Table Of Contents
Application Note
Introduction
Network Topology
Limitations
Calling Name and Number Feature
Call Forwarding
System Components
Hardware Requirements
Software Requirements
Features
Configuring the Alcatel 4400 PBX
Configure "ISO Function" System Parameters
Configure Board
Configure Digital Access Options
Configure Trunk Group
Configure Trunk Detail
Configure T2 Access
Configuring CCM—6608-E1 Gateway Configuration
Enbloc Route Pattern Configuration
Overlap Sending Route Pattern Configuration
References
Appendix A
Alcatel 4400 Software Release
Cisco Catalyst 6000 Switch Configuration
Test Configuration
Appendix B
Test Results
Application Note
Alcatel 4400 Rel3.2 PBX with Cisco CallManager (CCM) Using 6608-E1 QSIG Protocol as MGCP Gateway
This application note illustrates connectivity for Alcatel 4400 Release 3.2 PBX with Cisco CallManager using Cisco 6608-E1 QSIG as MGCP Gateway.
Introduction
Key attributes of the network environment used for this application note:
•
The network topology diagram shows the test set-up for end-to-end interoperability with the Cisco CallManager (CCM) connected to the PBX via 6608-E1 QSIG link as MGCP Gateway.
•Connectivity is achieved by using the PRI ISO QSIG E1 protocol type on the MGCP gateway and ABC-F (ISO version) as protocol type on the Alcatel 4400 PBX.
•Basic Calls worked fine in both directions with Calling/Connected Name and Calling Number features support.
•Alcatel 4400 PBX does not honor Connected Number Identification Presentation feature. Alcatel phone does not display Connected Number even though CCM is sending the information in the connect message.
•Call Forwarding did not work when the IP phone calls an Alcatel phone that is setup to forward to another Alcatel phone.
Network Topology
Figure 1
Network Topology or Test Setup
Limitations
Calling Name and Number Feature
When calling from Alcatel digital phone to Cisco 7960 IP phone, the Cisco IP phone displays Calling Name and Number when the call is answered. Alcatel phone however displays connected Name only, No connected Number when the call is answered. Alcatel does not honor Connected Number Identification Presentation QSIG service even though CCM is sending the Connected Number information in the connect message.
CCM Seaview release does not support sending Alerting Name or Busy Name Identification information.
Call Forwarding
Calls did not complete when the IP phone calls an Alcatel phone that is setup to forward to another Alcatel phone. Alcatel in this case sends Facility IE for 'CallRerouting'. Since CCM3.3 Seaview release does not support CallRerouting, the call is not completed.
System Components
Hardware Requirements
Cisco Hardware:
•Cisco Cat6K switch with 6608-E1 Gateway
•Cisco CM 3.3
Alcatel 4400 PBX Hardware:
•PRA2, 3BA23076- E1 PRI/QSIG
Software Requirements
Alcatel: PBX Software version 3.2
Cisco: Cisco CM 3.3
Features
Key features supported:
•Calling Name Identification Presentation
•Calling Number Identification Presentation
•Calling Number Identification Restriction
•Connected Name Identification Presentation
•Connected Number Identification Presentation
•Honor Calling/Connected Name Identification Restriction
•Honor Calling/Connected Number Identification Restriction
•Key features not supported:
•Sending Alerting Name Identification
•Sending Busy Name Identification
•Sending Calling/Connected Name Identification Restriction
•Sending Connected Number Identification Restriction
•Updating Connected Number on Alcatel phone for Basic Calls
•Updating Connected Name and Number for Call Transfers
•Updating Connected Name and Number for Call Forwarding
Configuring the Alcatel 4400 PBX
Configure in the following sequence:
1. "Configure "ISO Function" System Parameters" on page 4
2. "Configure Board" on page 5
3. "Configure Digital Access Options" on page 10
4. "Configure Trunk Group" on page 12
5. "Configure Trunk Detail" on page 13
6. "Configure T2 Access" on page 15
Configure "ISO Function" System Parameters
Command: \compidea\System::1\Other System Param::1
Table 1 Other System Parameter Listing
Parameter
|
Value
|
Instance (reserved)
|
1
|
Trunk seizure via attendant
|
True
|
No detect.of On-hook tone
|
True
|
TrkGrp in ticket for trans.call
|
True
|
VPN service
|
False
|
ISVPN Node No.
