Configuring SIP IP Telephony Using Avaya Aura TM SIP

Configuring SIP IP Telephony Using Avaya Aura TM SIP
Avaya Solution & Interoperability Test Lab
Configuring SIP IP Telephony Using Avaya Aura TM SIP
Enablement Services, Avaya Aura TM Communication
Manager, and Cisco ATA 186 Analog Telephone Adapters Issue 1.0
Abstract
These Application Notes describe the configuration steps required to connect Cisco ATA 186
Analog Telephone Adapters to a SIP infrastructure consisting of an Avaya Aura TM SIP
Enablement Services (SES) and Avaya Aura TM Communication Manager. Cisco ATA 186 is
directly registered to Avaya SIP Enablement Services (SES) as SIP endpoint with Avaya
Analog phones connected to Cisco ATA 186. For registration Cisco ATA 186 uses UDP.
RJ; Reviewed:
SPOC 03/08/2010
Solution & Interoperability Test Lab Application Notes
©2010 Avaya Inc. All Rights Reserved.
1 of 22
ATA186_SES
1. Introduction
With the introduction of the SIP protocol standard that supports telephony as well as a wide
range of other communication modes, there is a much broader range of SIP telephones and
gateways available to customers. There will be sales opportunities involving customers who wish
to purchase the Avaya SIP offer, but already own analog telephones and analog gateways other
than those offered by Avaya. Customers may be interested in replacing their existing telephony
infrastructure (e.g., Cisco Unified Communications Manager) with Avaya servers, but wish to reuse the existing non-Avaya gateways. In addition, the Off-PBX Station (OPTIM) feature set can
be extended from Avaya AuraTM Communication Manager to analog telephones connected to
these gateways, providing enhanced calling features in advance of SIP protocol definitions and
implementation by gateway manufacturers. These Application Notes describe the configuration
steps for using Cisco ATA 186 Analog Telephone Adapters with Avaya AuraTM SIP Enablement
Services and Avaya AuraTM Communication Manager. Only those configuration steps pertinent
to interoperability of Cisco and Avaya equipment are covered. General administration
information can be found in the product documentation as well as the specific references listed in
References section.
RJ; Reviewed:
SPOC 03/08/2010
Solution & Interoperability Test Lab Application Notes
©2010 Avaya Inc. All Rights Reserved.
2 of 22
ATA186_SES
1.1. Configuration
The configuration used as an example in these Application Notes is shown in Figure 1. The
diagram indicates logical signaling connections. With the exception of the Avaya analog
telephones and fax machines, all components are physically connected to an Avaya G650 Media
gateway and are administered as a single subnet. Each Cisco ATA is configured to register to
SIP Enablement Services. The ATA supports two analog ports, representing each port as a
separate SIP telephone. An analog telephone and a fax machine are connected to the ports of
each ATA. Each ATA port is administered as an OPTIM station on Avaya Aura™ for Midsize
Enterprises running on an Avaya S8800 Server. The Avaya Modular Messaging application
resides on the S8730 Media Server and is used to support voice messaging. The PC supports a
TFTP server as well as a web browser for administration of the network components.
Figure 1: Avaya SIP Test Configuration with Cisco ATA 186 Analog Telephone Adapters
RJ; Reviewed:
SPOC 03/08/2010
Solution & Interoperability Test Lab Application Notes
©2010 Avaya Inc. All Rights Reserved.
3 of 22
ATA186_SES
2. Equipment and Software Validated
The following equipment and software/firmware were used for the sample configuration.
