EN8811EN00EN

Siemens Enterprise Communications GmbH & Co. KG 2011. All rights reserved.
Siemens Enterprise Communications GmbH & Co. KG is a trademark licensee of Siemens AG
1
OpenScape Voice Siemens
OpenScape Voice
Contents
1 CMP GUI Changes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5
1.1 CMP Overview. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5
1.2 Configuration tab . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 7
1.2.1 OpenScape Voice . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 8
1.2.2 OpenScape Branch . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 10
1.2.3 RG8700 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 11
1.2.4 Unified Communications . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 12
1.2.5 CMP. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 13
1.3 Maintenance tab . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 14
1.4 User Management . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 15
2 Flexible Ethernet circuit and IP Address Configuration . . . . . . . . . . . . . . . . . . . . 17
2.1 Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 17
2.2 Functional Description . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 19
2.3 Node Configuration Parameters Editor (NCPE) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 26
3 Remote Upgrade . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 29
3.1 Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 29
3.2 Supported Platfforms / Configurations. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 29
3.3 Upgrade Procedure . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 30
4 RTP Parameter Management via Assistant. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 33
4.1 Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 33
4.2 Functional Description . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 33
4.3 Usage . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 34
5 Provisioning Errors in CMP . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 37
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Siemens Enterprise Communications GmbH & Co. KG 2011. All rights reserved.
Siemens Enterprise Communications GmbH & Co. KG is a trademark licensee of Siemens AG
2
OpenScape Voice Siemens
5.1 Overview. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 37
5.2 Functional Description . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 38
5.3 Usage . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 39
6 OSV Node Status Management . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 41
6.1 Overview. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 41
6.2 Usage . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 42
6.2.1 Change Functional Level . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 44
6.2.2 Restart Node. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 45
6.2.3 Power Down . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 46
6.2.4 Change Operational Mode . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 47
6.2.5 SRXCTRL . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 49
7 Activate and Synchronize OSV partitions via CMP. . . . . . . . . . . . . . . . . . . . . . . . 51
7.1 Overview. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 51
7.2 Functional Description . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 51
7.3 Usage . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 53
7.3.1 Activate Backup version and Synchronize Active version to Backup . . . . . . 53
7.3.2 Activate backup version . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 56
7.3.3 Software Activation enhancements. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 58
8 Split Audit Functionality from
Registration Renewal 63
8.1 Overview. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 63
8.2 Functional Description . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 63
8.3 Provisioning . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 65
9 Emergency Call SNMP Trap. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 69
9.1 Overview. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 69
9.2 Functional Description . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 69
9.3 Emergency Call Routing. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 70
9.4 Provisioning . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 71
9.4.1 Configure Emergency Calling . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 71
9.4.2 Enabling E911 SNMP Traps . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 77
10 SIPQ CodSet4 Transit Counter IE . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 81
10.1 Overview. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 81
10.2 Functional Description . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 81
10.3 Provisioning . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 83
11 Best Effort SRTP for SDES . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 85
11.1 Overview. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 85
11.2 Functional Description . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 87
11.3 Provisioning . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 92
12 Support for multiple prime line registration. . . . . . . . . . . . . . . . . . . . . . . . . . . . . 103
12.1 Overview. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 103
12.2 Application secanrio . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 105
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Siemens Enterprise Communications GmbH & Co. KG 2011. All rights reserved.
Siemens Enterprise Communications GmbH & Co. KG is a trademark licensee of Siemens AG
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OpenScape Voice Siemens
12.3 Functional description . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 105
12.4 Provisioning . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 106
13 PIN for private / business calls. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 109
13.1 Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 109
13.2 Functional Description . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 110
13.3 Provisioning . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 112
13.3.1 CDR . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 116
13.4 Usage . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 119
14 Intercom / Speaker Control. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 121
14.1 Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 121
14.2 Functional Description . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 121
14.2.1 Limitations . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 122
14.3 Provisioning . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 123
14.4 Usage . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 132
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Siemens Enterprise Communications GmbH & Co. KG 2011. All rights reserved.
Siemens Enterprise Communications GmbH & Co. KG is a trademark licensee of Siemens AG
4
OpenScape Voice Siemens
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Siemens Enterprise Communications GmbH & Co. KG 2011. All rights reserved.
Siemens Enterprise Communications GmbH & Co. KG is a trademark licensee of Siemens AG
5
OpenScape Voice Siemens
1 CMP GUI Changes
1.1 CMP Overview
The new default screen after successful logon.
Active Alarm view
Current user settings The new tab bar
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Siemens Enterprise Communications GmbH & Co. KG 2011. All rights reserved.
Siemens Enterprise Communications GmbH & Co. KG is a trademark licensee of Siemens AG
6
OpenScape Voice Siemens
In Version 6 your personalized user options are limited to Number of Rows and Disabling Con-
firmations.
The timeout value has been moved to Configuration CMP System_Settings
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Siemens Enterprise Communications GmbH & Co. KG 2011. All rights reserved.
Siemens Enterprise Communications GmbH & Co. KG is a trademark licensee of Siemens AG
7
OpenScape Voice Siemens
1.2 Configuration tab
We have a completely redesigned Configuration tab where all the various assistants have
been moved.
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EN8811EN00EN
Siemens Enterprise Communications GmbH & Co. KG 2011. All rights reserved.
Siemens Enterprise Communications GmbH & Co. KG is a trademark licensee of Siemens AG
8
OpenScape Voice Siemens
1.2.1 OpenScape Voice
This is the biggest change for your daily use. The tabs are gone and everything is ICON based.
There explanations are at the bottom.
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EN8811EN00EN
Siemens Enterprise Communications GmbH & Co. KG 2011. All rights reserved.
Siemens Enterprise Communications GmbH & Co. KG is a trademark licensee of Siemens AG
9
OpenScape Voice Siemens
We have opened each ICON to show the available content. As you can see these are very sim-
ilar to what was available previously via the buttons on the top for BG Global General and
Administration.
V6
V5
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Siemens Enterprise Communications GmbH & Co. KG 2011. All rights reserved.
Siemens Enterprise Communications GmbH & Co. KG is a trademark licensee of Siemens AG
10
OpenScape Voice Siemens
1.2.2 OpenScape Branch
Items have been moved from buttons across the top, to the side, new items to be covered du-
ring the OpenScape Branch course.
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Siemens Enterprise Communications GmbH & Co. KG 2011. All rights reserved.
Siemens Enterprise Communications GmbH & Co. KG is a trademark licensee of Siemens AG
11
OpenScape Voice Siemens
1.2.3 RG8700
The RG8700 menu items have been moved from buttons across the top, to the side, but nothing
really new.
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Siemens Enterprise Communications GmbH & Co. KG 2011. All rights reserved.
Siemens Enterprise Communications GmbH & Co. KG is a trademark licensee of Siemens AG
12
OpenScape Voice Siemens
1.2.4 Unified Communications
The Unified Communications menu was previously found in the Users & Resources tab.
Plus a few new fields that will be shown during the UC Training course.
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Siemens Enterprise Communications GmbH & Co. KG 2011. All rights reserved.
Siemens Enterprise Communications GmbH & Co. KG is a trademark licensee of Siemens AG
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OpenScape Voice Siemens
1.2.5 CMP
You now have a tab for CMP information / configuration.
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Siemens Enterprise Communications GmbH & Co. KG 2011. All rights reserved.
Siemens Enterprise Communications GmbH & Co. KG is a trademark licensee of Siemens AG
14
OpenScape Voice Siemens
1.3 Maintenance tab
The Maintenance Tab is very similar to previous versions.
V6
V5
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Siemens Enterprise Communications GmbH & Co. KG 2011. All rights reserved.
Siemens Enterprise Communications GmbH & Co. KG is a trademark licensee of Siemens AG
15
OpenScape Voice Siemens
1.4 User Management
We also have a completely new tab for User Management. This will be used for global user
management (HPUM is the predecessor application). This will be explained later in the trai-
ning.
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Siemens Enterprise Communications GmbH & Co. KG 2011. All rights reserved.
Siemens Enterprise Communications GmbH & Co. KG is a trademark licensee of Siemens AG
16
OpenScape Voice Siemens
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Siemens Enterprise Communications GmbH & Co. KG 2011. All rights reserved.
Siemens Enterprise Communications GmbH & Co. KG is a trademark licensee of Siemens AG
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OpenScape Voice Siemens
2 Flexible Ethernet circuit and IP Address
Configuration
2.1 Overview
This feature allows for a flexible configuration of Ethernet circuits and IP addresses.
In extreme cases an OSV can now be installed with one used Ethernet circuit (pair in case of
redundancy) and one single IP address.
The OSV feature of flexible Ethernet circuits and IP addresses applies only to new installations,
i.e. split mode upgrade and toolkit migration will not support
G Reduction or combination of IP addresses or Ethernet circuits of current existing in-
stallations
OpenScape Voice V5 node with 4 ethernet circuit pairs
OSV Node 1
Admin
LAN / VLAN / IP subnet
Mtc Controller
(IMM, iRMC)
Signaling
LAN / VLAN / IP subnet
X-Channel
LAN / VLAN / IP subnet
Router
(admin)
Router
(billing)
Router
(signaling)
Router
(x-channel)
1 Admin Eth.
pair
1 Billing Eth .
pair
1 Signaling
Eth. pair
1 X-Ch Eth.
pair
Billing
LAN / VLAN / IP subnet
Layer 2
Switch
with
VLANs
Or
separate
LAN
switches
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18
OpenScape Voice Siemens
OpenScape Voice V6 node with 1 ethernet circuit pair and x-channel shared
However, once OSV is installed on the new SW release
G The Easy IP address changing feature (NCPE) can be used to change the number
of used Ethernet circuits and IP address values.
This feature also applies to virtual OSVs.
OSV Node 1
Mtc Controller
(IMM, iRMC)
1 Eth. Pair
for
Admin,
Billing ,
Signaling,
X-Channel
Admin and Signaling and Billing
and X-Channel Default Router
Layer 2
Switch
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19
OpenScape Voice Siemens
2.2 Functional Description
The OSV node can be installed to use less than the current 4 Ethernet circuit pairs (3 ports for
simplex) and with less IP addresses, down to one single IP address.
The underlying platform (physical and virtual) should still have the default number of Ethernet
ports to allow for future port expansion. The default number of 'platform' ports is:
Entry: 1
Simplex: 3
Duplex: 8
Virtual: 4
However, if a port is not used, i.e. not assigned to admin / billing / signaling / x-channel, it will
be ignored by OSV SW (e.g. never alarmed).
For the virtual OSV this means that the virtual machine always needs to be created with 4 net-
work connections, in the following sequence:
NIC 1: admin
NIC 2: signaling
NIC 3: billing
NIC 4: cross_connect
Even if, e.g., NIC 2 and NIC 3 are not used if signaling, billing and admin are assigned to the
same IP subnet labeled admin.
7
The billing and cross connection on virtual installations has been swapped in V6 !
Remember to change the guest vSwitch mappings !
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With this feature, OSV installation associates logical interfaces with physical interfaces by assigning each logical
interface to a nafo-group and Ethernet circuit or bond device:
Example of logical/physical interface mapping in a V5 OSV system using 4 separate subnets and in a V6 OSV
system using 1 shared subnet (eth assignment on IBM X3550 M2/M3 or FSC RX200 S6)
Logical-
Interface
Nafo
Group
Entry
Integrated
Simplex
Standard
Duplex
Virtual
V5 / V6 V5 V6 V5 V6 V5 V6
Admin nafo_alias
eth0
eth0
eth0
bond0 = eth0 + eth4
bond0 = eth0 + eth4
eth0
eth0
Signalling nafo_udp eth1 bond1 = eth1 + eth5 eth1
Billing nafo_bms eth2 bond2 = eth2 + eth6 eth2
X-Channel nafo_xch N/A bond3 = eth3 + eth7 eth3
F
N
9
8
2
1
F
N
U
T
F
N
_
B
A
B
Z
Z
Z
E
D
D
Z
E
N
8
8
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21
OpenScape Voice Siemens
OpenScape Voice IPs and Subnets
Function for
IP address
Subnet Remark Virtual
Admin
Admin
The node IP address used for installation and all OAMP
functions
no
LSM On this IP the OpenScape Voice receives SNMP traps
from routers reporting link status changes
yes
RSA Integrated Maintenance Controller of node
(IMM, iRMC or VMK)
no
Signaling
Signaling
Static IP address for the signaling network (no traffic). no
SIP SIP signaling over UDP, TCP or TLS yes
SIP MTLS SIP signaling over MTLS (mutual TLS) yes
MGCP MGCP / NCS signaling for media server yes
CSTA CSTA signaling over TCP yes
SBC LAN Local IP address for the SBC function of the OSV Entry,
used for signaling and media
no
Billing Billing For sending billing (CDR) files no
X-Channel X-Channel For the node to node communication. no
may move if nodes share the same network
>
The purpose of the Superuser IP is to control the 'access control list' that defines
which SOAP TCP/IP port can access what OSV data via the SOAP/XML interface.
