codec selected on call set-up. The fault is due to the incorrect handling of
the SIP UPDATE sent by the network to change the codec to T.38. This requires a fix on
IP Office that is not due for implementation in the short term.
2.3. Support
For technical support on Completel products please visit the website at www.Completel.net or
contact an authorized Completel representative.
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3. Reference Configuration
Figure 1 illustrates the test configuration. The test configuration shows an enterprise site
connected to Completel UCM Service. Located at the enterprise site is an Avaya IP Office 500
v2. Endpoints include an Avaya 1600 Series IP Telephone (with H.323 firmware), an Avaya
9600 Series IP Telephone (with SIP firmware), an Avaya 2420 Digital Telephone, Avaya
Analogue Telephone and fax machine. The site also has a Windows XP PC running Avaya IP
Office Manager to configure the Avaya IP Office as well as an IP Office Softphone client for
mobility testing. For security purposes, any public IP addresses or PSTN routable phone
numbers used in the compliance test are not shown in these Application Notes. Instead, public IP
addresses have been changed to a private format and all phone numbers have been obscured
beyond the city code.
Figure 1: Completel UCM to Avaya IP Office Topology
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4. Equipment and Software Validated
The following equipment and software were used for the sample configuration provided:
Equipment/Software
Avaya
Avaya IP Office 500 V2
Avaya 1140e SIP Telephone
Avaya 1603 Phone (H.323)
Avaya 1608 Phone (H.323)
Avaya 2420 Digital Phone
Avaya 98390 Analogue Phone
Avaya Softphone
Completel
ACME Net-Net 4500 SBC
Cirpack Softswitch
Release/Version
Avaya IP Office R8.1(67)
04.03.09.00
1.3100
1.3100
N/A
N/A
3.2.3.48 6700
SCX6.2.0 MR-5 GA (Build 777)
v4.2J14
5. Configure Avaya IP Office
This section describes the Avaya IP Office configuration to support connectivity to Completel
UCM. Avaya IP Office is configured through the Avaya IP Office Manager PC application.
From a PC running the Avaya IP Office Manager application, select Start Programs IP
Office Manager (not shown) to launch the application. Navigate to File Open
Configuration (not shown), select the proper Avaya IP Office system from the pop-up window,
and log in with the appropriate credentials. A management window will appear similar to the one
in the next section. All the Avaya IP Office configurable components are shown in the left pane
known as the Navigation Pane. The pane on the right is the Details Pane. These panes will be
referenced throughout the Avaya IP Office configuration. All licensing and feature configuration
that is not directly related to the interface with the Service Provider (such as twinning) is
assumed to already be in place.
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5.1. Verify System Capacity
Navigate to License SIP Trunk Channels in the Navigation Pane. In the Details Pane verify
that the License Status is Valid and that the number of Instances is sufficient to support the
number of SIP trunk channels provisioned by Completel.
5.2. LAN2 Settings
In the sample configuration, the LAN2 port was used to connect the Avaya IP Office to the
external internet. To access the LAN2 settings, first navigate to System GSSCP_IPO2 in the
Navigation Pane where GSSCP_IPO2 is the name of the IP Office. Navigate to LAN2 LAN
Settings tab in the Details Pane. The IP Address and IP Mask fields are the public interface of
the IP Office, Primary Trans. IP Address is the next hop, usually the default gateway address.
All other parameters should be set according to customer requirements. On completion, click the
OK button (not shown).
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On the VoIP tab in the Details Pane, check the SIP Trunks Enable box to enable the
configuration of SIP trunks. The IP Office Softphone uses SIP. If Softphone along with any other
SIP endpoint is to be used, the SIP Registrar Enable box must also be checked. The RTP Port
Number Range can be customized to a specific range of receive ports for the RTP media. Based
on this setting, Avaya IP Office would request RTP media be sent to a UDP port in the
configurable range for calls using LAN2.
Avaya IP Office can also be configured to mark the Differentiated Services Code Point (DSCP)
in the IP Header with specific values to support Quality of Services policies for both signalling
and media. The DSCP field is the value used for media and the SIG DSCP is the value used for
signalling. The specific values used for the compliance test are shown in the example below. All
other parameters should be set according to customer requirements. On completion, click the OK
button (not shown).