Network Component
Avaya 9600-Series Telephones
Avaya 9600-Series Telephones
Avaya Aura™ Solution for Midsize
Enterprises S8800 server
Avaya S8720 Server
Avaya G650 Media Gateway
IPSI TN2312BP
MedPro TN2602AP
CLAN TN799DP
ANALOG TN793CP
Avaya Modular Messaging on Avaya S8730
Messaging Servers (MAS and MSS)
Avaya AuraTM Communication Manager
Feature Server
Avaya AuraTM System Manager Server S8510
Avaya AuraTM Session Manager Server S8510
Cisco ATA 186 Analog Telephone Adapter
Software Version
3.002 (H.323)
2.5.0 (SIP)
5.2.1.2.5
Avaya Aura™ Communication
Manager 5.2.1
(S8720-015-02.1.016.4 with update
17774)
HW15 FW049
HW08 FW049
HW01 FW034
HW09 FW008
Avaya Modular Messaging 5.2
(9.2.150)
Avaya Aura™ Communication
Manager 5.2.1
(S8720-015-02.1.016.4 with update
17774)
Avaya AuraTM System Manager
5.2.0.1- SP0
Avaya AuraTM Session Manager
5.2.0.1- SP0
3.2.1 (SIP)
3. Administer Avaya AuraTM SIP Enablement Services
The following steps describe configuration of SIP Enablement Services for use with the Cisco
186 ATA phone and fax connected to ATA. SES is configured via an Internet browser using the
administrator web interface. It is assumed that SES software and the license file have already
been installed on the server. Access the SES administration web interface by entering
http://<SES-ip-addr>/admin as the URL in an Internet browser.

Administer SIP OPTIM Users
RJ; Reviewed:
SPOC 03/08/2010
Solution & Interoperability Test Lab Application Notes
©2010 Avaya Inc. All Rights Reserved.
4 of 22
ATA186_SES
3.1. Administer SIP OPTIM Users
Log in with appropriate credentials and then select the Launch SES Administration Interface
link from the main page.
RJ; Reviewed:
SPOC 03/08/2010
Solution & Interoperability Test Lab Application Notes
©2010 Avaya Inc. All Rights Reserved.
5 of 22
ATA186_SES
From the main page, click on Users as shown in the sample configuration below.
RJ; Reviewed:
SPOC 03/08/2010
Solution & Interoperability Test Lab Application Notes
©2010 Avaya Inc. All Rights Reserved.
6 of 22
ATA186_SES
On the left panel expand Users. From Users click Add. Enter the required details as shown in
the sample configuration below and click Add. Repeat the same steps for other SIP OPTIM
users.
 Primary Handle:
40040 User Extension
 User ID:
40040 User Extension
 Password:
Extension Password
 Host:
135.64.186.89, i.e. SES IP Address
 Add Communication Manager Extension:
Tick the check box. This will prompt administrator to create
extension as shown in the next page.
RJ; Reviewed:
SPOC 03/08/2010
Solution & Interoperability Test Lab Application Notes
©2010 Avaya Inc. All Rights Reserved.
7 of 22
ATA186_SES
Click Continue (not shown). The Add Communication Manager Extension screen will be
displayed. Enter the required details as shown in the sample configuration. Click Add and
Continue.
RJ; Reviewed:
SPOC 03/08/2010
Solution & Interoperability Test Lab Application Notes
©2010 Avaya Inc. All Rights Reserved.
8 of 22
ATA186_SES
4. Configure Avaya AuraTM for Midsize Enterprises
Communication Manager
This section highlights administering the OPTIM features on Communication Manager for Cisco
ATA 186.
4.1. Verify OPTIM Capacity
Use the display system-parameters customer-options command to verify that Maximum OffPBX Telephones – OPTIM has been set to the value that has been licensed, and that this value
will accommodate addition of the ATA-supported analog ports. Note that there are two ports on
each Cisco ATA 186.
display system-parameters customer-options
OPTIONAL FEATURES
G3 Version: V15
Location: 2
Platform: 25
1 of
11
Software Package: Standard
RFA System ID (SID): 1
RFA Module ID (MID): 1
Platform Maximum Ports:
Maximum Stations:
Maximum XMOBILE Stations:
Maximum Off-PBX Telephones - EC500:
Maximum Off-PBX Telephones - OPTIM:
Maximum Off-PBX Telephones - PBFMC:
RJ; Reviewed:
SPOC 03/08/2010
Page
44000
2400
2400
2400
2400
2400
USED
107
15
0
2
10
2
Solution & Interoperability Test Lab Application Notes
©2010 Avaya Inc. All Rights Reserved.