|
1
|
Nb Digits displayed on sets
|
1
|
Melody Ringing Type
|
1
|
Int.Call Ringing Cadence No
|
1
|
Ext.Call Ringing Cadence No
|
1
|
Executive Type Ringing Cadence No
|
1
|
Prioritary Call Cadence No
|
1
|
ISO Function
|
True
|
Booking B Channel
|
True
|
Uncontrol.Business Account Code
|
False
|
Business Pref.With Business No.
|
True
|
Project prefix With Code
|
False
|
Follow-Me on Remote forwarding
|
False
|
BC HLC Fax
|
1
|
VG Recording Gain from an UA set
|
3
|
Calling Id.length
|
7
|
Nb Of Secret Code Errors
|
0
|
Transfer All Business Call Types
|
True
|
Attendt Keep Local on Enqu.cancel
|
False
|
Compatibility GF
|
True
|
Alphanum.Char.Entering-mode2
|
True
|
Spain version 2
|
False
|
Delete First STAR Digit
|
True
|
QSIG1 (reserved)
|
0
|
QSIG2 (reserved)
|
0
|
Stop Tie Line Supervision
|
False
|
Tie Line Germany
|
0
|
Entity For Virtual Set
|
0
|
Nb Of Business Code Errors
|
0
|
Disabled Code Duration
|
0
|
Poor ARS Rerouting memo.Dur.
|
20
|
Poor ARS Route Inhibit Dur.
|
180
|
Charging by Nb MiniMessages
|
True
|
Remote Numeric Gain For 4630
|
True
|
Project Code In Redial Key
|
True
|
Cx on Progress message
|
False
|
Differred transm.(Swiss trick)
|
False
|
Send NDS NDI
|
False
|
Calls Distributed On Att.In Order
|
False
|
Nb non answ. int. messages by set
|
16
|
NS_read_before_ack
|
False
|
SNCM
|
0
|
Configure Board
Note: Interface type must be set to PRA2
Command: \compidea\Shelf::0\Board::4
Table 2 Board Parameter Listings
Board Parameter
|
Value
|
Board Address
|
4
|
Interface Type
|
PRA2
|
Administrative status
|
Enabled
|
Usage State
|
Busy
|
Operational State
|
Enabled
|
Main/Standby State
|
Main (Master)
|
Number Of Sets Being Connect.
|
1
|
Remote Shelf Address
|
255
|
Remote Board Address
|
255
|
Synchronisation Priority
|
255
|
IO2 With SPB
|
NO
|
AUXU Parameters 1
|
None
|
AUXU Parameters 2
|
None
|
AUXU Parameters 3
|
None
|
AUXU Parameters 4
|
None
|
CRC4
|
YES
|
Country Protocol Type
|
Default
|
Time Slots\0
|
0
|
Time Slots\1
|
1
|
Time Slots\2
|
1
|
Time Slots\3
|
1
|
Time Slots\4
|
1
|
Time Slots\5
|
1
|
Time Slots\6
|
1
|
Time Slots\7
|
1
|
Time Slots\8
|
1
|
Time Slots\9
|
1
|
Time Slots\10
|
1
|
Time Slots\11
|
1
|
Time Slots\12
|
1
|
Time Slots\13
|
1
|
Time Slots\14
|
1
|
Time Slots\15
|
1
|
Time Slots\16
|
0
|
Time Slots\17
|
1
|
Time Slots\18
|
1
|
Time Slots\19
|
1
|
Time Slots\20
|
1
|
Time Slots\21
|
1
|
Time Slots\22
|
1
|
Time Slots\23
|