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On the Network Topology tab in the Details Pane enter the information required if NAT is to be
used. During test, NAT was not required and there was no requirement for a STUN server. To
disable this facility, 0.0.0.0 is entered in the STUN Server IP Address and Public IP Address
fields. If NAT is to be used, this tab can also be used to set the Binding Refresh Time for the
periodic sending of OPTIONS, 300 seconds or 5 minutes is shown as an example. Alternatively,
the periodic sending of OPTIONS can be specified in the User settings, see Section 5.10 for
more details. On completion, click the OK button (not shown).
5.3. System Telephony Settings
Navigate to the Telephony Telephony tab on the Details Pane. Choose the Companding
Law typical for the enterprise location. For Europe, ALAW is used. Uncheck the Inhibit OffSwitch Forward/Transfer box to allow call forwarding and call transfer to the PSTN via the
Service Provider across the SIP trunk. On completion, click the OK button (not shown).
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5.4. System Twinning Settings
Navigate to the Twinning tab, check the box labeled Send original calling party information
for Mobile Twinning. With this setting, Avaya IP Office will send the original calling party
number to the twinned phone in the SIP From header (not the associated desk phone number) for
calls that originate from an internal extension. For inbound PSTN calls to a twinned enabled
phone, Avaya IP Office will continue to send the associated host phone number in the SIP From
header (used for the caller display). This setting only affects twinning and does not impact the
messaging of other redirected calls such as forwarded calls. If this box is checked, it will also
override any setting of the Send Caller ID parameter on the SIP line (Section 5.6). On
completion, click the OK button (not shown).
5.5. Codec Settings
Navigate to the Codecs tab on the Details Pane. Check the Available Codecs boxes as required.
Note that G.711 ULAW 64K and G.711 ALAW 64K are greyed out and always available. Once
available codecs are selected, they can be used or unused by using the horizontal arrows as
required. Note that in test, G.729(a) 8K CS-ACELP and G.711 ULAW 64K were used. The
order of priority can be changed using the vertical arrows. On completion, click the OK button
(not shown).
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5.6. Administer SIP Line
A SIP line is needed to establish the SIP connection between Avaya IP Office and the Completel
UCM service. To create a SIP line, begin by navigating to Line in the Navigation Pane. Rightclick and select New
SIP Line (not shown). On the SIP Line tab in the Details Pane, configure
the parameters below to connect to the SIP Trunking service.
• Set ITSP Domain Name field to the domain name used by Completel. In test this was
trusted.voip.completel.fr.
• Set Send Caller ID to None as it is only required if the box labeled Send original
calling party information for Mobile Twinning is unchecked in Section 5.4.
• Ensure the In Service box is checked.
• Default values may be used for all other parameters.
On completion, click the OK button (not shown).
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Select the Transport tab and set the following:
• Set ITSP Proxy Address to the IP address of the Completel SIP proxy
• Set Use Network Topology Info to None if NAT is not to be used as was the case during
test
• Set Layer 4 Protocol to UDP
• Set Send Port and Listen Port to 5060
On completion, click the OK button (not shown).
After the SIP line parameters are defined, the SIP URIs that Avaya IP Office will accept on this
line must be created. To create a SIP URI entry, first select the SIP URI tab. Click the Add
button and the New Channel area will appear at the bottom of the pane.
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For the compliance test, a single SIP URI entry was created that matched any number assigned to
an Avaya IP Office user. The entry was created with the parameters shown below.
• Set Local URI to *, this setting allows all calls on this line. If the number set in the SIP
tab of the User as shown in Section 5.8.
• Set Contact, Display Name and PAI to Use Internal Data
• For Registration, only 0: <None> is available at this point. Registration details are
defined under the SIP Credentials tab and, once defined, are available in this drop down
menu.
• Associate this line with an incoming line group by entering a line group number in the
Incoming Group field. This line group number will be used in defining incoming call
routes for this line. Similarly, associate the line to an outgoing line group using the
Outgoing Group field. The outgoing line group number is used in defining short codes
for routing outbound traffic to this line. For the compliance test, a new incoming and
outgoing group 18 was defined that was associated to a single line (line 18).
• Set Max Calls per Channel to the number of simultaneous SIP calls that are allowed
using this SIP URI pattern.
On completion, click the OK button.