9 of 22
ATA186_SES
4.2. Administer the Dial Plan
Use the change dialplan analysis command to define the dial plan used in the system. This
includes all telephone extensions.
change dialplan analysis
Page
DIAL PLAN ANALYSIS TABLE
Location: all
Dialed
String
0
1
2
209
3
4
5
6
7
8
9
*
#
Total
Length
1
5
5
5
5
5
5
5
5
5
1
3
3
Call
Type
fac
ext
aar
ext
aar
ext
ext
ext
aar
aar
fac
dac
dac
Dialed
String
Total Call
Length Type
1 of
12
Percent Full:
Dialed
String
0
Total Call
Length Type
4.3. Administer Coverage Path
Configure the coverage path to be used for the voice messaging hunt group, which is group h1 in
the sample configuration. The default values shown for Busy?, Don’t Answer?, and
DND/SAC/Goto Cover? can be used for the Coverage Criteria. In this case, the Number of
Rings before the call goes to voice messaging has been extended from the default of 2 to 4 rings.
change coverage path 1
Page
1 of
1
COVERAGE PATH
Coverage Path Number: 1
Cvg Enabled for VDN Route-To Party? n
Next Path Number:
Hunt after Coverage? n
Linkage
COVERAGE CRITERIA
Station/Group Status
Active?
Busy?
Don't Answer?
All?
DND/SAC/Goto Cover?
Holiday Coverage?
Inside Call
n
y
y
n
y
n
Outside Call
n
y
y
n
y
n
Number of Rings: 4
COVERAGE POINTS
Terminate to Coverage Pts. with Bridged Appearances? n
Point1: h1
Rng: 2 Point2:
Point3:
Point4:
Point5:
Point6:
RJ; Reviewed:
SPOC 03/08/2010
Solution & Interoperability Test Lab Application Notes
©2010 Avaya Inc. All Rights Reserved.
10 of 22
ATA186_SES
4.4. Administer Stations
Use the add station command to add a station for each ATA port to be supported. Assign the
same extension as the Communication Manager extension administered in SIP Enablement
Services. Use 9620 for the Station, and be sure to include the Coverage Path for voice
messaging if applicable. The Name field is optional and is shown on the display of Avaya nonSIP telephones when receiving calls from this station. Use defaults for the other fields on Page 1.
add station 40040
Page
1 of
5
STATION
Extension:
Type:
Port:
Name:
40040
9620
S00017
Cisco ATA1
Lock Messages? n
Security Code:
Coverage Path 1: 1
Coverage Path 2:
Hunt-to Station:
BCC:
TN:
COR:
COS:
0
1
1
1
STATION OPTIONS
Loss Group: 19
Speakerphone:
Display Language:
Survivable GK Node Name:
Survivable COR:
Survivable Trunk Dest?
2-way
english
internal
y
Time of Day Lock Table:
Personalized Ringing Pattern: 1
Message Lamp Ext: 40040
Mute Button Enabled? y
Media Complex Ext:
IP SoftPhone? n
IP Video? n
Customizable Labels? y
RJ; Reviewed:
SPOC 03/08/2010
Solution & Interoperability Test Lab Application Notes
©2010 Avaya Inc. All Rights Reserved.
11 of 22
ATA186_SES
On Page 2, if this telephone will have a bridged appearance for another telephone (see Page 3
for this station), then Bridged Call Alerting should be set to y, so that this telephone will ring
when the other telephone is called. Note that no other operational behaviors of the bridged
appearance feature apply to SIP telephones (e.g., off-hook indication, bridge-on, etc.). By
default, the last call appearance is reserved for outgoing calls from the telephone. If it is
desirable to allow an incoming call to use the last available call appearance when all others are
occupied, set the Restrict Last Appearance field to n. In this mode, all call appearances are
available for making or receiving calls. Enter the “sip-adjunct” administered for this system in
MWI Served User Type.
change station 40040
Page
2 of
5
STATION
FEATURE OPTIONS
LWC Reception:
LWC Activation?
LWC Log External Calls?
CDR Privacy?
Redirect Notification?
Per Button Ring Control?
Bridged Call Alerting?
Active Station Ringing:
spe
y
n
n
y
n
y
single
Auto Select Any Idle Appearance?
Coverage Msg Retrieval?
Auto Answer:
Data Restriction?
Idle Appearance Preference?
Bridged Idle Line Preference?
