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different codecs can be used for sending and for receiving. It is also possible

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Voice over IP (VoIP) SIP Telephony You will also usually need the services of a SIP provider. A SIP provider operates a special server (the SIP Registrar) to handle connections. The SIP provider also operates a gateway to the ordinary telephone network which users pay to use and which enables the SIP provider to provide calls to the telephone network. A SIP connection can also accept incoming calls from the telephone network. The same voice transmission techniques as those explained in Fundamentals starting on page 107 are used for SIP telephony. SIP telephony has the following distinctive features: ■ Subscribers are identified through an e-mail-like "SIP ID" such as [email protected] or [email protected]. ■ SIP transmits dialling numbers always in a single data package (block dialling). Dialling can therefore be concluded with the hash key #on the system terminal, or the end of the number will be indicated by a time-out. The value for this time-out can be defined for each SIP provider separately. ■ You must log on ("Login") to the SIP registrar before you can use SIP telephony. Use the OpenCom 100 to manage important information for the registration (user name and password) of one or more SIP accounts. It is possible to make several calls simultaneously using a single SIP account. ■ A SIP connection causes constant Internet data traffic, so do not use SIP with Internet access which is paid for according to the time used. ■ RTP call data is also exchanged directly between terminals for SIP telephony, so different codecs can be used for sending and for receiving. It is also possible to change codecs dynamically during a call. You should use every codec available in the VoIP profile at least once, because this will enable you to establish connections with as many SIP subscribers as possible. ■ Fairly large packet lengths are quite normal on the Internet. They compensate for the longer packet propagation delay. ■ A bidirectional RTP data stream with a dynamically-assigned UDP port number is used to set up calls between subscribers. For this reason, incoming RTP calls often fail to get past the Firewall or NAT configuration of the Internet gateway product used. If you do not use the OpenCom 100 as the Internet gateway, the product should be compatible with SIP telephony. These products provide a "Full Cone NAT" setting for this application. 116

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

116

You will also usually need the services of a SIP provider. A SIP provider operates a

special server (the SIP Registrar) to handle connections. The SIP provider also

operates a gateway to the ordinary telephone network which users pay to use and

which enables the SIP provider to provide calls to the telephone network. A SIP

connection can also accept incoming calls from the telephone network.

The same voice transmission techniques as those explained in

Fundamentals

starting on page 107 are used for SIP telephony. SIP telephony has the following

distinctive features:

■

Subscribers are identified through an e-mail-like “SIP ID” such as

[email protected] or [email protected].

■

SIP transmits dialling numbers always in a single data package (block dialling).

Dialling can therefore be concluded with the hash key

on the system ter-

minal, or the end of the number will be indicated by a time-out. The value for

this time-out can be defined for each SIP provider separately.

■

You must log on (“Login”) to the SIP registrar before you can use SIP telephony.

Use the OpenCom 100 to manage important information for the registration

(user name and password) of one or more SIP accounts. It is possible to make

several calls simultaneously using a single SIP account.

■

A SIP connection causes constant Internet data traffic, so do not use SIP with

Internet access which is paid for according to the time used.

■

RTP call data is also exchanged directly between terminals for SIP telephony, so

different codecs can be used for sending and for receiving. It is also possible to

change codecs dynamically during a call. You should use every codec available

in the VoIP profile at least once, because this will enable you to establish con-

nections with as many SIP subscribers as possible.

■

Fairly large packet lengths are quite normal on the Internet. They compensate

for the longer packet propagation delay.

■

A bidirectional RTP data stream with a dynamically-assigned UDP port number

is used to set up calls between subscribers. For this reason, incoming RTP calls

often fail to get past the Firewall or NAT configuration of the Internet gateway

product used. If you do not use the OpenCom 100 as the Internet gateway, the

product should be compatible with SIP telephony. These products provide a

“Full Cone NAT” setting for this application.