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Aastra OpenCom 131 User Guide - Page 119

Internal SIP Subscribers, SIP Provider, Telephony, Trunks, Route, Configurator, SIP trunks

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Voice over IP (VoIP) SIP Telephony ■ To enable the use of multiple devices on a single Internet connection, the IP addresses used in a LAN (often 192.168.x.x) are translated to a valid IP address using address translation (NAT - Network Address Translation), but no status information is available for NAT on an incoming RTP connection. To avoid this problem, the IP address of a workstation computer or telephone visible on the Internet is determined using a STUN server (STUN: Simple Traversal of UDP over NAT). You can ask your SIP provider for the STUN server's IP address and port number. If you don't need a STUN server, leave the SIP Provider field empty. ■ For direct SIP telephony using OpenCom 100, only SIP IDs consisting of numbers for identifying subscribers registered with the SIP provider specified can be addressed ■ You can integrate an external SIP connection in the Telephony: Trunks: Route menu into the route configuration. You can use a network provider rule to specify the routing of numbers within a specific range to use SIP telephony as a preference (see also PBX Networking, under Configuration starting on page 159). You can configure SIP connections in the Configurator on the pages Telephony: Trunks: SIP provider and Telephony: Trunks: SIP trunks. Enter the technical attributes of a specific SIP provider, such as the IP addresses for the registrar and the STUN server under SIP provider. Under SIP trunks enter the information for an existing SIP account, such as the user name, password, assigned call number and the maximum number of simultaneous calls possible. 9.4.2 Internal SIP Subscribers The OpenCom 130 / 150 becomes available as the SIP server for internal SIP subscriber telephony switching services. SIP telephones connected via LAN or SIP programmes installed on workstation computers can thus establish connections to all other devices or trunks connected to the OpenCom 100. For operation as a SIP server a MGW Interface Card is required. Licence Assignment The number of possible SIP subscribers is determined by the number of licences purchased. In order to provide you with the greatest possible flexibility regarding usage of available licences, licence assignment is dynamic via the "Floating licence". Using a user/password combination ("SIP log on") you can have several SIP subscribers under the same call number. Only every new SIP log-on occupies a new licence. The technical log-on process of a SIP subscriber with a valid user 117

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

117

■

To enable the use of multiple devices on a single Internet connection, the IP

addresses used in a LAN (often 192.168.x.x) are translated to a valid IP address

using address translation (NAT - Network Address Translation), but no status

information is available for NAT on an incoming RTP connection. To avoid this

problem, the IP address of a workstation computer or telephone visible on the

Internet is determined using a STUN server (STUN: Simple Traversal of UDP over

NAT). You can ask your SIP provider for the STUN server’s IP address and port

number. If you don’t need a STUN server, leave the

SIP Provider

field empty.

■

For direct SIP telephony using OpenCom 100, only SIP IDs consisting of

numbers for identifying subscribers registered with the SIP provider specified

can be addressed

■

You can integrate an external SIP connection in the

Telephony

Trunks

Route

menu into the route configuration. You can use a network provider rule to

specify the routing of numbers within a specific range to use SIP telephony as a

preference (see also

PBX Networking

, under

Configuration

starting on

page 159).

You can configure SIP connections in the

Configurator

on the pages

Telephony

Trunks

SIP provider

and

Telephony

Trunks

SIP trunks

. Enter the technical

attributes of a specific SIP provider, such as the IP addresses for the registrar and

the STUN server under

SIP provider

. Under

SIP trunks

enter the information for

an existing SIP account, such as the user name, password, assigned call number

and the maximum number of simultaneous calls possible.

9.4.2

Internal SIP Subscribers

The OpenCom 130 / 150 becomes available as the SIP server for internal SIP sub-

scriber telephony switching services. SIP telephones connected via LAN or SIP pro-

grammes installed on workstation computers can thus establish connections to all

other devices or trunks connected to the OpenCom 100. For operation as a SIP

server a

MGW Interface Card

is required.

Licence Assignment

The number of possible SIP subscribers is determined by the number of licences

purchased. In order to provide you with the greatest possible flexibility regarding

usage of available licences, licence assignment is dynamic via the “Floating

licence”. Using a user/password combination (“SIP log on”) you can have several

SIP subscribers under the same call number. Only every new SIP log-on occupies a

new licence. The technical log-on process of a SIP subscriber with a valid user