Restrict Last Appearance?
n
y
none
n
n
n
n
EMU Login Allowed? n
n
Per Station CPN - Send Calling Number? y
as-needed
EC500 State: enabled
enhanced
Audible Message Waiting? n
sip-adjunct
Display Client Redirection? n
Select Last Used Appearance? n
Coverage After Forwarding? s
Multimedia Early Answer? n
Direct IP-IP Audio Connections? y
Emergency Location Ext: 40040
Always Use? n IP Audio Hairpinning? n
H.320 Conversion?
Service Link Mode:
Multimedia Mode:
MWI Served User Type:
RJ; Reviewed:
SPOC 03/08/2010
Solution & Interoperability Test Lab Application Notes
©2010 Avaya Inc. All Rights Reserved.
12 of 22
ATA186_SES
On Page 4, under the heading BUTTON ASSIGNMENTS, fill in the number of call
appearances (call-appr buttons) that are to be supported for the telephone. For the ATA 186 the
recommended number is 2. Configure brdg-appr on this extension for 40001. In the sample
configuration, 40001 is another SIP endpoint.
change station 40040
Page
4 of
5
STATION
SITE DATA
Room:
Jack:
Cable:
Floor:
Building:
Headset?
Speaker?
Mounting:
Cord Length:
Set Color:
ABBREVIATED DIALING
List1:
BUTTON ASSIGNMENTS
1: call-appr
2: call-appr
3: brdg-appr B:1
List2:
E:40001
n
n
d
0
List3:
4:
5:
6:
voice-mail Number:
4.5. Administer the OPTIM
Use the change off-pbx-telephone station-mapping command to map the Communication
Manager extension (40040) to the same SIP Enablement Services extension. Enter the field
values shown. For the sample configuration, the Trunk Selection value indicates the SIP trunk
group. The Configuration Set value can reference a set that has the default settings in
Communication Manager.
change off-pbx-telephone station-mapping 40040
STATIONS WITH OFF-PBX TELEPHONE INTEGRATION
Station
Extension
40040
RJ; Reviewed:
SPOC 03/08/2010
Application Dial
CC
Prefix
OPTIM
-
Phone Number
40040
Trunk
Selection
3
Solution & Interoperability Test Lab Application Notes
©2010 Avaya Inc. All Rights Reserved.
Page
1 of
Config
Set
1
3
Dual
Mode
13 of 22
ATA186_SES
On Page 2, change the Call Limit to match the number of call-appr entries in the add station
form. Also make sure that Mapping Mode is set to both (the default value for a newly added
station).
change off-pbx-telephone station-mapping 40040
STATIONS WITH OFF-PBX TELEPHONE INTEGRATION
Station
Extension
40040
Appl
Name
OPTIM
Call
Limit
2
Mapping
Mode
both
Calls
Allowed
all
Page
Bridged
Calls
both
2 of
3
Location
5. Configure the Cisco ATA 186 Analog Telephone Adapter
5.1. Registration and Basic Calling
Cisco ATAs can be configured using three methods:
1
Configuration file downloaded from a TFTP server specified via DHCP at boot time.
Two such files can be installed on the TFTP server. At boot time, the ATA will attempt
to load a device-specific file of the form ata<MAC-address>.cfg, where
<MACaddress> is the MAC address of the ATA. If this file is not present, the ATA will
attempt to load atadefault.cfg, a default configuration file containing parameter
settings that apply to all ATAs. Note that if the device-specific file is accessible, the
ATA will not attempt to load the default file.
2
Web interface – for configuring a small number of adapters, or to inspect the current
parameter settings on an adapter. Access http://IP_address using a web browser.
3
Manual configuration of the ATA telephone using voice configuration menu, accessible
using the dial pad on an analog telephone connected to port 1. This is typically used to
set the IP network parameters and to reset the gateway to its factory default parameter
values.
RJ; Reviewed:
SPOC 03/08/2010
Solution & Interoperability Test Lab Application Notes
©2010 Avaya Inc. All Rights Reserved.
14 of 22
ATA186_SES
Parameters that are manually changed using the web or voice configuration interfaces will revert
back to the values in the configuration file when the telephone is re-booted, unless the DHCP
and TFTP parameters have been manually changed. For the sample configuration, the IP address
of the telephone and its TFTP server were manually entered at the telephone using the voice
configuration menu. The remaining configuration was done via the configuration file. Excerpts
of the configuration file are shown, including explanatory comments supplied with the product
sample file.
