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SIP Provider, PBX Networking, Configuration

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Voice over IP (VoIP) SIP Telephony ■ You must log on ("Login") to the SIP registrar before you can use SIP telephony. Use the OpenCom 510 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, incomng RTP calls often fail to get past the Firewall or NAT configuration of the Internet gateway product used. Do not use OpenCom 510 as an Internet gateway if the product used is to be compatible with SIP telephony. These products provide a "Full Cone NAT" setting for this application. ■ 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 workplace 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 510, only SIP IDs consisting of numbers for identifying subscribers registered with the SIP provider specified can be addressed ■ For each SIP account you can create just one bundle. You can specify this bundle in routes as a connection option. 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 157). 125

Section	Page
OpenCom 510	1
Mounting and Commissioning	1
Contents	3
1. Features	11
2. Factory Settings	16
2.1 Telephony Functions	16
2.2 Authorisations	16
2.3 Internet Functions	19
3. Preliminary Information	21
3.1 Construction of the OpenCom 510	21
3.2 Scope of Delivery	21
3.3 Declarations of Conformity	22
4. Installation	23
4.1 Safety Precautions	23
4.1.1 General Instructions	23
4.1.2 Notes on the Mains Supply	24
4.1.3 Notes on EMC and Earthing	24
4.1.4 Notes on Installing the OpenCom 510	25
4.1.5 Notes on Installing Terminals	26
4.2 Installation, Ambient Conditions	26
4.3 Installation in a 19\	27
4.4 Installing Modules	28
4.4.1 Slots	29
4.4.2 Notes on Disconnecting the Mains Supply	30
4.4.3 Installing the Central Control Module	31
4.4.4 Installing Interface Cards	33
4.4.5 Installing a Power Supply Unit	35
4.5 Backplane	36
4.6 Uninterruptible Power Supply	39
4.7 Power Failure	39
5. Interfaces and Connectible Terminals	41
5.1 Overview	41
5.2 S0 Ports	42
5.2.1 Termination	43
5.2.2 External ISDN Ports (S0 External)	45
5.2.3 Internal S0 Ports	46
5.3 Upn Ports	46
5.3.1 Terminals Connected to Upn Ports	46
5.3.2 Technical Information	47
5.3.3 DECT Base Station Connection Variants	48
5.4 a/b Ports	50
5.4.1 Terminals Connected to a/b Ports	50
5.4.2 Technical Information	52
5.5 LAN Port	53
5.5.1 DSL Port	53
5.5.2 Service PC	54
5.6 COM Port	54
6. Accessories and Adapters (OpenPhone 61, 63, 65)	56
6.1 Upn Adapter	56
6.2 a/b Adapter	56
6.3 Audio Adapter	57
6.4 Approved Devices / Approved Accessories	58
6.5 Device Combinations	59
7. Extensions and accessories for system telephones (OpenPhone 71, 73, 75)	63
7.1 Power supply unit	63
7.2 Keypad extensions	64
7.3 Headset	66
8. Modules	67
8.1 Module Naming Conventions	68
8.2 Overview of Available Modules	69
8.3 MPS+1-AC	71
8.4 MC+1-3	74
8.5 MT+S2M1-1	78
8.6 MX+S01-8	82
8.7 MS+UPN1-8	85
8.8 MS+UPN2-8	87
8.9 MS+A1-8	89
8.10 MS+A1-4	91
9. Configuration	93
9.1 Brief Guide to Initial Configuration	94
9.2 Configuring the OpenCom 510	96
9.2.1 Preparing the Configuration	96
9.2.2 Starting the Web Console	96
9.2.3 Loading the Online Help	98
9.2.4 Finishing the Configuration	98
9.2.5 Preconfiguration	98
9.2.6 Offline Configurator	99
9.2.7 Remote Configuration	99
9.2.8 Codes for IP Configuration	101
9.2.9 Saving and Loading the Configuration	102
9.2.10 Receiving System Messages as E-Mail	102
9.2.11 Loading SW Updates	103
9.2.12 Resetting the System Data	103
9.2.13 Generating Your Own MoH Files	104
10. Configuration Examples	105
10.1 OpenCom 510 in Computer Networks	105
10.2 Introduction to TCP/IP	106
10.3 OpenCom 510 in a Serverless LAN	107
10.3.1 DNS Name Resolution	108
10.3.2 Internet Access	108
10.3.3 RAS Access	109
10.4 OpenCom 510 in a LAN with an IP-enabled Server	110
10.4.1 DNS Name Resolution	110
10.4.2 Internet Access	111
10.4.3 RAS Access	112
10.5 Branch Link	113
10.6 Useful Information on Internet Access	114
10.6.1 Costs	114
10.6.2 Using the Web	114
10.6.3 E-mail	115
10.6.4 NAT	115