1
|
Time Slots\24
|
1
|
Time Slots\25
|
1
|
Time Slots\26
|
1
|
Time Slots\27
|
1
|
Time Slots\28
|
1
|
Time Slots\29
|
1
|
Time Slots\30
|
1
|
Time Slots\31
|
1
|
Voice-->Data TS
|
YES
|
SU shelf Type
|
2 PCM Shelf
|
DECT Location area number
|
255
|
Send Init Dynamic Msg
|
False
|
Param By Default
|
True
|
Clock Mode
|
Internal
|
CPU with Optimized B ChannelAccess
|
NO
|
Board with DTM
|
False
|
Incidents Teleservice
|
YES
|
Max.VG Recording Duration
|
0
|
DASS2 Simulate Network
|
NO
|
DPNSS Layer 2 Address
|
A
|
ISDN Board Layer 2 Parameters\Retransmission Timer
|
100
|
ISDN Board Layer 2 Parameters\TEI Identity Check Timer
|
100
|
ISDN Board Layer 2 Parameters\Polling Timer
|
1000
|
ISDN Board Layer 2 Parameters\Nb_Of_Retransmission
|
3
|
ISDN Board Layer 2 Parameters\Max Frame Size (Bytes)
|
260
|
ISDN Board Layer 2 Parameters\Window Size In Frames SAPI S T0
|
1
|
ISDN Board Layer 2 Parameters\Window Size In Frames SAPI P T0
|
3
|
ISDN Board Layer 2 Parameters\Window Size In Frames SAPI S T2
|
7
|
ISDN Board Layer 2 Parameters\Window Size In Frames SAPI P T2
|
7
|
Number of configurated ports
|
1
|
Associated CPU
|
255
|
Number of configurated E1 ports
|
8
|
Synchronisation mode
|
Adaptive method
|
In Band Signalling
|
NO
|
Passive board
|
NO
|
SS7 signalling
|
NO
|
PRA7 TS signalling
|
16
|
Use Data Compression
|
NO
|
Mutual Aid
|
YES
|
LIO Daughter Board
|
6 Compressors
|
Tone on Board
|
R2 Tone
|
Number of Used Compressors
|
0
|
GNISC in Rack
|
255
|
GNISC in position
|
255
|
Usage State
|
Suite Slave
|
Atm address
|
|
TS used on PCM 0
|
0
|
TS used on PCM 1
|
0
|
TS used on PCM 2
|
0
|
TS used on PCM 3
|
0
|
TS used on PCM 4
|
0
|
TS used on PCM 5
|
0
|
TS used on PCM 6
|
0
|
TS used on PCM 7
|
0
|
Daughter board equipped
|
NO
|
Number of Used Compressors
|
0
|
Mode
|
Gateway IP
|
Embedded Ethernet
|
YES
|
Voice Guide Lang Index
|
1
|
CLIP Signalization
|
No CLIP
|
IVR Protocol
|
No IVR Protocol Protocol
|
4615 Present
|
NO
|
LIOE coupler 1 address
|
255
|
LIOE coupler 2 address
|
255
|
Associated BBC2 coupler
|
255
|
Associated BBC2 access
|
255
|
Use of volume in system
|
YES
|
Local volume (dB)
|
0
|
Configure Digital Access Options
Note: Network mode must be set to Yes for (Master/Network) or No- (Slave/User); Access Type must be set to T2.
Command: \compidea\Shelf::0\Board::4\Digital Access::0
Table 3 Digital Access Options Listing
Parameter
|
Value
|
T0/T2 Access No.