1.
Edit the default or device-specific configuration file, or use the web interface to configure
ATA parameters. ATA software updates include a sample file sip_example.txt,
annotated with explanations for each parameter (not included in these Application Notes).
The screen below shows the relationship between the parameters that must be configured
for the ATA and those administered in SIP Enablement Services. Sample values are
shown for extension 40040 on the ATA 186. Parameter names in the configuration file
are the same as in the web interface. Parameter names ending in 0 or 1 refer to telephone
ports 1 and 2, respectively.
RJ; Reviewed:
SPOC 03/08/2010
Solution & Interoperability Test Lab Application Notes
©2010 Avaya Inc. All Rights Reserved.
15 of 22
ATA186_SES
The following screen summarizes other important network and SIP related parameters that can
be configured in the ATA, and the values used in the sample configuration. The ATA uses the
DialPlan parameter to determine when enough digits have been pressed to complete dialing, so
that the user need not press a send key such as * or # to launch the call. Note that DialPlan has
been specified to support * and #, since they are used in feature access codes.
2.
3.
4.
Continue editing the configuration file, specifying the audio codec and audio modes to be
used for voice and fax. The table below summarizes the combination of values used in
the sample configuration. The low bit rate codec parameter must be assigned the
compressed codec to be used after the preferred codec (G.711 µ-law) in the preference
list used by the ATA. The audio mode flags bit vector (AudioMode) was set to enable
adding the compressed codec, silence suppression, fax CallED (CED) tone detection, and
support for out of band DMTF signaling via RFC 2833 for both ATA ports. See
Appendix A for the complete configuration file used for the ATA 186 in the sample
configuration.
Audio Parameter Definition Parameter Name Example Value
 Audio codec preference (receive) RxCodec 2 (G.711 µ-law)
 Audio codec preference (transmit) TxCodec 2 (G.711 µ-law)
 Low bit rate codec LBRCodec 3 (G.729A)
 Audio mode flags (ports 1 and 2) AudioMode 0x00150015
The completed configuration file must be compiled into a form downloadable by the
ATA via TFTP. Use the Cisco-supplied cfgfmt command line tool in a DOS window of
the PC from which the configuration file will be downloaded. The command should be
of the form: cfgfmt -sip edited_configuration_file compiled_configuration_file Move
the file compiled_configuration_file to the default TFTP directory. As mentioned earlier
RJ; Reviewed:
SPOC 03/08/2010
Solution & Interoperability Test Lab Application Notes
©2010 Avaya Inc. All Rights Reserved.
16 of 22
ATA186_SES
5.
in this section, the name of this file should be either ata<MAC-address>.cfg
(recommended) or atadefault.cfg.
Use the voice configuration menu to assign the initial network parameters. The voice
configuration menu is accessed by connecting an analog telephone to the RJ-11 jack
labeled Phone1 on the back of the ATA. Lift the receiver and press the function button
on top of the ATA as shown below.
A voice prompt will be played, requesting a command. Commands are entered using telephone
key pad sequences, terminated by the # key. The basic parameters shown in the table below
should be set using this method.
IVR Menu Number
Feature
1
IP Address (StaticIP)
2
Default Gateway (StaticRoute)
10
Subnet Mask (StaticNetMask)
20 (0 = Disable, 1 = Enable)
DHCP (Dhcp)
80
Check IP Address
After assigning the above parameters, press # to exit the voice configuration menu. The function
button light should flicker, indicating that the parameters are being loaded and the ATA is
resetting. The configuration file created in the previous step will be downloaded and the ATA
RJ; Reviewed:
SPOC 03/08/2010
Solution & Interoperability Test Lab Application Notes
©2010 Avaya Inc. All Rights Reserved.
17 of 22
ATA186_SES
will register with SIP Enablement Services. Successful registration will be indicated by the
reception of dial tone when lifting the receiver on the analog telephone.
6. Local Calling Features on Cisco ATA 186
The following sections describe how to administer local telephone features supported by the
ATA that are compatible with the Avaya SIP offer. These features are controlled by parameter
values in the configuration file for the ATA. As described in Section 5.1, the procedure is to edit
the configuration text file with the desired parameter values, compile the file, and reset the ATA
so that the compiled configuration file will be downloaded. These Application Notes address
only the parameter values that support the particular feature operations described.