This is needed to set up the access control when OSV is configured in a hosted
scenario with customers accessing the OSV directly (not through CMP).
In the hosted scenario, it is expected that access control is made available to an Ad-
ministrator logged on to CMP.
So he can specify the access to different ports based on the customer's access to
specific data, e.g. BG related.
Typically the super IP address is set to CMP's address.
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22
OpenScape Voice Siemens
IPs and Subnets outside OpenScape Voice
Function for
IP address
Subnet Remark
Default GW
Admin
Default admin gateway for source based routing
Gateway for RSA Router IP to reach maintenance controller of partner node
SuperUser IP address of management server with super user
privileges (generally the integrated Assistant)
Survival
Authority
Needs to be accessible from both nodes admin network and
has to be installed at a third location
Default Router
Signaling
Default router for all HiPath traffic not explicitly routed, e.g. by
static routes or source based routing
Default GW Default signaling gateway for source based ruting
Default GW Billing Default billing gateway for source based routing
Gateway GW X-Channel Router IP, if the x-channels subnets of the two nodes are
different
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23
OpenScape Voice Siemens
Standard IP address assignment (4 network segments)
The following tables shows the default IP addresses used from the NCPE with 4 networks.
Note: All netmasks are 255.255.255.240.
Management
1
= If node 1 is down
2
= Not for IBM X3250 M2
Signaling
Billing
Cluster Interconnect
Node Same subnet Separated subnets Simplex Entry
Node 1 Node 2 Node 1 Node 2
Admin Subnet 1.2.3.0 1.2.3.0 1.2.4.0 1.2.3.0 1.2.3.0
Physical .6 .7 .6 .7 .6 .6
LSM .14 .14
(1)
.14
IMM / iRMC .10 .11 .10 .11 .10 .10
(2)
Node Same subnet Separated subnets Simplex Entry
Node 1 Node 2 Node 1 Node 2
Signaling Subnet 1.2.3.48 1.2.3.48 1.2.4.48 1.2.3.48 1.2.3.48
Physical .52 .53 .54 .55 .52 .53
SIP .54 .55 .54 .55 .54 & .55 .55
MLTS .56 .57 .56 .57 .56 & .57 .57
MGCP .58 .59 .58 .59 .58 & .59 .59
CSTA .60 .61 .60 .61 .60 & .61 .61
Node Same subnet Separated subnets Simplex Entry
Node 1 Node 2 Node 1 Node 2
Billing Subnet 1.2.3.32 1.2.3.32 1.2.4.32 1.2.3.32 1.2.3.32
Billing .38 .39 .38 .39 .38
Node Same subnet Separated subnets Simplex Entry
Node 1 Node 2 Node 1 Node 2
X-Ch. Subnet 10.1.3.0 10.2.3.16 10.2.4.16
X-Ch. Logical .1 .2 .22 .23
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24
OpenScape Voice Siemens
Standard IP address assignment (admin, signalling and billing network merged)
The following tables shows the default IP addresses used from the NCPE with 2 networks.
Note: All netmasks are 255.255.255.240.
Management / Signalling / Billing
1
= If node 1 is down
2
= Not for IBM X3250 M2
Node Same subnet Separated subnets Simplex Entry
Node 1 Node 2 Node 1 Node 2
Admin Subnet 1.2.3.0 1.2.3.0 1.2.4.0 1.2.3.0
Physical .6 .7 .6 .7 .6
LSM .14 .14(
1
) .14
IMM / iRMC .10 .11 .10 .11 .10 .10 (
2
)
SIP
.12 .13 .6 .7 .6
SIP-MTLS
MGCP
CSTA
X-Ch. Subnet 10.1.3.0
same as
Admin
same as
Admin
X-Ch. Logical .1 .2 .22 .23
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25
OpenScape Voice Siemens
Standard IP address assignment (minmized set of IP addresses)
The following tables shows the default IP addresses used from the NCPE with one network.
Note: All netmasks are 255.255.255.240.
Management / Signalling / Billing
1
= If node 1 is down
2
= Not for IBM X3250 M2
3
= When the OSV is installed with the same IP address for SIP(UDP/TCP/TLS) and for
SIP(MTLS), then non-standard port 5161 has to be used. All SIP endpoints that are to be
connected via MTLS have to use this non-standard TCP/IP port when connecting to the OS-
VSIP-MTLS uses Port 5161.
This port can be changed via the Parameter Srx/Sip/IpPortMTLS.
UC Mediaserver on Entry / Integrated Simplex
The UC Media Server SIP listening ports and the MS SIP endpoint in the integrated OSV (in-
cluding OSV Entry) will be associated with the non-standard port numbers 5062 and 5063 for
SIP and SIP-TLS signaling between OSV and MS. This includes the creation of the necessary
packet filter rule and the automatic change during upgrade to V6.
Make sure that the UC Media Server SIP listening ports will be changed via CMP from 5060 /
5061 to 5062 / 5063 !
Node Duplex
Same subnet
Duplex
Separated subnets
Simplex Entry
Node 1 Node 2 Node 1 Node 2
Admin Subnet 1.2.3.0 1.2.3.0 1.2.4.0 1.2.3.0 1.2.3.0
Physical Admin
.6 .7 .6 .7
.6 Physical Signalling
Physical Billing
X-Channel
IMM / iRMC .10 .11 .10 .11 .10 .10 (
2
)
SIP
.12
.13
.6 .7 .6 (
3
) .7 (
3
)
SIP-MTLS
MGCP
CSTA
LSM .12 (
1
)
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26
OpenScape Voice Siemens
2.3 Node Configuration Parameters Editor (NCPE)
There are two important changes on the NCPE in the IP Configuration menu:
IP Configuration 1 of 6
>
The VLAN support for the OpenScape Voice Entry has been removed already ion
V5.
Software Installation Server
(not used for OpenScape Voice)
Bondig interface (for redundancy)
Network Interface Configuration
(should not be changed)
Physical IP for Node 1
1.
2.
Default router
Subnet sharing. Can be ether:
G Mgmt-Billing-Signaling-Separated
Same scenario as used up to V5
G Mgmg-Billing-Shared
Combines the Managemnt / Billing
G Mgmt-Billing-Signalling-Shared
Combines all regular networks
Combines X-Link with Managent
Gateway for this segment
Network definition (Node 1 + 2 !)
Physical IP for Node 1
New in V6
Link Status Manager IP
}
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OpenScape Voice Siemens
All IPs which belongs to a specific nework are now located under this screen. e.g. the signalling
IPs:
Only the IPv6 IPs are still located on the 2nd menu (see next page).
Bondig interface (for redundancy)
Network Interface Configuration
(should not be changed)
Physical IP for Node 1
Gateway for this segment
Network definition (Node 1 + 2 !)
Physical IP for Node 2
}
}
}
Virtual IPs for Node 1
Virtual IPs for Node 1
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OpenScape Voice Siemens
IP Configuration 2 of 6
The IP Configuration 2 of 6 menu contains now only the in OpenScape Voice V5 introduced
IPV6 configuration for the SIP, SIP-MTLS and MGCP/NCS signalling managers.
All IPV4 configuration has been moved to the IP Configuration 1 of 6 menu as described on the
previous page.
All other menus of NCPE are unchanged !
}
IPV6 Signalling Managers for Node 1
IPv6 Subnet mask for Signalling Network
IPv6 Default Router for Signalling Network
}
IPV6 Signalling Managers for Node 2
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29
OpenScape Voice Siemens
3 Remote Upgrade
3.1 Overview
The Remote Upgrade procedure can be used to execute the upgrade from on release or DVD
version to another without physical access to the OSV server.
It allows the administrator to upload the Image ISO to the system(s) as well as any patches and
have the system(s) get updated from a remote location.
The procondition is that a Remote Management Card (RSA / RMC) is installed in the server.
3.2 Supported Platfforms / Configurations
The remote upgrade is availabe on a Integrated Simplex, Standard Duplex and Virtual config-
urations (with vSphere Client) on all servers with a Remote Management Card:
G IBM X3250M3
G IBM X3550M2 / M3
G IBM X3650T
G FSC RX330S1
G FSC RX200
The OpenScape Voice Entry (IBM X3250M2) is not supported as it does not have a Remote
Management Card.
The minimum installed release must be V3.1 SMR2 (imagebased installation) or the system
must be converted with the upgrade MOP to a Multi-Partioning installation.
>
This feature has been introduced with V5+.
7
During Remote Upgrades there will be a Service Outage so please schedule this up-
grade during a maintenance window.
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30
OpenScape Voice Siemens
3.3 Upgrade Procedure
The upgrade will be prepared in the following steps:
1. Initially transfer the Toolkit RPM to /software directory
2. Install Toolkit RPM via
rpm -i --replacefiles --replacepkgs UNSPmigration-<version>.rpm:
(This will create the /repository/upload directory for you).
3. Move the Toolkit RPM from /software to /repository/upload
4. Upload the following files to /repository/upload directory on the system(s)
Image ISO
Node configuration (node.cfg.primary|secondary depending on node you are on)
Patchsets (Optional, please use cumulative if possible)
OSV License Files
Response configuration (Optional, the image already creates default response.cfg
files for installation of OSV APPs. Should only be placed on Integrated Systems!)
5. Repeat steps above for Node 2 in a duplex configuration.
# cd /software
# rpm -i --replacefiles --replacepkgs UNSPmigration-V1.05.rpm
Checking rsync package : rsync-2.6.8-53.7
Checking rsync package : package rsync is not installed
Migration tools installed successfully.
# cd /software
# mv UNSPmigration-V1.05.rpm /repository/upload
/
repository
upload
<OSV Image Installation>.iso
node.cfg.<primary or secondary>
V5.00.01.ALL.11_PS0001.tar
V5.00.01.ALL.11_PS0002.tar
V5.00.01.ALL.11_PS0008.tar
xx-xx-xx-xx-xx-xx.xml
.
.
.
V5.00.01.ALL.11_PS0008E01.tar
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OpenScape Voice Siemens
Now, connect to the Remote Management Card to make sure that the upgrade process can be
monitored. Start the upgrade on node 1 by executing the upgrade8k command (this will export
the current configuration and lock the partition) and reboot the system.
Once the system has been rebooted, the license agreement must be confirmed and the ques-
tion for overriding the harddisk must be answered with yes.
# upgrade8k
common8k: Your current working directory is valid.
//////////////////////////////////
Current Active image<kn2100master>:
//////////////////////////////////
Active:primary, Fallback:secondary
Detected scsi disk.
Toolkit Version : UNSPmigration-1.06-01
- Checking root password for expiry.
- Checking solid password for expiry.
- Checking srx password for expiry.
.
.
.
//////////////////////////////////
Current Active image<kn2100master>:
//////////////////////////////////
Active:primary, Fallback:secondary
- Processing for current build V5.00.01.ALL.11.
- Admin ip: 1.21.210.101
Export completed : Thu Sep 8 20:09:49 CEST 2011.
upgrade8k: Checking for licenses to import.
Extracting image dvd.
167216 blocks
upgrade8k: Setting up primary boot device(sda).
/////////////////////////////////////////////////////////////////
upgrade8k:Remember to reboot the node(s) to complete the upgrade.
/////////////////////////////////////////////////////////////////
Done.
Note: Typing yes or format will erase all data from unlocked partition on the disk.
: yes - Erases data from the first unlocked image.
: no - Aborts installation.
: format - Reformats harddisk, user looses all the data.
: lockprim - Erases data from the secondary partition.
: locksec - Erases data from the primary partition.
Do you want to continue with Installation
- Enter yes/no/format/lockprim/locksec) ? yes
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OpenScape Voice Siemens
The upgrade process wil lbe started and executes the following steps:
G Image Installation will be started and the user will be asked (on both nodes if duplex)
whether to proceed with the Image Installation on Inactive Partition. Once user accepts (on
both nodes if duplex), Image Installation will commence.
G System(s) will reboot once Image Install completes.
G If patches were provided these will also get applied.
G Once system(s) reach state 4 the configuration will be imported and the license files will
be applied.