|
0
|
Access Type
|
T2
|
Used Access
|
YES
|
Synchronisation Priority
|
255
|
Network Mode
|
YES
|
Max Nb Of Used B Channels
|
30
|
Max_Nb_Of_Compressed_B_Channels
|
0
|
Nb Of Signalization TS
|
1
|
TieLine Mode
|
YES
|
With Alarm
|
NO
|
Access Type S0
|
NO
|
Reserved1
|
NO
|
Reserved2
|
NO
|
Network Date Time Update
|
NO
|
CRC4
|
YES
|
Port Class
|
NOT SIG
|
Multiframe Type
|
SF
|
Line Type
|
Short Haul 0 to 35 meters
|
Pulses Encoding
|
AMI
|
Retransmission Timer
|
100
|
TEI Identity Check Timer
|
100
|
Polling Timer
|
1000
|
Nb_Of_Retransmission
|
3
|
Max Frame Size (Bytes)
|
260
|
Window Size In Frames SAPI S
|
7
|
Window Size In SAPI P
|
7
|
B Channel Rate
|
64K
|
| |
|
| |
64K
|
Configure Trunk Group
Note: Q931 signal variant is used to set the protocol type to ABC-F
Command: \compidea\Trunk Groups::1
Table 4 Trunk Group Parameter Listing
Parameter
|
Value
|
Trunk Groups
|
|
Trunk Group Id
|
1
|
Trunk Group Type
|
T2
|
Trunk Group Name
|
PRI-ABCF
|
Node number
|
1
|
Transcom Trunk Group
|
False
|
Auto.reserv.by Attendant
|
False
|
Overflow trunk group No.
|
-1
|
Tone on seizure
|
False
|
Private Trunk Group
|
False
|
Paging Trunk Group
|
False
|
Paging Table Id
|
-1
|
Paging Signalization
|
NDDI
|
Security Patrol
|
False
|
Q931 signal variant
|
ABC-F
|
Operator Id
|
ANSI
|
Number Compatible With
|
-1
|
Prefix Sending
|
False
|
Number Of Digits To Send
|
4
|
Channel selection type
|
Quantum
|
Remote Network
|
15
|
Shared Trunk Group
|
False
|
T.line Calling last dig.length
|
0
|
auto.DTMF dialing on outgoing call
|
NO
|
T2 Specificity
|
None
|
Public Network Category
|
0
|
DDI transcoding
|
False
|
Special Services
|
Nothing
|
Can support UUS in SETUP
|
True
|
Register Signalling
|
Decadic/MF Q23
|
Configure Trunk Detail
Command: \compidea\Trunk Groups::1\Trunk Group::1
Table 5 Trunk Detail Listing
Parameter
|
Value
|
Trunk Group
|
|
Instance (reserved)
|
1
|
Trunk Group Type
|
T2
|
Public Network Ref.
|
|
Dialling end to end
|
NO
|
DTMF end to end signal.
|
NO
|
Paying Incoming Calls
|
NO
|
TS Permanently assigned
|
NO
|
Min. Nb.of digits on seize
|
0
|
Signal.with access code
|
NO
|
Trunk group used in DISA
|
NO
|
DISA Secret Code
|
|
VG for non-existent No.
|
YES
|
Routing To Executive
|
NO
|
Trunk Category Id
|
19
|
Nb of digits unused (ISDN)
|
4
|
B Channel Choice
|
NO
|
Channels Reserved By Attend.