6.1. Call Waiting
Call waiting allows a second call to be answered while a call is in progress. Switch hook flash
can be used to switch between active and held calls. The user can also disconnect the active call
by hanging up, after which the telephone will ring as a reminder that a held call still exists.
Answering the telephone will resume the held call. To configure these call waiting capabilities,
make sure that the following string sequence is included in the CallCmd parameter in Cisco
ATA 186 configuration file.
Kf;EFh;HF
6.2. Fax
The ATA supports pass-through fax mode using G.711 µ-law if the audiomode and
connectmode ATA configuration file parameters are set for pass-through. Both parameters are
represented as bit strings, where the appropriate bits must be set to support fax pass-through.
The bits are numbered 0-31 from right to left. In the sample configuration, audiomode was set
to 0x00150015, where the low order and high order 16 bits apply to analog port 1 and port 2,
respectively. The bits that pertain to fax for port 1 are:
Bit 0 = 1
Enables G.711 silence suppression (VAD).
Bit 2 = 1
Enables Fax CED tone detection and switchover upon detection.
Bit 4 = 1, Bit 5 = 0 DTMF transmission method = out-of-band through negotiation.
Bit 6 = Bit 7 = 0
Disable sending out switch-hook flash.
All 32 bits of the connectmode parameter bit string represent settings that are applied to both
ports. In the sample configuration, this parameter was set to 0x02060400. The applicable bits for
fax pass-through are bit 2 and bits 7-15:
Bit 2 = 0
Uses RTP payload number 126/127 for fax up-speed to G.711µ−law/
G.711A-law.
Bit 7 = 0
Disables fax pass-through redundancy.
Bits {12, 11, 10, 9, 8} = {0, 0, 1, 0, 0}
Sets the offset to NSE payload-type number 96 to 4. Setting the offset to 4
results in the Cisco ATA sending a Named Signaling Event (NSE)
payload-type value of 100 by default.
Bit 13 = 0
Uses G.711µ−law for fax pass-through up-speed.
Bit 14 = Bit 15 = 0 Enables fax pass-through mode using the Cisco proprietary method, if
supported at the far end.
RJ; Reviewed:
SPOC 03/08/2010
Solution & Interoperability Test Lab Application Notes
©2010 Avaya Inc. All Rights Reserved.
18 of 22
ATA186_SES
7. Verification Steps
Features tested using the sample configuration included call bridging, voice mail with messaging
waiting indicator, fax, codecs, and telephony features. The following steps can be used to verify
and/or troubleshoot installations in the field.
1
After rebooting the ATA, use its web interface to verify that the parameters set in the
compiled configuration file have been loaded. Lift the handset on the analog telephones
and verify dial tone, indicating that registration has occurred. If there is no dial tone,
check that the Proxy IP address is correct, and that SIPRegOn is set to 1. The web
interface can be used to change ATA parameters during troubleshooting, but remember
that if a configuration file is normally used for administration, any changes must be
reflected in the source and compiled files.
2
Verify basic feature set administration by lifting the handset and making calls to other
telephones.
3
Call an ATA supported telephone that currently has no voice messages, and leave a
message. Verify that stutter dial tone is heard when lifting the handset on the called
telephone. Call the voice messaging system from that telephone. Use the voice
messaging menus to retrieve and delete the voice message, verifying that DTMF is
interpreted correctly by the system. Hang up, and then lift the handset again, verifying
that normal dial tone is heard.
4
Connect analog fax machines to ATA ports and other analog ports supported on Avaya
Media Gateways, as applicable. Confirm accurate fax transmissions among the ports.
8. Conclusion
The Cisco ATA 186 SIP gateway was successfully tested with the Avaya SIP offering. During
testing the following observations were made:



G.729AB is not supported by Cisco ATA 186
Message Waiting Indication does not function when using the Cisco ATA 186.