G Partitions will be unlocked
upgrade8k started via
Remote Managemnt Card
Configuration will
be exported
Partition will be locked
Image
Installation
Install OSV
Patches
Partition will be unlocked
Configuration will
be reimported
Apply
License File
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4 RTP Parameter Management via Assistant
4.1 Overview
This feature introduces a generic interface in OpenScape Voice Assistant GUI for the manage-
ment (view / modify) of OpenScape Voice RTP Parameters. This interface will support RTP pa-
rameter data such as
Current value
Default/Recommended value
Value type/range/unit
Description
4.2 Functional Description
Service technicians, can view RTP paramters names and values in OpenScape Voice Assis-
tant in the form of a list and search for a specific RTP parameter or for a set of RTP parameters
with certain search criteria (e.g. Srx/Sip/*).
On an RTP parameter basis, information such as usage, default / recommended value, type,
value ranges, process restart, as well as current value is displayed. The change of the RTP
parameter value is also possible for some RTP parameters.
>
This feature has been introduced with V5+.
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34
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4.3 Usage
The RTP Parameter can be accessed via
Configuration OpenScape Voice Administration General Settings RTP
With the filter bar, a parameter can be search. e.g. to search a specific parameter enter the
parameter name or the first part of the parameter in the Search for: field, change the value in
the in field to Name and click on Search. Only the matched parameters will be shown:
Click on Show All to get the full parameter list.
>
If a parameter has been changed, it will appear in red color.
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To get list of all changed parameters, set the in field to Is Default Value and the Search for:
field to No. As the result, all parameters which are not on his default value will be shown:
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36
OpenScape Voice Siemens
To change a parameter, click on the parameter name. The following window will be opened:
If the parameter should be changed, enter or choose the new value in the Value field and click
on Save.
Parameter name
Value type / range/ unit
(i.e. string/integer/boolean)
Range
(i.e. min. / max., false / true)
Current value
Value
Default / Recommended
value
Unit (i.e. seconds / number)
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OpenScape Voice Siemens
5 Provisioning Errors in CMP
5.1 Overview
Currently (V6 and earlier) Provisioning errors due to inconsistency between the OSV and the
devices are logged in log files (/log/HiqLogProvisioning.log.*).
Prior to Version 6 you were required to review the error logs in file format which looked similar
to this:
This new feature will allow us to display these logged provisioning errors in a format which is
more user friendly.
Accomplished by:
1. Processing all log entries of provisioning errors and extracting specific information.
2. Sending the zipped result file to the users desktop.
3. Extracting files from zipped location to user specified folder
4. User imports the file (based on | character as a delimiter) to an Excel sheet and reviews
the results.
{
serial_number => 38891,
time => 'Tue Apr 12 16:48:07 2011',
time_secs => 1302641287,
time_usecs => 68806,
process_name => 'sipRegistrar11',
file_name => 'sipRegAppl.cc',
line_num => 966,
pid => 31642,
log_priority => 'HIGH',
log_category => 'SUB_PROVISIONING_ERR',
log_appl_reason => 69,
component_id => 'UNKNOWN',
appl_event_id => 69,
num_prior_occurences => 60,
managed_suppression_count => 10,
managed_suppression_time => 1,
data_length => 130,
data => 'REGISTER mesg received with invalid SIP transport from DN=18146610892
device=10.41.10.93:5060. Device transport=UDP configured=TCP',
}
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38
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5.2 Functional Description
This feature extracts specific information from /log/HiqLogProvisioning.log.* files:
G IP address of the node
G Log Category
G Number of Occurrences
G Time and Date of the event
G Data / Description of the error
From this data a unique record for each event with date & time of first and last occurrence will
be created (Execution of processing in asynchronous manner). The result will be provided as
a compressed .csv file using the pipe symbole as the field delimiter.
Provisioning errors that are logged:
1. Contact header in REGISTER msg has Expires value less than allowable OSV Expires
value.
2. Invalid transport type in incoming REGISTER message.
3. Registering DN is not a valid Line appearance on this device.
4. Not Prime Registration. Secondary line is not allowed to register before primary line.
5. Registering endpoint not configured in the OSV as a subscriber line.
6. Digest Auth credentials are not correct or no credentials are configured.
7. Media renegotiation will occur due to unsupported secure media in the SDP.
8. CPU directed feature invoked by subscriber not provisioned for this feature.
9. Subscriber for call pickup service not provisioned in the OSV.
10. A keyset SUBSCRIBE message was received for a DN that is provisioned as a non-keyset
device on the OSV.
11. Invalid event header received in Message Waiting Indicator NOTIFY msg.
12. Invalid Toggle access code invoked or provisioned on the OSV
13. Toggle invoked for subscriber who does not have feature provisioned on the OSV.
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5.3 Usage
The feature is invoked from CMP under:
Maintenance Monitoring Tools & Utilities OpenScape Voice Logging
A .zip archive will be downloaded:
Select the OSV (Switch Name)
from witch the log files should
be retrieve and a log catgory.
Optional, check the Set Time
and Date and choose a Start
Time / Date to reduce the
amount of data to be retrieved.
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The included file can be imported in excel by using the | (pipe key) as a delimiter:
Your content will be displayed and you can then review using the various tools from Excel.
The first line shows the ammount of downloaded records. The main section (row 2) contains
the categories and will help you sort based on these categories or their associated content.
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OpenScape Voice Siemens
6 OSV Node Status Management
6.1 Overview
Service has requested to support the following:
G Status of the nodes
This displays OSV Cluster name, OSV Configuration, Node name, Node Hardware, Level
of operation, Operation Mode, Database Role, X-Channel status, Connectivity with OSV.
G Change the level Operation
This will allow the change of the Node operation level to 2, 3 or 4.
G Restart the Node
This will allow the Node to reboot and attain a level 2, or 3, 4.
G Graceful Shut Down
This will allow to control the shut down of the Node gracefully with an online command in-
stead of power down.
G Operational Mode changes
This will allow to change of the operation mode of the Node to Normal, Stand Alone Sec-
ondary and Stand Alone Primary.
Until V6 none of the above is possible from the CMP and only some of the above functions
were available via CLI.
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6.2 Usage
Navigate to
Maintenance Invetory Nodes
and click on the OSV name which should be controlled:
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Here you can open the window to show the current status of both nodes as well change their
functional state or even power off one node.
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6.2.1 Change Functional Level
The following node status change will be allowed only when the connectivity is UP, if not the
request must be rejected by the CMP with a reason cannot communicate with OpenScape
Voice.
Level <2-OS / 3-DB / 4-OSV>
2 - OS Implies that only Operating System will be running after the execution.
3 - DB Implies that only OS and Solid DB will be running after the execution.
4 - OSV Implies that OS, Solid DB and Applications such as SIPSM, SOAPServer etc are
running afterthe execution.
Hazard warning of total outage (Confirmatory note) shall be provided if the Nodes current level
is 4-OSV and the partner node is not in level 4-OSV.
Request will be executed only when the Hazard warning is accepted.
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6.2.2 Restart Node
This function allows the user to reboot the Node to a desired level OpenScape Voice Operation.
Following are the desired levels and this function can be invokedindependent of the current le-
vel of the Node:
2 - OS Implies that the operating system will be restarted on theNode and the resulting node
state is the OS level.
3 - DB Implies that the Operating system and DB will be restarted on the Node and the re-
sulting node state is the DB level.
4 - OSV Implies that the Operating system, DB and RTP will be restarted on the Node and
the resulting node state is the 4-OSV level.
For all levels above there will be a pop-up dialog warning of total outage (Confirmation Dialog)
shall be provided if the Nodes current state is 4-OSV and the partner node is not in state 4-
OSV. Request shall be executed only when the Hazard warning is accepted.
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6.2.3 Power Down
This function will allow the user to power off the Node.
Hazard warning of total outage (Confirmatory note) shall be provided if the Nodes current state
is 4-OSV and the partner node is not in state 4-OSV.
Request shall be executed only when the Hazard warning is accepted.
An additional note must be provided to power back up the node remotely use the Maintenance
Controller of the Node from CMP available under Operations & Maintenance ->Overview ->3rd
party nodes.
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6.2.4 Change Operational Mode
This function allows changing the operational mode of a Node. CMP will check the compatibi-
lity with the status of the other node as stated below before allowing changing the operational
mode to avoid data loss.
Any of the status change must be initiated within 1 minute of the status being displayed. This
is needed to ensure that user would not change the status of a node assuming the other node
is in proper state but in reality the status could change in time.
New Operation Mode
Normal StandAlone-secondary StandAlone-primary
Partner Node
Operation Mode
Normal
StandAlone
StandAlone-secondary
StandAlone-primary
StandAlone-synch
Sutting down
Node Out of Service
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Example Scenario:
G Retrieve the status of the Nodes.
G After half hour the user decides to change the level of a Node to 3.
G Assistant checks against the status it retrieved half hour back and let the user execute with-
out any warning.
G Before the command is executed assume the level of partner node has changed to state
that do not handle any calls for various reasons.
G This condition will cause a total outage i.e. loss of call processing.
To avoid this CMP must grey out the four operations on the screen after 1 minute if no action is
taken forcing the user to refresh the screen with the current status before an action is taken.
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6.2.5 SRXCTRL
Additional the srxctrl command has been enhanced and provides now the additional opti-
ons:
Restart srxctrl -r <2|3|4>
Power Down srxctrl -p
Change Operational Mode srxctrl -o <Normal|Standalone_primary|Standalone_secondary>
All the operations that are performed via srxctrl will be applied only to the local node. Prior to
V6 srxctrl could only be executed from a Root login. Since this command has to be executed
by the CMP in V6, srxctrl can now be executed from srx login as well.
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51
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7 Activate and Synchronize OSV partitions via CMP
7.1 Overview
The application scenarios introduced with this feature:
G Manage the OSV active / fallback versions via CMP
the versions that are installed in the system which is the active and which is the backup
version
G Quick and Easy Repair of the OSV System after an RU failure
If the downtime is allowed, the administrator can use the easy and quick way by activating
the backup version. The backup version is becoming active now and the call processing is
available again in a few minutes. If the administrator wants to completely repair the system
he can synchronize the two versions after the activation.
7.2 Functional Description
The Multi Partition Capability introduced together with the image based installation in V3.1
SMR2 provides the capability to hold two different SW versions on the same system / harddisc.
This allows an easier way of upgrading the system by keeping the previous version on the same
harddisc.
Primary Partition
Secondary Partition
(e.g. V4R1)
(e.g. V5)
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The sync8k command provides the capability to copy / mirror the content of the active partition
to the fallback partition. For e.g., this can be used to make a copy of the current software load
before installing a patchset or MOP
But the sync8k command had to invoked via the CLI. This feature allows now easely to syn-
chronize the partitions via the CMP.
Primary
Secondary
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(Fallback)
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(Active)
Primary is active partition Secondary is active partition
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7.3 Usage
7.3.1 Activate Backup version and Synchronize Active version to
Backup
The administrator navigates to OpenScape Voice dashboard.
Maintenance Inventory Modes
Click on your node.
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The Active and the Backup version are displayed. Additionally, two new functionalities are offe-
red so the administrator is able to Activate the Backup version or Synchronize the Active versi-
on to Backup.
Dashboard Synchronize Option:
The option to synchronize the Active version to Backup version, has been added to the CMP.
This is using the CLI command sync8k, in prior versions this was only a CLI option.
The administrator selects to synchronize the two versions. Since the existing data in the backup
version will be overwritten by the data of the active version the administrator has to verify this
selection.
Content of Active / Backup partition Synchronize partitions
or
switch to backup version (requires reboot)
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Synchronize Example:
>
This procedure cannot be canceled via CMP.
1.
2. 3.
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7.3.2 Activate backup version
The has been enhanced in order to provide the information about the active and the backup
version of the OSV system.
Once the administrator selects to activate the backup version of the OSV system, CMP will
login to the OSV system as srx user and trigger the activate8k to switch from the active to the
fallback and reboot the OSV.
In case of a problem, CMP gets the error message from OSV and informs the user.
The activation of the backup version is followed by a system reboot. Once the system reboot is
initiated the communication between CMP and OSV is lost, so CMP cannot monitor this job
anymore.
State description of the procedure Activate Backup Version
State 1: Initial State
a) Event: Activate
Action(s): A warning pop-up is created
Next State: Wait for user's verification
State 2: Wait for user's verification
a) Event: OK
Action(s): Call MMGR to start a backup version Activation Job
Next State: Start Activation
b) Event: Cancel
Action(s): The warning pop-up closes
Next State: Initial State
State 3: Start Activation
a) Event: MMGR response: OK
Action(s): Get the Job Id and inform user
Next State: Reboot System
b) Event: MMGR response: Failure
Action(s): Inform User
Next State: Wait for user's input: [Close]
(close the window and return to the initial state)
State 4: Reboot System End of the procedure
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Dialog box display from the Activate button
7
This will cause a full system outage !