|
0
|
Dissuasion For ACD
|
NO
|
DTO joining
|
NO
|
Enquiry Call On B Channel
|
NO
|
DDI Mode
|
NO
|
Automated Attendant
|
NO
|
Calling party Rights category
|
0
|
Entity Number
|
0
|
TS Overflow
|
YES
|
Number To Be Added
|
|
Supervised by Routing
|
NO
|
Access Cluster Id
|
-1
|
VPN Cost Limit for Incom.Calls
|
0
|
Immediat Trk Listening For VPNCall
|
YES
|
VPN TS %
|
50
|
Csta Monitored
|
NO
|
Max.% of trunks out CCD
|
0
|
Charge Calling And ADN Creation
|
NO
|
Ratio analog.to ISDN tax
|
|
Collect Calls Allowed
|
YES
|
Priority of Call
|
NO
|
PCM Network Mode
|
NO
|
LogicalChannel
|
1__15 & 17__31
|
TS Distribution on Accesses
|
YES
|
Use Split Acces
|
NO
|
Heterogeneous Remote Network
|
NO
|
Barring mode
|
Not barred
|
ARS class of service
|
31
|
Quality profile for voice on IP
|
Profile #1
|
IP compression type
|
Default
|
Use of volume in system
|
YES
|
Local volume (dB)
|
0
|
Configure T2 Access
Command: \compidea\Trunk Groups::1\T2/T1/T0 Access::0-4-0
Table 6 T2 Access Parameter Listing
Parameter
|
Value
|
T2/T1/T0 Access
|
|
Physical Address
|
0-4-0
|
Access Type
|
T2
|
Access Cluster Id
|
-1
|
Time Slots T2
|
01111111111111110111111111111111
|
Time Slots T0
|
011
|
Time Slots T1 CCS
|
0111111111111111111111110
|
Time Slots Virtual
|
01111110000000
|
DLCI
|
16
|
Committed Information Rate
|
48
|
Extended Information Rate
|
64
|
CIR Measurement Interval
|
10
|
Support Time Slot Address
|
|
ISDN Compression Number
|
|
Release Support Timer (100ms)
|
0
|
Protection Timer (1mn)
|
0
|
Configuring CCM—6608-E1 Gateway Configuration
Enbloc Route Pattern Configuration
Overlap Sending Route Pattern Configuration
References
Alcatel 4400 Rel4.2 PBX with CCM using 6608-E1 QSIG protocol as MGCP Gateway (Application Note)
Appendix A
CCM Software release:
Alcatel 4400 Software Release
Command: \compidea\node
Table 7 Alcatel Software Parameter Info
Parameter
|
Value
|
Node
|
|
Node Number (reserved)
|
1
|
Software Version
|
R3.2
|
Version name
|
c1.712
|
Patch No.
|
5
|
Notes
|
|
Object Identity
|
|
Node Number (reserved)
|
1
|
Ethernet Notes\Netmask
|
|
Ethernet Notes\Local CPU\Name
|
x000000_tun
|
Ethernet Notes\Local CPU\IP Address
|
172.30.253.253
|
Ethernet Notes\Twin Cpu\Name
|
|
Ethernet Notes\Twin Cpu\IP Address
|
|
Ethernet Notes\Main Cpu\Name
|
xm000000
|
Ethernet Notes\Main Cpu\IP Address
|
10.253.253.3
|
Ethernet Notes\StandBy Cpu\Name
|
|
Ethernet Notes\StandBy Cpu\IP Address
|
|
SL Notes
|
|
IP/X25 Tunnel Notes\Netmask
|
255.255.0.0
|
IP/X25 Tunnel Notes\Local Node\Name
|
x000000_tun
|
IP/X25 Tunnel Notes\Local Node\IP Address
|
172.30.253.253
|
Cisco Catalyst 6000 Switch Configuration
Console> (enable) sh version
WS-C6506 Software, Version NmpSW: 6.1(4)
Copyright (c) 1995-2001 by Cisco Systems
NMP S/W compiled on May 15 2001, 12:27:20
System Bootstrap Version: 5.3(1)
Hardware Version: 2.0 Model: WS-C6506 Serial #: TBA04110341
Mod Port Model Serial # Versions
--- ---- ------------------- ----------- --------------------------------------
1 2 WS-X6K-SUP1A-2GE SAD041504XL Hw : 3.1
WS-F6K-PFC SAD0413097K Hw : 1.1
3 48 WS-X6248-RJ-45 SAD04150CK1 Hw : 1.2
5 8 WS-X6608-E1 SAD043300AJ Hw : 1.1
HP1: D00403030007; DSP1: D005b033 (3.6.
HP2: D00403030007; DSP2: D005b033 (3.6.
HP3: D00403030007; DSP3: D005B033 (3.6.
HP4: D00403030007; DSP4: D005B033 (3.6.
HP5: C00103010007; DSP5: C002E031 (3.3.
HP6: C00103010007; DSP6: C002E031 (3.3.
