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Features, System Info, Codes, Notes, sible features.

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Voice over IP (VoIP) SIP Telephony characters after "@" is the "Domain Name" or the "SIP Domain". The IP address of the OpenCom 100 must always be used here. ■ A STUN server (Simple Traversal of UDP over NAT) or a SIP proxy is not required because internal SIP subscribers on the LAN are directly connected to the OpenCom 100. Switch these functions off if possible. ■ With a SIP terminal, you can enter an international phone number with a leading plus. When you enter a call number in the E.123 format, the plus char is substituted by the "00" number sequence and the immediate line seizure via the standard route is activated for the call. If you prefer to dial in this number format, you should activate the international call number conversion (see E.164 conversion starting on page 163). Features SIP subscribers can establish connections to all other terminals and trunks. The SIP protocol generally works with block dialling. This is why the selected call number is only activated after an internal clock has expired or activated immediately via the hash key ("#") when dialling. This is why code number procedures without the hash key and code number procedures with a concluding hash key can be used. An overview of code number procedures that can be used is located in the Configurator on the System Info: Codes page. Activate "SIP phones". Please also note the corresponding information in the "OpenCom 100, Operation on Standard Terminals" user guide. Alongside code number procedures, SIP subscribers can also use a series of functional features realized via the SIP protocol. The OpenCom 100 is the ending for all SIP connections as opposed to what is usually the case on the Internet. This enables SIP subscribers to use OpenCom 100 features. Direct data exchange is thus not possible between two SIP subscribers. The following table shows the possible features. Features Incoming and outgoing calls with call number display (CLIP) Parallel connection of multiple SIP subscribers Enquiry, toggling, call waiting, threeway conference, reject Notes A SIP telephone requires a call number display for CLIP. SIP subscribers must be logged on under the same user identification. Operation or feature must be available on the SIP telephone or in the SIP software. 119