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7.3.3 Software Activation enhancements
This feature introduces changes in CMP at the start and at the end of software activation pro-
cedure.
Initially the user has the option to perform versions synchronization or not.
If the user selects not to perform versions synchronization, CMP initiates the software activation
as in previous versions.
If the user selects to perform versions synchronization (default value), CMP shall initiate versi-
ons synchronization on both nodes. Once this job is completed successfully, CMP continues
with the software activation as it has in previous releases.
If the Rolling Upgrade is completed with no error, CMP handles the Software Activation as it
currently does. Currently if an error occurs the user is prompted to contact next level of techni-
cal support. This feature introduces an additional option in case of a failure.
The user has the capability to repair the system quickly by activating the backup software ver-
sion. Once the user verifies his selection, MCP initiates backup version activation (aka
activate8k ).
Additionally, CMP shall offer the option to download log files. When this option is selected, CMP
shall login to the system as srx user and trigger the script spt_log_collector.sh. Then CMP
transfers via FTP the log files and the user can save the log files locally to his laptop.
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If there is any problems during patching, it is recommended to download the log files so devel-
opment can locate the source of the problem.
The administrator selects the option Download Logs and the pop-up to save the log files locally
in his laptop is presented. The administrator saves the log files and is ready to proceed with the
fix of the Rolling Upgrade Failure.

He has two options:

G the existing one is to escalate the problem directly to next level of technical support. This
option is applicable if no downtime is allowed and the administrator has enough time to de-
bug the issue.
G the new one is the Quick Repair. This option will activate the Backup version of the Open-
Scape Voice system and the system will be rebooted. This option is applicable if the down-
time is possible, the administrator has no time to debug the issue and it is very important
to have the system back to level 4 as soon as possible.
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OSV Assistants additional checks
When one of the Jobs, version synchronization, backup version activation, software activation,
is in progress OSV Assistant will not allow the initiation of another job.
E.g. When a Software Activation is in progress, OSV Assistant shall prevent the user to start a
Version Synchronization or a Backup Version Activation.
The following table summarizes the additional checks that will be done from OSV Assistant:
User wants
to start
Job
In progress
Software
Activation
Version
Synchronization
Activate Backup
Version
Software Activation Get SW activation
status
Not allowed Not allowed
Version Synchronization Not allowed Not allowed
(covered by mmgr)
Not allowed
Activate Backup Version Not allowed Not allowed Not allowed
(covered by mmgr)
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8 Split Audit Functionality from
Registration Renewal
8.1 Overview
This Feature intends to split the audit functionality off from the Registration Renewal feature
and make it an automatic feature of OpenScape Voice.
Registration Renewal is a system-wide feature that when turned on, triggers OpenScape
Voice to extend a SIP subscriber's registration when OSV determines that it cannot communi-
cate to a branch office (identified by the subscriber and its dynamically associated endpoint).
This feature will now be turned on always. It was introduced in V2.2 and requires that the dy-
namically associated endpoint does not have the Do Not Audit endpoint attribute set (i.e. can
be audited).
While OSV cannot communicate with the branch office the SIP subscriber moves into the
Suspended state when its registration expires or when OSV tries to reach the subscriber. The
advantage of the Suspended state is that OSV will not attempt to contact the phone via the
WAN until the audit to the dynamically associated endpoint succeeds. If the Subscriber
Rerouting feature is enabled, Subscriber Rerouting will be attempted immediately - improving
the user friendliness of OSV.
Up to OSV V5 auditing of gateways is only done when the Registration Renewal feature is
turned on this was fine up to now, however by extending the audit mechanism to all SIP
endpoints and having this audit mechanism controlled by the Registration Renewal feature
does not make sense anymore.
At the same time the Audit mechanism is decoupled from the Alarming.
8.2 Functional Description
The current behavior of SIP endpoints alarming, the auditing mechanism and registration
renewal has been decoupled from each other.
OpenScape Voice will always alarm when it detects that it cannot communicate with a SIP
endpoint. The alarm will be cleared as soon as communication is reestablished.
The Reestablished State is now Depending on the state of the Do Not Audit endpoint attri-
bute.
Communication is considered reestablished:
G When Do Not Audit is disabled (i.e. auditing is enabled), after a successful audit to the SIP
endpoint.
G When Do Not Audit is enabled (i.e. auditing is disabled), as soon as a SIP request is re-
ceived from the SIP endpoint.
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The following operational states are defined for a SIP endpoint:
G Normal State:
The Normal operational state of a SIP endpoint indicates that communication with the end-
point is possible. In this state the SIP endpoint must have the Inaccessible endpoint at-
tribute disabled.
G Inaccessible state:
The Inaccessible state of a SIP endpoint indicates that a scheduled audit is running against
the SIP endpoint. In this state the SIP endpoint must have the Inaccessible endpoint at-
tribute enabled and the Do Not Audit attribute must be disabled. In this state only OP-
TIONS requests are sent to the endpoint.
G Communication Lost state:
The Communication Lost state indicates that OSV could not communicate with the SIP
endpoint. In this state no audit is scheduled or running against the endpoint. In this state
the SIP endpoint must have the Do Not Audit and Inaccessible endpoint attributes enabled.
In this state, no restrictions are imposed on sending messages to the endpoint, even
though they are expected to time out.
Normal
Inaccessible Communication Lost
Endpoint Operational States
-Inaccessible: false
-Do Not Audit: false
Normal
-Inaccessible: false
-Do Not Audit: true
Set Do Not Audit to true
Set Do Not Audit to false
-Inaccessible: true
-Do Not Audit: false
-Inaccessible: true
-Do Not Audit: true
Set Do Not Audit to true
(cancel scheduled audit +
all Suspended to Registered)
Set Do Not Audit to false
(+ on-demand + scheduled
audit)
C
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8.3 Provisioning
Pre V6 Registration Renewal feature setting was optional to be set by the administrator under:
OpenScape Voice Administration Signalling Managament SIP
The capability of disabling the Registration Renewal feature is removed from the SIP Signaling
Manager Administration in V6 and set now automatic by the OSV.
V5
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Application Scenario
The Registration Renewal feature has become an automatic feature of OpenScape Voice in V6
and the responsible attribute has been removed from the SIP Settings screen.
The audit parameters (quantity, interval and supervision time) can still be configured under:
OpenScape Voice Administration Signalling Managament SIP
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Per default the audit of a SIP Endpoint is enabled. By setting the Do Not Audit endpoint at-
tribute on a specific SIP Endpoint, the audit of that specific SIP Endpoint can be turned off.
>
Note that the Do Not Audit endpoint attribute should only be used for dummy
endpoints.
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The current behavior of SIP endpoints alarming, the auditing mechanism and registration re-
newal has been decoupled from each other.
OpenScape Voice will always alarm when it detects that it cannot communicate with a SIP end-
point.
The alarm will be cleared as soon as communication is reestablished.
The Reestablished State is now Depending on the state of the Do Not Audit endpoint
attribute.
Communication is considered reestablished:
G When Do Not Audit is disabled (i.e. auditing is enabled), after a successful audit to the
SIP endpoint.
G When Do Not Audit is enabled (i.e. auditing is disabled), as soon as a SIP request is re-
ceived from the SIP endpoint.
The CMP is showing the Inaccessible endpoint attribute and shows the operational states Nor-
mal, Communication Lost and Inaccessible in the endpoint overview:
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OpenScape Voice Siemens
9 Emergency Call SNMP Trap
9.1 Overview
This feature generates an SNMP trap on an emergency call and sends all the pertinent data to
the NOC in real-time for event notification.
The SNMP Trap contains:
Dialed Digits
Calling Party Number
Calling Party IP Address
LIN
LIN Description
Route Number (postfix digits)
9.2 Functional Description
Some customers asked to be notified in case an emergency call will be executed so that they
are able to send their own security personal to the emergency location. This feature extends
the existing Emergency Feature by sending an optional a SNMP trap to the configured recei-
vers when an emergency call will be placed.
In same cases, only one gateway is used but the destination number must be modified. In this
case, all destinations points to the same GW.
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9.3 Emergency Call Routing
The emergency call will be executed in two steps. In the 1st step, the call will be sent via the
PAC to a destination code which points to the emergency call service. This service adds spec-
ified information, e.g. LIN, based on the subscribers IP address to the call. The call goes now
again to the subscribers PNP PAC and can use a special routing entry.
In same cases, only one gateway is used but the destination number must be modified. In this
case, all destinations points to the same GW.
PAC
PNP
Dest Code DESTINATION A
ROUTE
Dest Code
E911 Definition
x.x.x.x
x.x.x.x 999
999
IP / Net. Route
DESTINATION B
ROUTE
GW_B GW_A
Dest Code
PAC
PNP
Dest Code DESTINATION A
ROUTE
Dest Code
E911 Definition
x.x.x.x
x.x.x.x 999
999
IP / Net. Route
DESTINATION B
ROUTE
GW
Dest Code
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9.4 Provisioning
9.4.1 Configure Emergency Calling
The already a few version ago implemented emergency calling feature is able to identify the
subscriber location based on his current IP address. This definition must be done under
Configuration OpenScape Voice Business Group General Emergency Calling
Subnet
The Emergency Calling Subnets definition assigns a route ID and a Location Identifier Number
(LIN) to the subscriber based on a single IP address or a entire IP subnet.
Subnet or Single IP
Netmask
1.21.11.0 /
255.255.255.0 defines
a class C network
or
1.21.11.200 /
255.255.255.255
defines a single IP
Use LIN as Calling Party Number
(must be defined on LIN tab)
Route Number max 4 digits (will be
added to original dialed number)
Passcode and ALI DB ID is current-
ly not used but mandatory parame-
ter (6 digits)
Description (Optional)
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For our Labs we replaced the dialed EmergencyNumber + RouteID with a routable E164 num-
ber in order to receive the subscriber in the PSAP Center.
1st, create a destination for emergency calls:
And save it.
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Open the previosly created destination again. On the Routes tab, add the gateway which
should be used.
Replace dialed Emer-
gency Number (1121)
with our Test Subscriber
and set correct NOA
(International).
Enter priority for this
route between 1 (high-
est) and 255 (lowest)
Enter / choose endpoint
(gateway)
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Add a prefix access code which forwards the emergency call to the Destination Code. e.g. the
emergency calling number, which is dialed from is subscriber is 112. In the 2nd step, the same
number extended with the route ID goes again into this PAC.
Emergency Number
Allows to dial plain Emergen-
cy Number as well with Route
ID with the same PAC
Forward number unmodified to
next level
Call Type is Off-net Access
and will be forwarded to the
Destination Code
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We need two different Destination Codes. The first code points to the service destination, which
is used to lookup the emergency calling information.
PAC (enter manually or press
to select) and Nature of Ad-
dress from previous step
Traffic Type Emergency
ignores call restrictions
Service Emergency looks up
Emergency Calling Subnet
Definitions
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For every Routing ID, which is defined in the emergency calling subnets, a destination code is
required to send the call to a gateway.
PAC followed by the definied
route ID(must be entered man-
ually) and Nature of Address
from previous step
Traffic Type Emergency
ignores call restrictions
Destination EmergencyMUC
will be used to send the call to
the PSAP
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9.4.2 Enabling E911 SNMP Traps
The feature will be enabled by changing the SRX Parameter
Srx/Service/E911/EmergencySNMPEvent
to RtpTrue. This can be done in the CMP under:
Configuration OpenScape Voice Administration General Settings RTP
Locate the System Parameter manual or via the Search for field and change the parameter
to True. The feature becomes be immediately active.
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If an emergency call will be initated, an SNMP trap will be sent to all configured SNMP receiv-
ers. This alarm will be shown in the CMP under
Maintenance Monitoring Alarms Active Alarms
with the Alarm ID 948:
By clicking on the Alarm ID, the alarm details will be shown on the Description tab:
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Here is an example of a sucessfull and a unsucessfull alarm:
Sucessfull emergency call
sysUpTime: 7 days, 18:17:43.26
rtpEvtCurrId: 30198
rtpEvtSet: 250
rtpEvtNum: 1
rtpEvtReportingObject: APPL:v6sys1n1.uce2
rtpEvtFaultyObject: APPL:v6sys1n1.Endpoint 498970071002 192.168.50.60
rtpEvtEndkey: noRecoveryDefined
rtpEvtAlarmType: service
rtpEvtShortText: FaultyObject = Endpoint 498970071002 192.168.50.60: Emergency Call
from endpoint DN 498970071002 to Emergency Number = 112.CPN=498970071002, BG=SEN,
IP=192.168.50.60, RT=89, AoR=498970071002, SubIP=192.168.50.60, SubPort=10001.
rtpClusterName: vmv6cluster
rtpEvtFilterLastTransId: 5450
hiQEmergencyCallIdTrap: The OSV server is processing an Emergency Call .