HP7: C00103010007; DSP7: C002E031 (3.3.
HP8: C00103010007; DSP8: C002E031 (3.3.
Module Total Used Free Total Used Free Total Used Free
------ ------- ------- ------- ------- ------- ------- ----- ----- -----
1 65408K 43645K 21763K 16384K 5327K 11057K 512K 245K 267K
Uptime is 199 days, 22 hours, 13 minutes
Console> (enable) sh module
Mod Slot Ports Module-Type Model Sub Status
--- ---- ----- ------------------------- ------------------- --- --------
1 1 2 1000BaseX Supervisor WS-X6K-SUP1A-2GE yes ok
3 3 48 10/100BaseTX Ethernet WS-X6248-RJ-45 no ok
5 5 8 E1 WS-X6608-E1 no ok
Mod Module-Name Serial-Num
--- ------------------- -----------
Mod MAC-Address(es) Hw Fw Sw
--- -------------------------------------- ------ ---------- -----------------
1 00-d0-d3-37-f9-8e to 00-d0-d3-37-f9-8f 3.1 5.3(1) 6.1(4)
00-d0-d3-37-f9-8c to 00-d0-d3-37-f9-8d
00-01-63-af-5c-00 to 00-01-63-af-5f-ff
3 00-01-97-4a-10-30 to 00-01-97-4a-10-5f 1.2 5.1(1)CSX 6.1(4)
5 00-01-64-12-22-80 to 00-01-64-12-22-87 1.1 5.4(2) 6.1(4)
Mod Sub-Type Sub-Model Sub-Serial Sub-Hw
--- ----------------------- ------------------- ----------- ------
1 L3 Switching Engine WS-F6K-PFC SAD0413097K 1.1
Console> (enable) sh port 5/1
Port Name Status Vlan Duplex Speed Type
----- ------------------ ---------- ---------- ------ ----- ------------
5/1 connected 1 full 2.048 E1
Port DHCP MAC-Address IP-Address Subnet-Mask
-------- ------- ----------------- --------------- ---------------
5/1 enable 00-01-64-12-22-80 10.10.10.104 255.255.255.0
Port Call-Manager(s) DHCP-Server TFTP-Server Gateway
-------- ----------------- --------------- --------------- ---------------
5/1 10.10.10.1 10.10.10.1 10.10.10.1 10.10.10.1
Port DNS-Server(s) Domain
-------- ----------------- -------------------------------------------------
Port CallManagerState DSP-Type
-------- ---------------- --------
Port NoiseRegen NonLinearProcessing
----- ---------- -------------------
Test Configuration
Figure 2
Test Configuration Diagram
As shown in Figure 2, an Alcatel 4400 PBX was connected via an ISDN E1 PRI QSIG link to a Cisco 6608-E1 Gateway, which in turn, was connected to an Ethernet switch. The interoperability testing involved Layers 1, 2 and 3 on the ISDN PRI QSIG link between a Cisco 6608-E1 and the PBX.
Layer 1 (Physical Layer)
The Alcatel 4400 PBX configuration screen for the ISDN PRI QSIG trunk interface is reached using both Alcatel Board and Board\Digital Access menus, setting the PRI physical layer parameters.
Layers 2 & 3 (Q.921 and Q.931)
Layer 2 and 3 packet exchanges were monitored using an Acacia Clarinet protocol analyzer, bridged across the PRI link in high impedance mode.
Layer 2 Q.921 packets were monitored to ensure that each PBX/6608-E1 software configuration properly exchanged SABME/UA packets to initialize the ISDN link, and then RR packets were exchanged every 30 seconds.
Layer 3 Q.931 packets were monitored to ensure that the appropriate call setup/teardown packets were exchanged for each configuration, and that the SETUP packets contained the mandatory Information Elements with the necessary details, as well as optional IEs such as Calling Name and Number.
Telephone calls were made end-to-end in both directions through the Cisco 6608-E1 Gateway, and a check was made to ensure that there was an audio path in both directions for each call.