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Select the VoIP tab to set the Voice over Internet Protocol parameters of the SIP line. Set the
parameters as shown below:
• Select Custom in the Codec Selection drop down menu to specify the preferred codecs
• Highlight codecs in the Unused box that are to be used on this line and click on the right
arrows to move them to the Selected box
• Highlight codecs in the Selected box that are not to be used and click on the left arrows
to move them to the Unused box
• Highlight codecs in the Selected box and use the up and down arrows to change the
priority order of the offered codecs, for testing with Completel this was G.729(a) 8K CSACELP followed by G.711 ALAW 64K
• Select T38 in the Fax Transport Support drop down menu to allow T.38 fax operation
• Select RFC2833 in the DTMF Support drop down menu. This directs Avaya IP Office
to send DTMF tones using RTP events messages as defined in RFC2833
• Uncheck the VoIP Silence Suppression box
• Check the Re-invite Supported box, to allow for codec re-negotiation in cases where the
target of the incoming call or transfer does not support the codec originally negotiated on
the trunk
• Check the PRACK/100rel Supported box to allow for reliable responses to provisional
call set-up messages such as 183 Session progress and 180 Ringing.
• Default values may be used for all other parameters
• On completion, click the OK button (not shown).
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Select the T.38 Fax tab to set the T.38 parameters for the line. Un-check the Use Default Values
box (not shown) and select 0 from the T38 Fax Version drop down menu. Set the Max Bit Rate
(bps) to 14400. All other field may retain their default values. On completion, click the OK
button (not shown).
Select the SIP Credentials tab to configure the authentication parameters for Completel UCM.
The authentication takes placed on the SIP registration. Define the login details in the User
name and Authentication Name fields. Define the domain in the Contact field and the
password in the Password field. Check the Registration required box
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Note: It is advisable at this stage to save the configuration as described in Section 5.11 to make
the Line Group ID defined in Section 5.6 available.
5.7. Short Codes
Define a short code to route outbound traffic to the SIP line. To create a short code, right-click
Short Code in the Navigation Pane and select New. On the Short Code tab in the Details Pane,
configure the parameters as shown below.
• In the Code field, enter the dial string which will trigger this short code, followed by a
semi-colon
• The example shows 90N; which will be invoked when the user dials 9 followed by a
public number.
• Set Feature to Dial. This is the action that the short code will perform.
• Set Telephone Number to 0N which will insert the public number with leading 0 into
the Request URI and To headers in the outgoing SIP INVITE message
• Set the Line Group Id to the outgoing line group number defined on the SIP URI tab on
the SIP Line in Section 5.6
On completion, click the OK button (not shown).
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Short codes are also used for routing of national calls and Operator calls. An example for
Operator calls, in this case directory enquiries, is shown below.
• The example of an Operator call shows 118 for the first three digits with no leading digit
9. This allows 118 numbers to be dialled without dialling 9 for an outside line
• Set Telephone Number to 118 and any subsequent dialled digits. In test, a 6 digit
number was dialed for directory enquiries.
• Set other parameters as shown in the first example.
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5.8. User
Configure the SIP parameters for each user that will be placing and receiving calls via the SIP
line defined in Section 5.6. To configure these settings, first navigate to User in the Navigation
Pane. Select the User tab if any changes are required. The example below shows the changes
required to use IP Office Softphone which was used in test. Softphone replaced Phone Manager
at IP Office 8.0.
• Change the Name of the User if required, this will be used for login to the IP Office
Softphone
• Select Teleworker User from the Profile drop down menu
• Check the Enable Softphone box
IP Office Softphone uses SIP for signalling and hence required setting of the SIP Registrar
Enable as described in Section 5.2. Call forwarding and transfer make use of the SIP REFER
message. To handle SIP REFER on IP Office, the Call waiting function is used.
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To turn on Call Waiting, navigate to Telephony
Call Settings. Check the Call Waiting On
box.
Next Select the SIP tab in the Details Pane. To reach the SIP tab click the right arrow on the
right hand side of the Details Pane until it becomes visible. The values entered for the SIP Name
and Contact fields are used as the user part of the SIP URI in the From header for outgoing SIP
trunk calls. These allow matching of the SIP URI for incoming calls without having to enter this
number as an explicit SIP URI for the SIP line (Section 5.6). As such, these fields should be set
to one of the DDI numbers assigned to the enterprise from Completel.