Unsucessfull emergency calll
sysUpTime: 7 days, 17:55:50.77
rtpEvtCurrId: 30124
rtpEvtSet: 250
rtpEvtNum: 1
rtpEvtReportingObject: APPL:v6sys1n1.uce1
rtpEvtFaultyObject: APPL:v6sys1n1.Endpoint 498970071002 192.168.50.60
rtpEvtEndkey: noRecoveryDefined
rtpEvtAlarmType: service
rtpEvtShortText: FaultyObject = Endpoint 498970071002 192.168.50.60: Emergency Call
from endpoint DN 498970071002 to Emergency Number = 112.CPN=498970071002, BG=SEN,
IP=192.168.50.60, RT=89, AoR=498970071002, SubIP=192.168.50.60, SubPort=10001,
Unsuccessful.
rtpClusterName: vmv6cluster
rtpEvtFilterLastTransId: 5429
hiQEmergencyCallIdTrap: The OSV server is processing an Emergency Call .
Legend
CPN = Calling Party Number
BG = Business Group
RT = Route Number (E911 definition)
AoR = Address of Record
SubIP = Subscriber IP address
SubPort = Subscriber Port Number (default 5060)
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10 SIPQ CodSet4 Transit Counter IE
10.1 Overview
The codeset 4 Transit Counter IE is an optional QSIG supported IE sent / received in a Q.931
SETUP message. It applies to both call-related and non-call related (i.e., CISC) connections.
Its purpose is to limit the number of transit nodes that a call setup request may be routed th-
rough. If the limit is exceeded, call clearing will occur. The IE provides a provisional level of con-
trol for loop prevention within the private network.
OSV already supports in previous versions incrementing this IE when received, but, not gene-
rating it or checking against a max. threshold value.
Typically, the maximum number of transit nodes allowed for a call is provisioned consistently at
each node in the network.
10.2 Functional Description
The codeset 4 Transit counter IE will be supported over SIPQ for call-related / non-call related
SETUP messages as follows:
Origination
Send a locking shift to codeset 4 and Transit counter IE with the count = 0.
Tandem
G If the transit count field has a value that is less than the maximum (provisional) limit, that
information element will be included in the SETUP message sent to the Subsequent PINX.
The value of the transit count field in the outgoing Transit counter information element will
be incremented by one (i.e., greater than the value received).
G If a Transit counter information element is not received, the Transit PINX will include a Tran-
sit counter information element in the SETUP message sent to the Subsequent node. The
value of the transit count field in this element will be set to a value of 1.
G if the Transit counter information element in which the transit count field has a value that is
greater than or equal to the maximum (provisional) limit of Transit PINX's through which the
call may be routed, the call will be rejected. The acceptable max. limit will not exceed 31.
A new RTP parameter Srx/Sip/SipQMaxTransitCount has been introduced, range 1
31, default = 5 to control this.
Terminating
Ignore the IE
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Example 1: Transit Counter not reached
Example 2: Transit Counter reached
HiPath 4000
(Origination)
HiPath 4000
(Terminating)
OSV OSV OSV
Tandem 1 Tandem 2 Tandem 3
QSIG SETUP
Srx/Sip/SipQMaxTransitCounter = 5
TC=0 TC=1 TC=2 TC=3
TC = Transit Counter
HiPath 4000
(Origination)
HiPath 4000
(Terminating)
OSV OSV OSV
Tandem 1 Tandem 2 Tandem 3
QSIG SETUP
QSIG PROGRESS / DISCONNECT / RELEASE
Srx/Sip/SipQMaxTransitCounter = 2
TC=0 TC=1 TC=2 TC=3
Cause = 1
In-band tone (e.g. fast busy)
TC = Transit Counter
CorNet-N-Argument:
codeSet0Data: 0802E0810C0780313733353835740200C0 {.......173585t...}
InfoElem: Cause (H'8)
Length: 2 (H'2)
CodingStandard: Standard specific to identified location
Location: User
CauseValue: Loop/satellite Counter exceeded
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10.3 Provisioning
To check / change the max. Tansit Counter parameter, navigate to
Configuration OpenScape Voice Administration General Settings RTP
and Lookup the parameter Srx/Sip/SipQMaxTransitCount in the list or use the search function.
A value can be choosen between 1 and 31 (default 5).
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In addition, the endpoint profile allows to choose if a announcement will be played in case a call
will be blocked during the transit counter parameter:
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11 Best Effort SRTP for SDES
11.1 Overview
Support for SDES (SDP Security Descriptions) in addition to what has been implemented for
MIKEY a few versions ago.
The best effort SRTP mechanism defined in the OSCAR Signaling and Payload Encryption
Specification follows a standards based approach and allows for backward compatibility.
There are defined provisions for a SIP end point to reject the SRTP offer if it does not support
this mechanism.
But certain 3rd party end points (that do not clearly follow the SIP standards / specifications)
fail in the SDP negotiation and as a result the call setup fails.
With this feature, OSV will ensure SDP compatibility between devices that support best effort
SRTP and devices that do not by allowing the call to be established with RTP instead of failing,
also for SDES.
This change is essential for ensuring basic interworking with devices that do not support the
SDES SDP offer.
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PSTN
SIP
SIP
SIP/SIPQ
SIPQ
vHG 3550
vHG 3550
SIPQ
Mediatrix
(GateWay)
(SDES Support)
OSCV V6 - B
(SDES Support)
SIP Openstage
Subscribers
Branch
OpenBranch
Proxy/SBC
SIP Openstage Subscribers
(SDES Support)
Main Office
Personal
Edition
(SDES Support)
Xpressions
(SDES Support)
Offboard UC
server
HMS
(SDES Support)
Solution Configuration
Testing SDES
Audiocodes Median
3000
(SDES Support)
HFA Openstage
Subscribers
RG8700
(GateWay)
(SDES Support)
OSCV V6 - A
(SDES Support)
(SDES Support)
(SDES Support)
HiPath 4000
V6R1 - B
LAN:192.168.XXX.
XX
SDES Support
No SDES Support
SIP Subscribers
(No SDES Support)
Main Office
Xpressions
(SDES Support)
vHG 3550
HiPath 4000
V6R1 - A
HFA Openstage
Subscribers
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11.2 Functional Description
Siemens SIP devices, e.g. OpenStage phones, support the establishment of secure calls via
SRTP, by exchanging encryption keys during call setup. The mechanism for exchanging the
keys can be either MIKEY (at the session or the media level) or SDES (at the media level), but
not both at the same time.
However, other applications such as the HMS, may send an SDP offer with both keys SDES
and MIKEY included in the SDP offer.
Currently, if a device that supports SRTP with SDES initiates a call to a device that does not
support the best effort SRTP mechanism, the call will fail as shown below.
A similar situation may occur when the terminating side is behind a non-compliant SIP server
such as a Cisco Gateway, OpenSER, Asterisk, Juniper SBCs etc even if the device itself sup-
ports OSCAR Best Effort SRTP.
OpenScape Voice will support both key exchange mechanisms, even if they happen to co-exist
in the same SDP media as is the case for example with OMS.
4
Helen calls Peter requesti ng
a secure call wit h SDES
using here best- effort SRTP
compliant device
The call i s offered to Peter
Helen
OpenScape Voice
1
2
3
Peter
Call setup fails
415 Unsuppor ted media type
415 Unsupported medi a type
Peters device doesnt
support SRTP and
rejects the call
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When OSV receives a request for a secure call using SDES and the terminating side does not
support best effort SRTP, it will remove any security type token from the X-Siemens-Call-Type
header, change the o-line to a new value, re-insert the SRTP m-line and set the port to 0 before
sending it to the originating side. This will ensure that the terminating side shall not reject the
call but shall accept it in unsecured mode, as shown above. The same concept (completely re-
move SRTP media than just set the port to 0) will also be applied for MIKEY.
RTP
5
Helen calls Peter requesti ng
a secure call wit h SDES
using her best-effort SRTP
compliant device
Unsecure call is
offered to Peter
Helen
OpenScape Voice
1
3
4
Peter
Unsecure call is
establi shed
200 OK
Peter s device accepts
the unsecure call
2
OSV detects that Peters
device doesnt support
best-effort SRTP wi th
SDES and removes
SRTP media
200 OK
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SIP-SM logic
The new SIP-SM logic is shown in the following figure:
SDP offer arrives
Remove key
Via proxy and
use proxy SRTP
settings
SDES or MIKEY
key present
Continue
Remove SRTP
media
No
Yes
Yes
No
Two m-lines
RTP+SRTP present
For each contact
Yes
For SDES, MIKEY if feature is
turned on
No
Contact
SRTP support
Proxy SRTP
Support
Off Off
On On
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The logic of the SIP registrar is shown below.

Registration
User-Agent
present
Set contact SRTP
support with
SDES to Off

User-Agent
belongs to list of
non-supporting
devices
Continue
NO
YES
YES
NO
YES
NO
Set contact SRTP
support with
SDES to On

Feature
turned on
Best Effort
SRTP support
AUTO
OFF ON
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Call flow on a non secure call
Call flow on a non secure call
Best-effort
SRTP
compliant
device OSV
INVITE (ST-secure)
SDP
o= MxSIP 0 405720119 IN IP4 10.23.15.13
m= audio 5004 RTP/AVP
m= audio 5006 RTP/SAVP
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:d0Rm
Best-effort SRTP
non-compliant
device
100 Tryi ng
100 Trying
180 Trying 180 Tryi ng
INVITE
SDP
o= MxSIP 0 405720119 IN IP4 10.23.15.13
m= audio 5004 RTP/AVP
200 OK
o= MxSIP 0 1896657476 IN IP4 10.23.15.16
SDP
m= audio 5004 RTP/AVP
200 OK (ST-insecure)
SDP
o= OSCV 5246657476 5246657476 IN IP4 0.0.0.0
m= audio 5004 RTP/AVP
m= audio 0 RTP/SAVP
RTP
Best-effort
SRTP
compliant
device OSV
INVITE (ST-secure)
SDP
o= MxSIP 0 405720119 IN IP4 10.23.15.13
m= audio 5004 RTP/AVP
m= audio 5006 RTP/SAVP
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:d0Rm
Best-effort SRTP
non-compliant
device
100 Tryi ng
100 Trying
180 Trying 180 Tryi ng
INVITE
SDP
o= MxSIP 0 405720119 IN IP4 10.23.15.13
m= audio 5004 RTP/AVP
200 OK
o= MxSIP 0 1896657476 IN IP4 10.23.15.16
SDP
m= audio 5004 RTP/AVP
200 OK (ST-insecure)
SDP
o= OSCV 5246657476 5246657476 IN IP4 0.0.0.0
m= audio 5004 RTP/AVP
m= audio 0 RTP/SAVP
RTP
Best-effort SRTP
compliant device
OSV
INVITE (ST-secure)
SDP
o= MxSIP 0 405720119 IN IP4 10.23.15.13
m= audi o 5004 RTP/AVP
m= audi o 5006 RTP/SAVP
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:d0Rm
Best-effort SRTP non-
compliant device
100 Tryi ng
100 Tryi ng
180 Tryi ng 180 Tryi ng
INVITE
SDP
o= MxSIP 0 405720119 IN IP4 10.23.15.13
m= audio 5004 RTP/AVP
200 OK
o= MxSIP 0 1896657476 IN IP4 10.23.15.16
SDP
m= audio 5004 RTP/AVP
200 OK (ST-i nsecure)
SDP
o= MxSIP 0 1896657476 IN IP4 10.23.15.16
m= audio 5004 RTP/AVP
m= audio 0 RTP/SAVP
Best-effort SRTP
compliant device
OSV
INVITE (ST-secure)
SDP
o= MxSIP 0 405720119 IN IP4 10.23.15.13
m= audi o 5004 RTP/AVP
m= audi o 5006 RTP/SAVP
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:d0Rm
Best-effort SRTP non-
compliant device
100 Tryi ng
100 Tryi ng
180 Tryi ng 180 Tryi ng
INVITE
SDP
o= MxSIP 0 405720119 IN IP4 10.23.15.13
m= audio 5004 RTP/AVP
200 OK
o= MxSIP 0 1896657476 IN IP4 10.23.15.16
SDP
m= audio 5004 RTP/AVP
200 OK (ST-i nsecure)
SDP
o= MxSIP 0 1896657476 IN IP4 10.23.15.16
m= audio 5004 RTP/AVP
m= audio 0 RTP/SAVP
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11.3 Provisioning
The CMP allows now to control the Best Effort SRTP for either MIKEY, SDES or both under:
Configuration OpenScape Voice Administration Signalling Management SIP
Possible Best-Effort SRTP determination values:
MIKEY,SDES turns on Best-Effort SRTP interworking with MIKEY and SDES
MIKEY turns on Best-Effort SRTP interworking with MIKEY only
SDES turns on Best-Effort SRTP interworking with SDES only
None turns off both features (OSV will not impact the SRTP offer)
V6
V5
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Additional, each blacklisted device identifier allows now to choose with the new introduced Key
Exchange Mechanism parameter the supported SRTP key method:
After turning on the Best Effort SRTP feature, in this example the device identifier list contains
the following devices (default excepting the last entry):
In this example, every other device can use MIKEY and SDES as long the Best Effort SRTP
support attribute is set to Automatic (see next page).