11. Voice over IP (VoIP)	117
11.1 Fundamentals	118
11.1.1 Propagation Delay and Bandwidth	118
11.1.2 Latency and Packet Length	119
11.1.3 Voice Quality	120
11.1.4 Optimisation	122
11.1.5 Call set-up	122
11.1.6 Useful services	123
11.2 Media Gateway (MGW)	123
11.2.1 MG+ETH1-1	124
11.3 SIP Telephony	126
11.4 VoIP System Telephones	128
11.4.1 Device Properties	128
11.4.2 VoIP System Telephone Configuration	129
11.4.3 LAN DHCP Server	130
11.4.4 Start Procedure	131
11.5 OpenPhone IPC	133
11.5.1 Installation	134
11.5.2 Configuration	134
12. DECTnetIP	136
12.1 Properties	136
12.1.1 DECT base stations	136
12.1.2 Features	137
12.2 Configuration	138
12.2.1 Dual operation	139
12.2.2 Synchronisation	139
12.2.3 Setting up the WLAN function	140
12.2.4 Configuring for a remote location	143
13. PBX Cascading	147
13.1 Variants of PBX Cascading	147
13.2 Functionality of PBX Cascading	147
13.3 Putting a Cascaded PBX into Operation	148
13.3.1 Notes	151
13.4 Licensing Information	152
14. PBX Networking	153
14.1 Connections	154
14.1.1 Protocol: Q.SIG or DSS1	154
14.1.2 Master/Slave	155
14.1.3 L1 Clock	155
14.2 Types of Point-to-Point Connections	156
14.2.1 Direct Connection	156
14.2.2 Connection via an Active Transmission System	157
14.2.3 Connection via the Public Network	157
14.2.4 Connection via Q.SIG.IP	158
14.3 Configuration	159
14.3.1 Trunk groups	159
14.3.2 Routes	159
14.3.3 Numbering	160
14.4 Technical Details	162
15. Team Functions	163
15.1 Introduction	163
15.1.1 Explanation of Keys	163
15.1.2 Team Configuration	165
15.2 Examples of Use	166
15.2.1 Executive/Secretary Team	166
15.2.2 Three-member Team	168
15.2.3 Unified Team	169
15.2.4 Toggle Team	170
16. Call Queue	173
16.1 Introduction	173
16.1.1 Activation of Queues	174
16.1.2 Call Forwarding	174
16.1.3 Pickup	175
16.1.4 Hunt Groups	175
16.2 Examples of Use	175
16.2.1 Enquiry Station for an Operator with Two System Telephones	175
16.2.2 Group of Three Enquiry Stations	177
17. Multi-Company Variant	179
17.1 Configuring the Multi-Company Variant	180
17.1.1 Activating the Multi-Company Variant	180
17.1.2 Configuring and Managing Companies	181
17.1.3 Assigning Users	181
17.1.4 Assigning Trunk Groups/SIP Accounts	182
17.1.5 Allocating Routing Codes	182
17.1.6 Configuring the Company Exchange	183
17.2 Working with the Multi-Company Variant	183
17.2.1 Company Telephone Book	183
17.2.2 Making Calls Between Companies	184
17.2.3 Billing Charges per Company	184
18. Configuring the PC Software	185
18.1 PC Offline Configuration	185
18.2 Setting up TAPI	187
18.3 Setting up NET CAPI	189
18.4 Using the Systray Display	190
18.5 Browser for OpenCTI and OpenHotel	191
18.6 Setting up Video Telephony	192
18.7 Synchronising the PC Clock	193
18.8 Address Queries using LDAP	194
19. Configuration Guide	196
19.1 Overview	197
19.2 PBX Ports	198
19.3 LAN	199
19.4 WAN Settings	200
19.5 RAS Settings	201
19.6 Branch Settings	202
19.7 E-mail Function	203
19.8 E-mail Access	204
20. Frequently Asked Questions	205
20.1 General/Hardware	205
20.2 Telephony	206
20.3 DECT	208
20.4 LAN	209
20.5 Internet	210
21. Technical Data	213
22. Notes on disposal	216

Match case Limit results 1 per page

Voice over IP (VoIP)

SIP Telephony

125

■

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

Use the OpenCom 510 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, incomng RTP calls

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

product used. Do not use OpenCom 510 as an Internet gateway if the product

used is to be compatible with SIP telephony. These products provide a “Full

Cone NAT” setting for this application.

■

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 workplace 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 510, only SIP IDs consisting of

numbers for identifying subscribers registered with the SIP provider specified

can be addressed

■

For each SIP account you can create just one bundle. You can specify this

bundle in routes as a connection option. 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 157).