Section	Page
130 OpenCom 100 131 150	1
Mounting and Commissioning	1
Contents j	3
1. Features	11
2. Factory Settings on Delivery	18
2.1 Telephony Functions	18
2.1.1 OpenCom 130	18
2.1.2 OpenCom 131	18
2.1.3 OpenCom 150	19
2.2 Authorisations	19
2.3 Internet Functions	24
3. Installation	26
3.1 Scope of Delivery	26
3.2 Safety Precautions	27
3.3 Declarations of Conformity	28
3.4 Mounting Location	28
3.5 Wall Mounting	29
3.6 Installing an Expansion Set	29
3.7 Installing Interface Cards	32
3.7.1 V.24 and Doorstation Equipment Slots	32
3.7.2 Slots for Additional Interface Cards	34
3.7.3 Installing an M100-AT4 Card	36
3.8 Available Ports	39
3.8.1 OpenCom 130	39
3.8.2 OpenCom 131	40
3.8.3 OpenCom 150	41
3.8.4 Positions of the Ports	43
3.9 Interface Cards	45
3.9.1 OpenCom 131 (1 Slot)	45
3.9.2 OpenCom 130 (3 Slots)	45
3.9.3 OpenCom 150 (5 Slots)	48
3.10 Port Assignment, Termination, Cable Lengths	50
3.10.1 S0 Ports	50
3.10.2 Upn Ports	52
3.10.3 a/b Ports	53
3.10.4 Actor/Sensor	54
3.10.5 LAN Port	54
3.10.6 DSL Port	55
3.10.7 PCM Port	56
3.11 Power Failure	56
3.12 Connectible Terminals	57
3.12.1 Internal/External S0 Ports	57
3.12.2 Upn Ports	58
3.12.3 a/b Ports	58
3.12.4 Actor/Sensor Ports	61
3.12.5 COM Port	62
3.12.6 LAN Port	62
3.13 Installing the Memory Card	63
3.13.1 Safety Precautions	63
3.13.2 To Install a Memory Card …	64
4. Aastra 677x: Extensions and Accessories	65
4.1 Power Supply Unit	65
4.2 Key Extensions	65
4.3 Headset	69
5. S2M Connector Module	70
5.1 Installation	71
5.2 Configuration	73
6. Mounting the OpenCom 150 Rack InfoCom System	74
6.1 Safety Precautions	74
6.2 Technical Data	75
6.3 Pinning of RJ 45 Jacks	76
6.4 Scope of Delivery	77
7. Configuration	78
7.1 Brief Guide to Initial Configuration	79
7.2 Configuring the OpenCom 100	81
7.2.1 Preparing the Configuration	81
7.2.2 Starting the Web Console	81
7.2.3 Loading the Online Help	83
7.2.4 Finishing the Configuration	83
7.2.5 Saving and Loading the Configuration	83
7.2.6 Preconfiguration	84
7.2.7 Offline Configurator	84
7.2.8 Remote Configuration	84
7.2.9 Codes for IP Configuration	87
7.2.10 Receiving System Messages as E-Mail	88
7.2.11 Loading SW Updates	88
7.2.12 Resetting the System Data	89
7.2.13 Basic Hardware Settings Switch	89
7.2.14 Generating Your Own MoH Files	90
8. Configuration Examples	92
8.1 OpenCom 100 in Computer Networks	92
8.2 Introduction to TCP/IP	93
8.3 OpenCom 100 in a Serverless LAN	94
8.3.1 DNS Name Resolution	95
8.3.2 Internet Access	95
8.3.3 RAS Access	96
8.4 OpenCom 100 in a LAN with an IP-enabled Server	96
8.4.1 DNS Name Resolution	98
8.4.2 Internet Access	98
8.4.3 RAS Access	99
8.5 Branch Link	100
8.6 Useful Information on Internet Access	101
8.6.1 Costs	101
8.6.2 Using the Web	101
8.6.3 E-Mail	102
8.6.4 NAT	102
9. Voice over IP (VoIP)	104
9.1 Quick Start	105
9.1.1 IP System Telephony	105
9.1.2 External SIP Line	106
9.1.3 Internal SIP Telephony	108
9.2 Fundamentals	109
9.2.1 Propagation Delay and Bandwidth	110
9.2.2 Latency and Packet Length	110
9.2.3 Voice Quality	111
9.2.4 Optimisation	113
9.2.5 Call Set-up	114
9.2.6 Useful Services	114
9.3 Media Gateway (MGW)	115
9.3.1 Software MGW	115
9.3.2 MGW Interface Card	116
9.4 SIP Telephony	117
9.4.1 External SIP Connections	117
9.4.2 Internal SIP Subscribers	119
9.4.3 Aastra 673xi/675xi SIP Telephones	122
9.4.3.1 Aastra 673xi/675xi Setup	124
9.4.3.2 Aastra 673xi/675xi DHCP	126
9.4.3.3 Aastra 673xi/675xi Hot Desking	127
9.5 VoIP System Telephones	128
9.5.1 Device Properties	129
9.5.2 VoIP System Telephone Configuration	130
9.5.3 LAN DHCP Server	130
9.5.4 Start Procedure	131
9.5.5 Local Configuration	134
9.6 Aastra 277xip (OpenPhone 7x IPC)	137
9.6.1 Installation	138
9.6.2 Configuration	138
10. DECT over IP®	140
10.1 Properties	140
10.1.1 DECT Base Stations	140
10.1.2 Features	141
10.2 Configuration	142
10.2.1 Dual Operation	143
10.2.2 Synchronisation	144
10.2.3 Setting up the WLAN Function	144
10.2.4 Configuring for a Remote Location	148
11. PBX Cascading	150
11.1 Variants of PBX Cascading	150
11.2 Functionality of PBX Cascading	150
11.3 Putting a Cascaded PBX into Operation	152
11.3.1 Notes	153
11.4 Licensing Information	154
12. PBX Networking	155
12.1 Connections	156
12.1.1 Protocol: Q.SIG or DSS1	156
12.1.2 Master/Slave	157
12.1.3 L1 Clock	157
12.2 Types of Point-to-Point Connections	158
12.2.1 Direct Connection	158
12.2.2 Connection via an Active Transmission System	159
12.2.3 Connection via the Public Network	159
12.2.4 Connection via Q.SIG.IP	160
12.3 Configuration	161
12.3.1 Bundles	161
12.3.2 Routes	161
12.3.3 Numbering	162
12.4 Technical Details	163
13. Telephony	165
13.1 E.164 conversion	165
13.1.1 Configuration	165
13.1.2 Example	167
13.1.3 Further Information	168
13.2 Call Forwarding	168
13.2.1 Attributes	169
13.2.2 Loop Detection	170
13.2.3 Virtual Call Numbers	171
13.2.4 Hunt Groups	171
13.2.5 External Call Forwarding	172
13.2.6 Information on the Update	173
13.3 PIN Code Telephony	173
13.3.1 Configuration	174
13.3.2 Implementation	175
13.4 Switch authorisation	175
13.4.1 Configuration	176
13.4.2 Implementation	176
14. Fixed Mobile Convergence	178
14.1 Configuring FMC Telephones	183
14.2 Configuring “Aastra Mobile Client” Software	185
15. Team Functions	189
15.1 Introduction	189
15.1.1 Explanation of Keys	189
15.1.2 Team Configuration	191
15.2 Examples of Use	191
15.2.1 Executive/Secretary Team	192
15.2.2 Three-member Team	193
15.2.3 Unified Team	195
15.2.4 Toggle Team	196
16. Call Queue	198
16.1 Introduction	198
16.1.1 Activation of Queues	199
16.1.2 Call Forwarding	199
16.1.3 Pickup	200
16.1.4 Hunt Groups	200
16.2 Examples of Use	200
16.2.1 Enquiry Station for an Operator with Two System Telephones	200
16.2.2 Group of Three Enquiry Stations	202
17. Multi-Company Variant	204
17.1 Configuring the Multi-Company Variant	205
17.1.1 Activating the Multi-Company Variant	205
17.1.2 Configuring and Managing Companies	206
17.1.3 Assigning Users	206
17.1.4 Assigning a Bundle/SIP Trunk	207
17.1.5 Allocating Routing Codes	207
17.1.6 Configuring the Company Exchange	208
17.2 Working with the Multi-Company Variant	208
17.2.1 Company Telephone Book	208
17.2.2 Making Calls Between Companies	209
17.2.3 Billing Charges per Company	209
18. Configuring the PC Software	210
18.1 PC Offline Configuration	210
18.2 Setting up TAPI	212
18.3 Setting up NET CAPI	214
18.4 Browser for OpenCTI and OpenHotel	215
18.5 Setting up Video Telephony	216
18.6 Synchronising the PC Clock	217
18.7 Address Queries using LDAP	218
19. Frequently Asked Questions	219
19.1 General/Hardware	220
19.2 Telephony	221
19.3 PBX Networking	223
19.4 DECT	224
19.5 LAN	225
19.6 Internet	226
20. Technical Specifications	228
21. Notes on Disposal	235