User/Network Settings
The Cisco 6608-E1 Gateway with ISDN protocol type setting of PRI ISO QSIG E1 supports both protocol sides by selecting "Network/User" in the Protocol Side field when configuring the Gateway via CCM.
The Alcatel 4400, supports both "USER" (slave) and "NETWORK" (master) protocol sides. Network/User options are set in the Board/Digital Access Options menu. Network mode must be set to Yes for (Master/Network) or No - (Slave/User).
The following options for Alcatel PBX are of particular interest:
•Trunk interface type must be set to PRA2.
•Access Type must be set to T2.
Q931 signal variant is used to determine Protocol type. This option was set to ABC-F.
Appendix B
Test Results
Testing was performed by Test Engineer(s): Samir Batio, July 24, 2002
Retesting with CCM3.3-256c load was performed by Test Engineer: Samir Batio, October 1, 2002
Test Setup 1
Test setup was as follows:
•PBX1 configured as ISO version ABC-F, emulates Network.
•Cisco 6608-E1 Gateway configured as PRI ISO QSIG E1, emulates User.
Table 8 Switch and Gateway Settings
Alcatel 4400 Switch-type / Protocol-Side Setting
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Cisco 6608-E1 ISDN protocol-type/Protocol-Side Setting
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ABC-F / Network
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PRI ISO QSIG E1/User
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Table 9 Basic Calls: (Enbloc Sending
Calls Made
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Call Comp?
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"Calling Number" Displayed on Final Destination?
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"Calling Name" Displayed on Final Destination?
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"Called Number" Displayed on Orig. Side?
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"Called Name" Displayed on Orig. Side?
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Notes
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Phone A to Phone C
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Yes
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Yes
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Yes
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No 1
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Yes
|
|
Phone C to Phone A
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Yes
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Yes
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Yes
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Yes
|
Yes
|
|
1 Alcatel ISO version ABC-F QSIG link does not honor Connected Number Identification Presentation.
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Table 10 Basic Calls with Overlap Sending/Receiving: (check trace to verify Overlap mode)
Calls Made
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Call Comp?
|
"Calling Number" Displayed on Final Destination?
|
"Calling Name" Displayed on Final Destination?
|
"Called Number" Displayed on Orig. Side?
|
"Called Name" Displayed on Orig. Side?
|
Notes
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Phone A to Phone C
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Yes
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Yes
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Yes
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No
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Yes
|
|
Phone C to Phone A
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Yes
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Yes
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Yes
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Yes
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Yes
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1
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1 Overlap Receiving on Alcatel PBX works when the Cisco gateway configuration for Called Numbering Plan field is set to `private', `unknown', or `ISDN'.
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Table 11 Call Transfers: (Supervised Local Transfers)
Calls Made
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Call Comp?
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Orig. "Calling Number" displayed on Final Dest. phone?
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Orig. "Calling Name" displayed on Final Dest. phone?
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"Called Number" display on Orig. phone updated after transfer?
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"Called Name" display on Orig. phone updated after transfer?
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Notes
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Phone C to Phone A Xfr to Phone B
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Yes
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Yes
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Yes
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No
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No
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1
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Phone A to Phone C Xfr to Phone D
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Yes
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Yes
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Yes
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No
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No
|
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1 CM3.3 Seaview release does not support Updating Connected Name and Number for Call Transfers QSIG supplementary Services.
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Table 12 Call Transfers: (Supervised Network/External)
Calls Made
|
Call Comp?
|
Orig. "Calling Number" displayed on Final Dest. phone?
|
Orig. "Calling Name" displayed on Final Dest. phone?
|
"Called Number" display on Orig. phone updated after transfer?
|
"Called Name" display on Orig. phone updated after transfer?
|
Notes
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Phone C to Phone A Xfr to Phone D
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Yes
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No
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No
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No
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No
|
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Phone A to Phone C Xfr to Phone B
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Yes
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No
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No
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No
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No
|
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Table 13 Call Conferencing (Local)
Calls Made
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Call Comp?