T.38 standard is not supported by Cisco ATA 186, used relay mode
9. Additional References
[1]
[2]
[3]
Installing and Administering SIP Enablement Services, Doc ID 03-602508, available at
http://support.avaya.com
Administrator Guide for Avaya Communication Manager, Doc ID 03-300509, available
at http://support.avaya.com
Cisco ATA 186 Analog Telephone Adaptor Administrator’s Guide for SIP (version 3.0),
available at http://www.cisco.com
RJ; Reviewed:
SPOC 03/08/2010
Solution & Interoperability Test Lab Application Notes
©2010 Avaya Inc. All Rights Reserved.
19 of 22
ATA186_SES
Appendix A
ATA 186 configuration file for the sample configuration.
#txt
UIPassword:0
UseTftp:1
TftpURL:135.64.186.244
CfgInterval:3600
EncryptKey:0
upgradecode:0,0x301,0x0400,0x0200,135.64.186.244,69,0x050616a,ata030201SIP050
616a.zup
dhcp:1
DNS1IP:135.64.186.5
DNS2IP:0.0.0.0
NTPIP:0.0.0.0
AltNTPIP:0.0.0.0
VLANSetting:0x0000002b
PortsSetting:0x00000044
L2KeepAlive:0
Proxy:135.64.186.89
AltGk:0
SecProxy:0
AltGkTimeOut:0
LoginID0:0
LoginID1:0
UseLoginID:0
SIPPort:5060
SIPRegInterval:3600
SIPRegOn:1
MaxRedirect:5
SipOutBoundProxy:0
NatServer:0
NatTimer:0x00000000
MsgRetryLimits:0x00000000
SessionTimer:0x00000000
SessionInterval:1800
MinSessionInterval:1800
DisplayName0:ATA186R P1
DisplayName1:ATA186R P2
ACRDN:0
MediaPort:16384
RxCodec:2
TxCodec:2
LBRCodec:3
RJ; Reviewed:
SPOC 03/08/2010
Solution & Interoperability Test Lab Application Notes
©2010 Avaya Inc. All Rights Reserved.
20 of 22
ATA186_SES
AudioMode:0x00150015
NumTxFrames:2
TOS:0x000068B8
PaidFeatures:0xffffffff
CallFeatures:0xffffffff
CallCmd:Af;AH;BS;NA;CS;NA;Df;EB;Ff;EP;Kf;EFh;HQ;Jf;AFh;HQ;I*67;gA*82;fA#90v#;
OI;H#72v#;bA#74v#;cA#75v#;dA#73;eA*67;gA*82;fA*70;iA*69;DA*99;xA;Uh;GQ;Af;AH;
R3;
FeatureTimer:0x00000000
FeatureTimer2:0x0000001e
SigTimer:0x01418564
ConnectMode:0x02060400
OpFlags:0x00000002
TimeZone:17
CallerIdMethod:0x00019e60
Polarity:0
FXSInputLevel:-1
FXSOutputLevel:-4
DialTone:2,31538,30831,1380,1740,1,0,0,1000,0,0
BusyTone:2,30467,28959,1191,1513,0,4000,4000,0,0,0
ReorderTone:2,30467,28959,1191,1513,0,2000,2000,0,0,0,0,0,0,0,0,0
RingBackTone:2,30831,30467,1943,2111,0,16000,32000,0,0,0
CallWaitTone:1,30831,0,5493,0,0,2400,2400,4800,0,0
AlertTone:1,30467,0,5970,0,0,480,480,1920,0,0
SITone:2,30467,28959,1191,1513,0,2000,2000,0,0,0,0,0,0,0,0,0
RingOnOffTime:2,4,25
DialPlan:*S.|#S.|6..|[37]....|91..........
IPDialPlan:1
NPrintf:0
TraceFlags:0x00000000
SyslogIP:0.0.0.0.514
SyslogCtrl:0x00000000
RJ; Reviewed:
SPOC 03/08/2010
Solution & Interoperability Test Lab Application Notes
©2010 Avaya Inc. All Rights Reserved.
21 of 22
ATA186_SES
©2010 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 Solution &
Interoperability Test Lab at interoplabnotes@list.avaya.com
RJ; Reviewed:
SPOC 03/08/2010
Solution & Interoperability Test Lab Application Notes
©2010 Avaya Inc. All Rights Reserved.
22 of 22
ATA186_SES
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