Match case Limit results 1 per page

Voice over IP (VoIP)

SIP Telephony

119

characters after “@” is the “Domain Name” or the “SIP Domain”. The IP address

of the OpenCom 100 must always be used here.

■

A STUN server (Simple Traversal of UDP over NAT) or a SIP proxy is not required

because internal SIP subscribers on the LAN are directly connected to the

OpenCom 100. Switch these functions off if possible.

■

With a SIP terminal, you can enter an international phone number with a

leading plus. When you enter a call number in the E.123 format, the plus char is

substituted by the “00” number sequence and the immediate line seizure via

the standard route is activated for the call. If you prefer to dial in this number

format, you should activate the international call number conversion (see

E.164

conversion

starting on page 163).

Features

SIP subscribers can establish connections to all other terminals and trunks. The SIP

protocol generally works with block dialling. This is why the selected call number

is only activated after an internal clock has expired or activated immediately via

the hash key (“#”) when dialling. This is why code number procedures without the

hash key and code number procedures with a concluding hash key can be used.

An overview of code number procedures that can be used is located in the Config-

urator on the

System Info

Codes

page

Activate “SIP phones”. Please also note

the corresponding information in the “OpenCom 100, Operation on Standard Ter-

minals” user guide.

Alongside code number procedures, SIP subscribers can also use a series of func-

tional features realized via the SIP protocol. The OpenCom 100 is the ending for all

SIP connections as opposed to what is usually the case on the Internet. This

enables SIP subscribers to use OpenCom 100 features. Direct data exchange is

thus not possible between two SIP subscribers. The following table shows the pos-

sible features.

Features

Notes

Incoming and outgoing calls with call

number display (CLIP)

A SIP telephone requires a call number

display for CLIP.

Parallel connection of multiple

SIP subscribers

SIP subscribers must be logged on un-

der the same user identification.

Enquiry, toggling, call waiting, three-

way conference, reject

Operation or feature must be available

on the SIP telephone or in the SIP soft-

ware.