Device Identifiert Key Exchange Mechamism Comment
Cisco MIKEY allows only MIKEY for this device
CSCO MIKEY allows only MIKEY for this device
Asterisk MIKEY allows only MIKEY for this device
X-Lite MIKEY allows only MIKEY for this device
XYZ device SDES allows only SDES for this device
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The OpenScape Voice CMP will also provide a field to indicate / enable support for the mech-
anism on a per SIP endpoint and subscriber basis. For the subscriber settings, navigate to:
Configuration OpenScape Voice Business Group Members Subscribers
And choose the subscriber which should be changed. On the Security tab, the Best Effort
SRTP support can be configured:
The following values can be choosen in the drop-down box:
MIKEY,SDES both MIKEY and SDES will be passed to this subscriber
MIKEY only MIKEY will be passed to this subscriber
SDES only SDES will be passed to this subscriber
Disabled SRTP (MIKEY + SDES) will be removed to this subscriber.
In this case, enryption is not possible !
Automatic Use the SIP Signalling black-list settings
For static registered subscribers the default value is MIKEY,SDES and for dynamic registered
subscribers Automatic.
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Also the SIP endpoint gots the new Best Effort SRTP support settings as the Subscribers on
the SIP tab:
The following values can be choosen in the drop-down box:
MIKEY,SDES both MIKEY and SDES will be passed to this endpoint
(Default for static registered endpoints)
MIKEY only MIKEY will be passed to this endpoint
SDES only SDES will be passed to this endpoint
Disabled SRTP (MIKEY + SDES) will be removed to this endpoint.
In this case, enryption is not possible !
Automatic Use the SIP Signalling black-list settings
(This is only available for dynamic registered endpoints and the
default settings)
For SIP-Q enabled enpoints the Best Effort SRTP support is fixed set to MIKEY,SDES. For
static registered endpoints the default value is MIKEY,SDES and for dynamic registered end-
points Automatic.
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And last, for the SBC / Proxy endpoints (SIP proxy attribute set):
The new Use Proxy SRTP settings for calls to subscribers allows the provisioning of the Best
Effort SRTP support method on the SBC / Proxy for all to this device connectec subscribers:
Use endpoint settings for subscribers if checked
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Phone Configuration
OpenStage devices with firmware version 3 supports enabling SDES.
To change the SRTP type to SDES, navigate on the Administrator Pages to
System Security System
Then the last item on the Open Stage to enable SDES status and ranking are available under
System Security SDES config
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Do not forgett to change the transport protocol to TLS under System SIP interface
And the SIP server and registrar port number to 5061 under Network Port configuration
Optional the SIP local port can be set to 5061 as well.
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MIKEY example
Legend
AVP = Audio Video Protocol
SAVP = Secure Audio Video Protocol
Content-Type: application/sdp
Content-Length: 896
v=0
o=MxSIP 0 973987686 IN IP4 10.237.0.232
s=SIP Call
c=IN IP4 10.237.0.232
t=0 0
a=key-mgmt:mikey AQAVgKJWG5ECAAAAAAAAAAAAAAA8/
L4BAAAAAAUBAAVtaWtleQsA0aTwmAzMzAAKFLpUkKhkD0JUmezrtMGBZfyfcNjzAQAAADYCAQEDB
AAAAKAEBAAAAHALBAAAAFAAAQEBBAAAAIAJAQAGAQAFAQAIAQEKAQEHAQEMBAAAAAAAAAAkABAAE
HwTPcrYcYpjczOmqw7bahgADpWgFtVwI74qzmaPAUGFAA==
m=audio 5004 RTP/AVP 0 8 9 18 4 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:101 telephone-event/8000
a=silenceSupp:off - - - -
a=fmtp:4 annexa=no
a=fmtp:18 annexb=no
a=fmtp:101 0-15
a=ptime:20
m=audio 5006 RTP/SAVP 0 8 9 18 4 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:101 telephone-event/8000
a=silenceSupp:off - - - -
a=fmtp:4 annexa=no
a=fmtp:18 annexb=no
a=fmtp:101 0-15
a=ptime:20
Port 5004
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Port 5006
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SDES example
Legend
AVP = Audio Video Protocol
SAVP = Secure Audio Video Protocol
Content-Type: application/sdp; charset=ISO-10646
Content-Length: 700
v=0
o=OpenStage-Line_1 418796685 362346688 IN IP4 10.237.1.111
s=SIP Call
c=IN IP4 10.237.1.111
t=0 0
m=audio 5012 RTP/SAVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=silenceSupp:off - - - -
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:PlwuZFT/isWMGTNFTlN8spGfxqzcbTG5enZGbjdB
a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:ZdhmOaKn1jvGLvRIS2RuM7QaypQs9vztAmyjw3rT
a=fmtp:18 annexb=no
a=fmtp:101 0-15
a=sendrecv
m=audio 5010 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=silenceSupp:off - - - -
a=fmtp:18 annexb=no
a=fmtp:101 0-15
a=sendrecv
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SRTP Key
Port 5012
Port 5010
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12 Support for multiple prime line registration
12.1 Overview
The goal of this feature is to allow a keyset line to register as the primary line in more than one
device at the same time.
Behavior up to V5:
G Primary line P1 is registered in device D1
G Device D2 attempts to register P1
G System will first unregister P1 from D1, then register P1 in D2
In case that a keyset subscriber wants to use additional phones for his primary line, each de-
vice needs a own primary line number with the main number configured as a secondary line:
Keyset A
Primary Line: 1001
Main Phone (D1)
Keyset B
Primary Line: 1002
Secondary Line: 1001
Sofa (D2) Conference table
Keyset C
Primary Line: 1003
Secondary Line: 1001
Keyset D
Primary Line: 1000
Secondary Line: 1001
Executive Assistant
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Behavior after this features is implemented in V6:
G Primary line P1 is registered in device D1
G Device D2 attempts to register P1
G System register P1 in D2, keeping it register in D1 as well
(similar to a multiple contact scenario)
So the OpenScape Voice allows now multiple registrations for the same primary line and the
same number can be used for multiple devices:
Keyset A
Primary Line: 1001
Main Phone (D1) Sofa (D2) Conference table
Keyset D
Primary Line: 1000
Secondary Line: 1001
Executive Assistant
Keyset B
Primary Line: 1001
Keyset C
Primary Line: 1001
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12.2 Application secanrio
The current implementation enforces a limit of one registered primary line for keyset devices.
If another device (Keyset B) registers the same primary line, the previous device (Keyset A)
with the existing prime line registration is automatically unregistered (in the system database -
the phone (Keyset A) doesn't get any message when that happens). Additionally, all secondary
lines registered for that primary line (Keyset A) are also automatically unregistered.
Once all the lines on that device (Keyset A) are unregistered, the new registration for Keyset B
is processed.
In other words, the primary line of a keyset number can only be present in one device at one
time.
Some customers, like Coke, want to have the same primary line accessible at several places
simultaneously.
Example of how it will be used:
1. Executive answers call using the desk phone
2. Executive puts the call on hold on the desk phone
3. Executive goes to the conference room, where another phone with the same primary
line is configured
4. Executive retrieves the call on the conference room phone and continues talking
This could have been solved by configuring the executive's phone number as a secondary line
appearance of the conference room phone.
However, it requires a DN to be used as the primary line for the conference room phone. This
costs an extra license, which customers are not willing to pay. From their perspective, this is
the same phone, being used from different places (much like a multiple contact phone being
registered from two places - should consume only one license in the system).
12.3 Functional description
OSCV will allow the registration of a DN as the primary line of more than one device simulta-
neously once this feature is implemented. The existing enforcement of "DN as primary line in
one device only" will be removed.
Once the enforcement is removed, the keyset DN will work similarly to multiple contacts: the
same DN may be used as the main DN in several devices.
Some feature interactions for keysets will also be adjusted to match the ones for multiple con-
tacts as close as possible.
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12.4 Provisioning
The feature can be enabled by changing the RTP parameter Srx/Xdm/Registrar/EnableMulti-
Primary to RtpTrue. This can be done via the startCli tool on the OpenScape Voice node:
sysad@kn200master:[/home/sysad] #64
$
# startCli
RTP Command Line Interface
Copyright (C) Fujitsu Siemens Computers GmbH 1999 - 2009
All Rights Reserved
Copyright (C) 2000-2008 Siemens Network Convergence LLC.
All Rights Reserved.
NOTE: You may use this software only in accordance with the terms of your
license agreement, located on any of the installation CDs for this product.
(This process is running as "RtpAdmCli01")
Management session established for user "sysad" on "kn200master"
Main Menu:
Configuration Management.......................1
Fault Management...............................2
Performance Management.........................3
Security Management............................4
System Management..............................5
Application-level Management...................6
Open Logfile............................94
Show Callback Output....................95
Wait for Callbacks......................96
Change Password.........................97
Expert Mode.............................98
Exit....................................99
Selection: 1
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Configuration Management:
Configuration Parameters.......................1
Logging........................................2
Return..................................99
Selection: 1
Configuration Parameters (methods):
browseParameterNames...........................1
getParameterInfo...............................2
modifyParameter................................3
createParameter................................4
deleteParameter................................5
loadConfigVersion..............................6
removeConfigVersion............................7
getAllSavedConfigVersions......................8
saveCurrentConfigVersion.......................9
Return..................................99
Selection: 3
modifyParameter:
name : Srx/Xdm/Registrar/EnableMultiPrimary
invariant settings:
name : Srx/Xdm/Registrar/EnableMultiPrimary
type : PARM_BOOL
usage : PARM_USAGE_CUSTOMER
lastUpdateMillis : 18.Jul.2011 13:35:43h (000 msec)
changeId : 0
descriptionString:
modifying variable parameters:
current value: RtpFalse
value <max length: 2047>: RtpTrue
enable
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13 PIN for private / business calls
13.1 Overview
The ability to make a 'private call', i.e. make a call for which the charges are personally alloca-
ted to the user who makes the call, shall be supported via the BG authorization code feature.
For this purpose, the authorization code shall be enhanced to optionally include the subscriber
number to whom it belongs as well as the type 'private' or 'business'.
A user shall be able to make a 'private' or 'business' call from any device that has the 'BG au-
thorization code' feature assigned using their 'private' or 'business' authorization codes respec-
tively. In both cases the subscriber number to whom the authorization code belongs to shall be
written in the 'paying party' field of the CDR, causing the billing application to charge the call to
this user. In case of a 'private' call, a 'private call' flag shall also be set in the CDR. If The billing
application detects the 'private call' flag it shall mark the call as 'private', otherwise it shall mark
it as 'business'. The customer may then choose to assign the costs of 'private' calls personally
to the user making these calls.
For making 'project' calls, the user shall use the 'BG account code' feature to enter an account
(project) code which shall be written in the CDR. If the billing application matches the account
code in the CDR to a 'project' then this call shall be marked as 'project' call and the call shall
be assigned to this project. The assignment of PINs to 'projects' and the creation of the project/
pins table in the The billing application DB is out of the scope of this FRN.
In addition, call transfer service and ONS shall be enhanced to carry over the BG account code
and BG authorization code from the original CDR to the call transfer CDR and the ONS CDRs
in order to provide correct accounting for these scenarios as well.