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"Calling Number" displayed on remaining conferee when the conferencing phone drops out?
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"Calling Name" displayed on remaining conferee when the conferencing phone drops out?
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"Connected Number" updated on Orig. Caller phone display when a conferee drops out?
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"Connected Name" updated on Orig. Caller phone display when a conferee drops out?
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Notes
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Phone C to Phone A,
Phone A conf Phone B
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Yes
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(A Drops out) Yes
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(A Drops out) Yes
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(A Drops out) No
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(A Drops out) No
|
1
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Phone C to Phone A,
Phone C conf Phone D
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Yes
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(C Drops out) No
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(C Drops out) No
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(D Drops out) No
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(D Drops out) No
|
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Phone A to Phone C,
Phone C conf Phone D
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Yes
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(C Drops out) No
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(C Drops out) No
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(C Drops out) No
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(C Drops out) No
|
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Phone A to Phone C,
Phone A conf Phone B
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Yes
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(A Drops out) No
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(A Drops out) Yes
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(B Drops out) No
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(B Drops out) Yes
|
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Table 14 Call Conferencing (Network/External)
Calls Made
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Call Comp?
|
"Calling Number" displayed on remaining conferee when the conferencing phone drops out?
|
"Calling Name" displayed on remaining conferee when the conferencing phone drops out?
|
"Connected Number" updated on Orig. Caller phone display when a conferee drops out?
|
"Connected Name" updated on Orig. Caller phone display when a conferee drops out?
|
Notes
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Phone C to Phone A,
Phone A conf Phone D
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Yes
|
(A Drops out) No
|
(A Drops out) No
|
(A Drops out) No
|
(A Drops out) No
|
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Phone C to Phone A,
Phone C conf Phone B
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Yes
|
(C Drops out) Yes
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(C Drops out) Yes
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(A Drops out) No
|
(A Drops out) No
|
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Phone A to Phone C,
Phone C conf Phone B
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Yes
|
(C Drops out) No
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(C Drops out) No
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(C Drops out) No
|
(C Drops out) No
|
|
Phone A to Phone C,
Phone A conf Phone D
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Yes
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(A Drops out) No
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(A Drops out) No
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(C Drops out) No
|
(C Drops out)Yes
|
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Table 15 Call Forward (Local)
Calls Made
|
Call Comp?
|
Original "Calling Number" Displayed on Final Dest.?
|
Original "Calling Name" Displayed on Final Dest.?
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Forwarding "Called Number" Displayed on Final Dest.?
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Forwarding "Called Name" Displayed on Final Dest.?
|
Final dest. "Connected Number" updated at orig. side?
|
Final dest. "Connected Name" updated at orig. side?
|
Notes
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Phone C to Phone A fwd to Phone B
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No
|
-
|
-
|
-
|
-
|
-
|
-
|
1
|
Phone A to Phone C fwd to Phone D
|
Yes
|
Yes
|
Yes
|
No
|
No
|
No
|
Yes
|
|
Table 16 Call Forward (Network/External)
Calls Made
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Call Comp?
|
Original "Calling Number" Displayed on Final Dest.?
|
Original "Calling Name" Displayed on Final Dest.?
|
Forwarding "Called Number" Displayed on Final Dest.?
|
Forwarding "Called Name" Displayed on Final Dest.?
|
Final dest. "Connected Number" updated at orig. side?
|
Final dest. "Connected Name" updated at orig. side?
|
Notes
|
Phone C to Phone A fwd to Phone D
|
Yes
|
Yes
|
No
|
No
|
Yes
|
Yes
|
No
|
|
Phone A to Phone C fwd to Phone B
|
Yes
|
Yes
|
No
|
No
|
No
|
No
|
Yes
|
|
Test Setup 2
Setup was as follows:
•PBX1 configured as ISO version ABC-F, emulates User
•Cisco 6608-E1 Gateway configured as PRI ISO QSIG E1, emulates Network
.