In the example below, one of the DDI numbers in the test range is used, though only the first five
and final digits are shown. The SIP Display Name (Alias) parameter can optionally be
configured with a descriptive name. On completion, click the OK button (not shown).
Note: The Contact field must be in E.164 format for the caller ID on the called phone to display
properly. Also note that the Anonymous box can be checked if The Calling Line Identity is to be
Restricted (CLIR).
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5.9. Incoming Call Routing
An incoming call route maps an inbound DDI number on a specific line to an internal extension.
To create an incoming call route, right-click Incoming Call Routes in the Navigation Pane and
select New (not shown). On the Standard tab of the Details Pane, enter the parameters as shown
below:
• Set the Bearer Capacity to Any Voice
• Set the Line Group Id to the incoming line group of the SIP line defined in Section 5.6
• Set the Incoming Number to the incoming number that this route should match on.
Matching is right to left
• Default values can be used for all other fields
Note: A number of digits of the DDI have been obscured. Number format is national.
On the Destinations tab, select the destination extension from the pull-down menu of the
Destination field. On completion, click the OK button (not shown). In this example, incoming
calls to the test DDI number on line 18 are routed to extension 89022.
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5.10. SIP Options
Avaya IP Office sends SIP OPTIONS messages periodically to determine if the SIP connection
is active. The rate at which the messages are sent is determined by the lower value of the
Binding Refresh Time (in seconds) set on the Network Topology tab in Section 5.2 and the
SIP_OPTIONS_PERIOD parameter (in minutes) that can be set on the Source Number tab of
the NoUser user. During test, the Network Topology information was not used as there was no
requirement for NAT and a STUN server. As a result, the SIP_OPTIONS_PERIOD was used
to define the rate at which OPTIONS messages were sent.
To configure the SIP_OPTIONS_PERIOD parameter, navigate to User NoUser in the
Navigation Pane. Select the Source Numbers tab in the Details Pane. Click the Add button
At the bottom of the subsequent Details Pane, the Source Number field will appear. Enter
SIP_OPTIONS_PERIOD=X, where X is the desired value in minutes. Click OK.
The SIP_OPTIONS_PERIOD parameter will appear in the list of Source Numbers as shown
below. For the compliance test, an OPTIONS period of 2 minutes was used.
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5.11. Save Configuration
Navigate to File Save Configuration (not shown) in the menu bar at the top of the screen to
save the configuration performed in the preceding sections.
6. Completel UCM Configuration
Completel is responsible for the configuration of the UCM service. The customer will need to
provide the public IP address used to reach the Avaya IP Office at the enterprise. Completel will
provide the customer the necessary information to configure the SIP connection to the SIP
Trunking service including:
• IP address of UCM SIP proxy
• Network SIP Domain
• Supported codecs
• DDI numbers
• All IP addresses and port numbers used for signalling or media that will need access to
the enterprise network through any security devices.
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7. Verification Steps
This section includes steps that can be used to verify that the configuration has been done
correctly.
7.1. SIP Trunk status
The status of the SIP trunk can be verified by opening the System Status application. A Windows
7 Laptop PC was used for testing and the application was opened by pressing the Start button
and selecting All Programs
IP Office System Status.
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Log in to IP Office System Status at the prompt using the Control Unit IP Address for the IP
office. The User Name and Password are the same as those used for IP Office Manager.
From the left hand menu expand Trunks and choose the SIP trunk (18 in this instance). The
status window will show the status as being idle and time in state if the Trunk is operational. IP
address has been changed.
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8. Conclusion
The Completel UCM service passed compliance testing. Interoperability testing of the sample
configuration was completed with successful results for Completel UCM. Refer to Section 2.2
for test observations.
9. Additional References
Product documentation for Avaya products may be found at http://support.avaya.com.
[1] IP Office 8.1 KnowledgeBase Technical Documentation CD, 17th December 2012.
[2] IP Office 8.1 Installing IP500/IP500 V2, Document number15-601042, 22nd August 2013.
[3] IP Office R8.1 FP1 Manager 10.1, Document number15-601011, 30th August 2013.
[4] IP Office 8.1 Using System Status, Document number15-601758, 24th May 2013
[5] IP Office Softphone Installation, Document number 100164693, 12th June 2012
[6] IP Office SIP Extension Installation, 3rd October 2011
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©2013 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.
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