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13.2 Functional Description
It shall be possible to associate a subscriber DN and a type ('private' or 'business') with an au-
thorization code. The Assistant page under "Business Group BG Options Authorization
Codes Add" shall be enhanced to allow the administrator to optionally select a subscriber
DN to whom this authorization code shall belong to as well as the type of the authorization code
('private' or 'business'). If a subscriber DN is selected then it shall not be possible to enter a
range, i.e. the 'End' box shall be grayed out; and vice versa if the 'End' box has a value (range
selected) it shall not be possible to select a subscriber DN. Option 'type' should not show at all
if a subscriber DN has not been selected. The default value for the 'Type' field shall be 'busi-
ness'.
The Assistant page under "Business Group BG Options Authorization Codes" where the
authorization codes are displayed shall be enhanced to also show the subscriber DN and type.
The administrator shall be able to 'edit' any field of an authorization code: auth code, subscriber
DN or type. The administrator shall be able to sort/filter per subscriber DN and type.
The Assistant page under "Business Group Members Subscribers Subscriber Screen
Security Tab" shall be enhanced to also display the 'Authorization code for Private calls' and
'Authorization code for Business calls' of this subscriber. The administrator shall be able to di-
rectly 'edit' these fields. If the subscriber already has an authorization code then saving the sub-
scriber info shall result in the update of the authorization code for this subscriber. If the subscri-
ber does not have an authorization code then saving the subscriber info shall result in the
creation of the authorization code for this subscriber. In both cases, if the 'saving' of the autho-
rization code fails because the authorization code already exists an appropriate error message
shall be displayed. If the authorization code entered is invalid, e.g. non-digit characters are
used; invalid length etc, then the operation shall fail with an appropriate error message.
FN9821FNUTFN_BABZZZEDDZ
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Siemens Enterprise Communications GmbH & Co. KG 2011. All rights reserved.
Siemens Enterprise Communications GmbH & Co. KG is a trademark licensee of Siemens AG
111
OpenScape Voice Siemens
Application Scenario 1 - Private Call
Application Scenario 2 - Business Call
22
Helen uses the phone in
the lobby to call Peter on
his cell phone. Shes
prompted to enter an
authorization code. Helen
enters her private
authorization code.
After a while Helen
and Peter hang up
CDR with paying
party=Helens number
and private call flag is
written.
Helen
OpenScape Voice OpenScape Voice
44
11
33
RG8700 RG8700
Peter answers
Billing Application Billing Application
Billing Application
charges the call
to Helen
55
22
Helen uses her colleague
Marks phone to call a
customer. Shes prompted
to enter an authorization
code. Helen enters her
business authorization
code.
After a while Helen
and customer hang
up
CDR with paying
party=Helens number
is written.
Helen
OpenScape Voice OpenScape Voice
44
11
33
RG8700 RG8700
Customer
answers
Billing Application Billing Application
Billing Application
marks the call as
business and assigns
it to Helen.
55
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Siemens Enterprise Communications GmbH & Co. KG 2011. All rights reserved.
Siemens Enterprise Communications GmbH & Co. KG is a trademark licensee of Siemens AG
112
OpenScape Voice Siemens
13.3 Provisioning
That the Authorization Code features will be logged later to the CDR, the Message Detail Re-
cord must be enabled on the General tab of the Business Group.
Configuration OpenScape Voice Business Group List
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Siemens Enterprise Communications GmbH & Co. KG 2011. All rights reserved.
Siemens Enterprise Communications GmbH & Co. KG is a trademark licensee of Siemens AG
113
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The following PAC must be created to activate the authorization code feature from the sub-
scriber.
Configuration OpenScape Voice Business Group Translation Prefix Access Code
Default for Authorization Code
Feature Access Code (3 Digits)
must be entered together with the
subscriber DN (30 Digits) on which
the feature should be executed.
Remove FAC (always)
Prefix Type is Vertical Service to
activate a feature.
Nature of Address is Unknown
Destination Type Service
Select feature name
Authorization Code
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Siemens Enterprise Communications GmbH & Co. KG 2011. All rights reserved.
Siemens Enterprise Communications GmbH & Co. KG is a trademark licensee of Siemens AG
114
OpenScape Voice Siemens
Next, the authorization code feature must be assigned to the subscriber either via the feature
profile or directly on the subscribers Feature tab.
Configuration OpenScape Voice Business Group Members Subscribers
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Siemens Enterprise Communications GmbH & Co. KG 2011. All rights reserved.
Siemens Enterprise Communications GmbH & Co. KG is a trademark licensee of Siemens AG
115
OpenScape Voice Siemens
The authorization codes can be provisioned in OSV either via the Assistant subscriber screen
under
Configuration OpenScape Voice Business Group Members Subscribers
Enter th Business and / or Private calls PIN which should be defined for this subscriber.
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Siemens Enterprise Communications GmbH & Co. KG 2011. All rights reserved.
Siemens Enterprise Communications GmbH & Co. KG is a trademark licensee of Siemens AG
116
OpenScape Voice Siemens
or via the Assistant authorization codes screen under
Configuration OpenScape Voice Business Group BG Options Authorization Co-
des
or via mass provisioning.
All authorization codes must be unique within a specific BG, i.e. two subscribers within the
same BG cannot have the same personal authorization code and a subscriber cannot have the
same authorization code for both 'business' and 'private' calls.
13.3.1 CDR
The OpenScape VoiceBilling and Usage Collection software is enhanced to provide and sup-
port the generation of the new additional data needed by this feature. The CDR software is im-
pacted to add two new values in existing field 106 (Per Call Feature Extension):
G 1048576 = 'Private call'
G 2097152 = 'Business call'
>
The Authorization Code (PIN) will not be logged in the CDR for security reasons.
Enter the PIN code in the
Start field.
Select the subscriber DN to
which the PIN should be
assigned with ...
and the PIN code type
Business or Private.
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Siemens Enterprise Communications GmbH & Co. KG 2011. All rights reserved.
Siemens Enterprise Communications GmbH & Co. KG is a trademark licensee of Siemens AG
117
OpenScape Voice Siemens
Example CDR record created with the cdrdecode tool
01. Sequential Record Number: 0
02. Record Type: 00000000 (Standard CDR)
03. Start Time: 2011-04-08 T 06:03:59.3+0200
04. Duration of Call (Tenths of seconds): 0
05. Switch ID: vmv6node1
06. Record ID: 2011-04-08T06:03:59.3+0200:FF000100000000004F363A4E73500000
09. Call Segment Duration (Tenths of seconds): 0
11. Term Number/Called Party: 491234567890
12. Orig Number/Calling Party: 498970071001
13. ANI: 498970071001
16. Call Type: 0 (Voice Call)
18. Attempt Indicator: 4 (Calling party aborted prior to answer)
19. Release Cause/Completion Indicator: 16 (normal call clearing)
20. Bearer Capability Request: 0 (Circuit mode speech)
23. Operator Indicator: 0 (Direct Dialed)
26. Originating Nature of Address: 0 (Voice Call)
29. Route Selection: 0 (Standard)
30. Ingress Signaling Type: 17 (SIP)
31. Egress Signaling Type: 17 (SIP)
38. Incoming Trunk Group Signaling Type: 5 (Not Provisioned)
39. Outgoing Trunk Group Signaling Type: 5 (Not Provisioned)
40. Originating Party Identifier: 900 (Originating Endpoint on the switch)
41. Terminating Party Identifier: 903 (Terminating Endpoint not on the switch)
42. JIP Source Indicator: 5 (Unknown)
43. LRN Source Indicator: 5 (Unknown)
46. Originating Party Query Status Indicator: 9 (No query done)
47. Terminating Party Query Status Indicator: 9 (No query done)
49. Call Release Time: 2011-04-08 T 06:04:14.0+0200
50. Incoming Carrier Connect Time: 2011-04-08 T 06:03:59.3+0200
51. Incoming Carrier Release Time: 2011-04-08 T 06:04:14.0+0200
52. Outgoing Carrier Connect Time: 2011-04-08 T 06:04:13.2+0200
53. Outgoing Carrier Release Time: 2011-04-08 T 06:04:14.1+0200
54. Dialing and Presubscription Indicator: 0 (No IC/INC involved in call)
55. Calling Party Subaddress Delivery: 1 (Feature used, presumed delivered)
56. Called Party Subaddress Delivery: 1 (Feature used, presumed delivered)
57. Low-Layer Compatibility Information Delivery: 1 (Feature used, presumed delivered)
58. High-Layer Compatibility Information Delivery: 1 (Feature used, presumed delivered)
59. User-to-User Information/Fast Select: 1 (Feature used, presumed delivered)
63. Service Feature Codes: 0 (No Services)
64. Per Call Feature: 0 (No Features)
68. Originating Side: TOS Used: 0
69. Originating Side: Reservation Used: 0 (Default)
70. Originating: Packets Sent: 0
71. Originating: Octets Sent: 0
72. Originating: Packets Rcvd: 0
73. Originating: Octets Rcvd: 0
74. Originating: Packets Lost: 0
75. Originating: Inter-arrival Jitter (milliseconds): 0
76. Originating: Average Transmission Delay (milliseconds): 0
77. Terminating Side: Codec Used: 268435457 (G.711 64k PCM a-law, Telephone Event(Other))
78. Terminating Side: TOS Used: 0
79. Terminating Side: Reservation Used: 0 (Default)
80. Terminating: Packets Sent: 0
81. Terminating: Octets Sent: 0
82. Terminating: Packets Rcvd: 0
83. Terminating: Octets Rcvd: 0
84. Terminating: Packets Lost: 0
85. Terminating: Inter-arrival Jitter (milliseconds): 0
86. Terminating: Average Transmission Delay (milliseconds): 0
87. BG Orig MDR Cust ID: 1234567890
88. BG Term MDR Cust ID: 1234567890
89. BG Authorization Code: ***
90. BG Account Code: ***
91. BG Orig Facility Type: 1 (BG Co-loc Line)
92. BG Term Facility Type: 1 (BG Co-loc Line)
93. BG Orig Station Facility ID: 0071001
94. BG Term Station Facility ID: 3333226
95. BG Call Completion Code: 8 (Not Completed/Unconnected: non-queued)
99. BG Access Code: 00
101.Original Dialed Digits: *82
106.Per Call Feature Extension: 1048576 (Private Call)
107.Call Event Indicator: 1048576 (A feature was Activated or Deactivated)
121.GCID Node: 0-0-0
122.GCID Number: 182361
124.GTID Node: 0-0-0
125.GTID Number: 182363
126.Ingress SIP Endpoint Address: 1.21.11.226
127.Egress SIP Endpoint Address: 1.50.11.82
129.Total Hold Time (Tenths of seconds): 0
Billing ID
Subscriber ID
FN9821FNUTFN_BABZZZEDDZ
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Siemens Enterprise Communications GmbH & Co. KG is a trademark licensee of Siemens AG
118
OpenScape Voice Siemens
01. Sequential Record Number: 1
02. Record Type: 00000000 (Standard CDR)
03. Start Time: 2011-04-08 T 06:04:17.2+0200
04. Duration of Call (Tenths of seconds): 0
05. Switch ID: vmv6node1
06. Record ID: 2011-04-08T06:04:17.2+0200:FF0001000000000061363A4E74500000
09. Call Segment Duration (Tenths of seconds): 0
11. Term Number/Called Party: 491234567890
12. Orig Number/Calling Party: 498970071001
13. ANI: 498970071001
16. Call Type: 0 (Voice Call)
18. Attempt Indicator: 4 (Calling party aborted prior to answer)
19. Release Cause/Completion Indicator: 16 (normal call clearing)
20. Bearer Capability Request: 0 (Circuit mode speech)
23. Operator Indicator: 0 (Direct Dialed)
26. Originating Nature of Address: 0 (Voice Call)
29. Route Selection: 0 (Standard)
30. Ingress Signaling Type: 17 (SIP)
31. Egress Signaling Type: 17 (SIP)
38. Incoming Trunk Group Signaling Type: 5 (Not Provisioned)
39. Outgoing Trunk Group Signaling Type: 5 (Not Provisioned)
40. Originating Party Identifier: 900 (Originating Endpoint on the switch)
41. Terminating Party Identifier: 903 (Terminating Endpoint not on the switch)
42. JIP Source Indicator: 5 (Unknown)
43. LRN Source Indicator: 5 (Unknown)
46. Originating Party Query Status Indicator: 9 (No query done)
47. Terminating Party Query Status Indicator: 9 (No query done)
49. Call Release Time: 2011-04-08 T 06:04:31.4+0200
50. Incoming Carrier Connect Time: 2011-04-08 T 06:04:17.2+0200
51. Incoming Carrier Release Time: 2011-04-08 T 06:04:31.4+0200
52. Outgoing Carrier Connect Time: 2011-04-08 T 06:04:31.1+0200
53. Outgoing Carrier Release Time: 2011-04-08 T 06:04:31.5+0200
54. Dialing and Presubscription Indicator: 0 (No IC/INC involved in call)
55. Calling Party Subaddress Delivery: 1 (Feature used, presumed delivered)
56. Called Party Subaddress Delivery: 1 (Feature used, presumed delivered)
57. Low-Layer Compatibility Information Delivery: 1 (Feature used, presumed delivered)
58. High-Layer Compatibility Information Delivery: 1 (Feature used, presumed delivered)
59. User-to-User Information/Fast Select: 1 (Feature used, presumed delivered)
63. Service Feature Codes: 0 (No Services)
64. Per Call Feature: 0 (No Features)
68. Originating Side: TOS Used: 0
69. Originating Side: Reservation Used: 0 (Default)
70. Originating: Packets Sent: 0
71. Originating: Octets Sent: 0
72. Originating: Packets Rcvd: 0
73. Originating: Octets Rcvd: 0
74. Originating: Packets Lost: 0
75. Originating: Inter-arrival Jitter (milliseconds): 0
76. Originating: Average Transmission Delay (milliseconds): 0
77. Terminating Side: Codec Used: 268435457 (G.711 64k PCM a-law, Telephone Event(Other))
78. Terminating Side: TOS Used: 0
79. Terminating Side: Reservation Used: 0 (Default)
80. Terminating: Packets Sent: 0
81. Terminating: Octets Sent: 0
82. Terminating: Packets Rcvd: 0
83. Terminating: Octets Rcvd: 0
84. Terminating: Packets Lost: 0
85. Terminating: Inter-arrival Jitter (milliseconds): 0
86. Terminating: Average Transmission Delay (milliseconds): 0
87. BG Orig MDR Cust ID: 1234567890
88. BG Term MDR Cust ID: 1234567890
89. BG Authorization Code: ***
90. BG Account Code: ***
91. BG Orig Facility Type: 1 (BG Co-loc Line)
92. BG Term Facility Type: 1 (BG Co-loc Line)
93. BG Orig Station Facility ID: 0071001
94. BG Term Station Facility ID: 3333226
95. BG Call Completion Code: 8 (Not Completed/Unconnected: non-queued)
99. BG Access Code: 00
101.Original Dialed Digits: *82
106.Per Call Feature Extension: 2097152 (Business Call)
107.Call Event Indicator: 1048576 (A feature was Activated or Deactivated)
121.GCID Node: 0-0-0
122.GCID Number: 182365
124.GTID Node: 0-0-0
125.GTID Number: 182367
126.Ingress SIP Endpoint Address: 1.21.11.226
127.Egress SIP Endpoint Address: 1.50.11.82
129.Total Hold Time (Tenths of seconds): 0
Billing ID
Subscriber ID
FN9821FNUTFN_BABZZZEDDZ
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Siemens Enterprise Communications GmbH & Co. KG 2011. All rights reserved.
Siemens Enterprise Communications GmbH & Co. KG is a trademark licensee of Siemens AG
119
OpenScape Voice Siemens
13.4 Usage
The feature can be used by the subscriber either by:
The authorization code and the phone number can optionally be confirmed with #.
#82 Dial the Prefix Access Code for deactivate Authorization Code
i or ^ Press OK or lift the handset
H The user will be prompted to enter the authorization code with:
Please enter your authorization code.
123456 Enter the configured authorization code for either business or
private calls.
H The user will be prompted to enter the phone number with:
Please enter the remaining digits required to complete your
call.
012345 Enter the phone number which should be called.
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Siemens Enterprise Communications GmbH & Co. KG 2011. All rights reserved.
Siemens Enterprise Communications GmbH & Co. KG is a trademark licensee of Siemens AG
120
OpenScape Voice Siemens
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OpenScape Voice Siemens
14 Intercom / Speaker Control
14.1 Overview
InterCom Calls, also known as Speaker Control Calls (1-way and 2-way), are used for many
applications such as Medical Centers, Executive/Assistant arrangements, automotive industry,
chemical and energy sector, emergency response centers, and by OScAR-Pro
[(OpenScape Alarm Response Professional) (formerly known as DAKS)].
For Intercom calls, the speaker (and microphone) on the destination device is automatically tur-
ned on when the call is connected.
Since Version 5, OpenScape Voice supports Intercom functionality locally and over SIP-Q. This
includes:
G One-way Intercom, variable and fixed (i.e. invoked via repertory dialing)
G Two-way Intercom, variable and fixed (i.e. invoked via repertory dialing)
G Community Groups
14.2 Functional Description
Two types of intercom calls are available:
Speaker Call 1-way
Provides an authorized originating party the capability to initiate a speaker call automatically
with a 1-way connection (send path only from the originator) to a local destination or a destina-
tion located over SIPQ. Access shall be via dialed access code + COM# or via programmed
rep dial key (known as Speaker Call Fixed 1-way). The speaker on the destination device will
automatically be turned on when the call is connected. An initial tone may first be provided to
the destination over the speaker to warn the user of an incoming voice call broadcast. The cal-
lers voice or announcement will be broadcast through the speaker for all in hearing distance to
hear. The caller goes on-hook and the call clears without the destination communicating back
to the caller.
>
This feature has been introduced with V5+.
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Siemens Enterprise Communications GmbH & Co. KG 2011. All rights reserved.
Siemens Enterprise Communications GmbH & Co. KG is a trademark licensee of Siemens AG
122
OpenScape Voice Siemens
Speaker Call 2-way
This variant is the same functionality and activation as 1-way Speaker calling with the added
capability of both the originator and destination bi-directional communicate (2-way speaker call
with both send and receive paths) automatically using the speaker and microphone facilities at
the destination. A different access code value for 1-way vs. 2-way speaker calling will be emp-
loyed. A periodic tone may be provisioned for the duration of the call and applied by the termi-
nating switch.
Intercom Features are accessible from:
Idle, dial, talk, and consultation dial states.
G OSV delivers an Intercom Call to an idle destination.
G When the intercom caller or called party is not provisioned for Intercom correctly or the
called party disallows an Intercom call from the calling party, the Intercom call will fail.
G The caller may re-dial the called party normally.
G If the called parties' device is not equipped with automatic intercom facilities, the call re-
verts to a 2-party normal call.
In general, most feature interactions are blocked at both the originating and terminating user.
Some exceptions include CALEA and emergency intrusion.
14.2.1 Limitations
G SIP devices supporting Intercom are restricted to OpenStage models and - OpenScape PE
(Personal Edition) clients (when equipped with an intercom speaker and microphone).
Devices that do not support intercom calling deliver a normal basic call when terminating
a speaker call.
G Shared keyset lines cannot be used for Intercom
G A 'rejection' key used to close a speaker call is not supported
G Network-wide members in survivability mode can not access this feature
Several feature interactions are blocked at both the originating and terminating user.
Interworking includes the capability for an incoming call to indicate override (e.g., CF, DND).
Intercom depends on that capability, for example, when the destination has activated Call
Forwarding or Do Not Disturb.
>
Intercom Overrides Call Forwarding and DND !
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123
OpenScape Voice Siemens
14.3 Provisioning
Configuration steps
G Create Subscribers
G Enable CSTA Over SIP for subscribers (e.g. via Feature Profile)
G Create Community Groups (Subscribers must exist in Originating & Terminating group)
G Enable devices to make and accept Intercom communication
G Configure Translation elements (Prefix Access Codes)
G Enable system wide parameters Terminating Intercom Control and DAKS Periodic Tones
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124
OpenScape Voice Siemens
In the left menu of OSV Assistant, within the group Teams a new menu item Community
Groups is created. Once this menu item is selected the list of the configured Community
Groups within the selected Business Group shall be presented in the right section of the page.
Community Groups are used to configure bilateral Intercom Calling capabilities and to provide
fast dialing for Intercom Calls.
Any OSV subscriber that shall originate or terminate Intercom calls, has to be provisioned with
a personal Community Group, defining the allowed Intercom targets / origins.
G Clone... a Community Group to another subscriber.
G Add... a Community Group for a specific subscriber of the
selected Business Group.
G Edit... the Community Group of a subscriber.
G Delete the Community Group of a subscriber.
COM Group for 498970071001
COM
No.
Destination
No.
DST Name Incoming
Block
1 498970071002 Yes
2 498970071003 Yes
3 15619231005 No
COM Group for 498970071002
COM
No.
Destination
No.
DST Name Incoming
Block
1 498970071001 No
COM Group for 15619231005
COM
No.
Destination
No.
DST Name Incoming
Block
1 498970071001 No
COM Group for 498970071003
COM
No.
Destination
No.
DST Name Incoming
Block
1 498970071001 No
o
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e

W
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t
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o

W
a
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Way
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In order to Add Community Group for a Subscriber. Select the DN and optionally provide a val-
ue for the Periodic Tone Timer.
During the creation of the Community Group it is mandatory to add at least one member in this
Group. OSV Assistant shall check this before allow the successful addition of the Community
Group.
Timer for Periodic
Tone that applies to
all terminating two-
way Intercom Calls.
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The administrator is allowed to administer (Add / Edi t/ Delete) the members of the Community
Group. The Members List of a Community Group can be sorted based on the Community Mem-
ber Number and on the Destination Number.
The COM# (Community Group Member Number or COM Number) is the (index) number to be
dialed after the one-way or two-way intercom access code. This column is searched to identify
the destination (member).
For local members, the Destination Number should be the member's FQDN. For remote mem-
bers, the member's calling party number (fully qualified directory number or display number, if
configured) must be used instead.
The Icoming Block if checked this member may not receive intercom calls (outgoing only!)
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127
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In the Subscriber screen a new tab Groups is introduced.
In this tab the Groups of the subscriber (e.g. Call Pickup Group, Multiline Hunt Group) are pre-
sented.
A new header Community Group Member is introduced in this tab to present the Community
Group Member List of the subscriber.
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OpenScape Voice Siemens
Terminating Intercom Control and DAKS Periodic Tones
A new menu item System-wide Settings is created under
OpenScape Voice Administration Feature Settings
The parameters Terminating Intercom Control and DAKS Periodic Tones will be provisioned
from the same Assistant menu.
Use the Terminating Intercom Control checkbox to specify whether OpenScape Vocie shall
safeguard terminating One-way Intercom Calls (default is True).
Specify the OScAR (DAKS)Periodic Tones interval, i.e. the periodic tones generation interval
for Two-way Intercom calls originated at DAKS (default is 120 seconds)
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There are two feature access codes required for this feature: Intercom one way and Intercom
two way. By dialing the desired Prefix Access Code (one way or 2way) followed by the COM#
(the destination index) a speaker phone call will be setup.
PAC for Intercom 1-Way
Feature Access Code (3 Digits)
must be entered together with the
Community Menber Number (0-99)
on which the feature should be ex-
ecuted.
Remove FAC (allways)
Prefix Type is Vertical Service to
activate a feature.
Nature of Address is Unknow
Destination Type Service
Select feature name
Intercom 1-Way
PAC for Intercom 2-Way
Feature Access Code (3 Digits)
must be entered together with the
Community Menber Number (0-99)
on which the feature should be ex-
ecuted.
Remove FAC (allways)
Prefix Type is Vertical Service to
activate a feature.
Nature of Address is Unknow
Destination Type Service
Select feature name
Intercom 2-Way
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Enable CSTA Over SIP for subscribers
The CSTA feature must be enabled with the option CSTA Over SIP on all intercom
subscribers (members). This can be done either via the Feature Profile or direct on the
subscriber:
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Device Enablement for Intercom
From the device web portal under
Administration System Features Configuration
Enable (check) Allow uaCSTA:
This is mandatory for all devices to originate or terminate Intercom.
Last, from the device web portal under
User Page Configuration Incoming calls CTI calls
Enable (check) Allow auto-answer:
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OpenScape Voice Siemens
14.4 Usage
To activate a one Way connection
To activate a two Way connection
*621Dial the Prefix Access Code for activate Intercom 1-Way followed by the
community member ID (e.g. 1)
i or ^ Press OK or lift the handset
*631Dial the Prefix Access Code for activate Intercom 2-Way followed by the
community member ID (e.g. 1)
i or ^ Press OK or lift the handset
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