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The SIP SwitchThe Best of Traditional Telephony, VoIP and Live Internet Communication!

SIP Switch users normally register to the built-in registrar proxy using proper email-like addresses like sue@smartcompany.coma. Th is gives free access to and from all users in the open SIP world!b

Users outside the SIP Switch/Firewall e.g. home workers or travelers, can also register in the SIP Switch and use all its features.Th e SIP Switch (or a user) registers to one or several SIP accounts at a VoIP service provider to get PSTN connectivity in and out.

All users of the SIP Switch can use this account for their calls to and from ordinary telephones.c,d

Incoming calls can be forwarded to one or several SIP-phones, or even your mobile!All SIP users benefi t from PBX-like functions, like using extensions, dialing 0 or 9 to reach the PSTN etc.

Th e Intertex Internet Gate products include a SIP capable fi rewall and a SIP server. Th e SIP Switch software integrates the usage of soft PC clients, SIP telephones and ordinary tele phones, adds PBX functionality and gives a unique connectivity with the old telephone network via common VoIP service provider’s SIP accounts.

Get PBX Functionality• Mimic conventional PBXs with the option to dial 0 (or 9) for an outside call.• Call internally using extensions, URIs (email like addresses), or conventional phone numbers.• Incoming calls can be mapped to specifi c clients on the LAN.• Forwarding and forking of telephone calls in parallel as well as in sequence.• Routing to voice mail server lets the caller leave a message when you’re not available.

• Restriction of incoming callers through blacklist and/or whitelist, based on caller or authenticated.• ENUM lookup before passing a call to the PSTN. If there is a known SIP address for the telephone number you are calling, the call will be forwarded to that address so that it will be IP communication all the way instead of using regular telephony.• Outgoing PSTN calls can be routed to the best gateway or VoIP provider.

A software for Intertex routers, fi rewalls and ADSL modems, that brings low cost, usefulness and ease of use to IP telephony.

The SIP Switch Combines it All• Reach all users in both the old telephony world and the open SIP world.• Use a single SIP account from a service provider for telephony connectivity for all SIP Switch users.• Use several service providers for the very lowest cost.• Calls to users in the open SIP world are FREE.• Get the most important PBX functions.• Use phone numbers, extensions or URIs (email like addresses).• Forward and parallel ringing (forking).• Video, presence and instant messaging for free to everyone in the open SIP world.

a. You can still also use the telephone numbers normally provided by the VoIP provider in (3)b. VoIP service providers usually only provides IP access to their own VoIP subscribers. Th ey often build closed islands, only connecting other VoIP users via charged PSTN instead of free Internet.c. Most VoIP service providers allow multiple inbound and outbound PSTN calls via a single SIP account.d. VoIP service providers seldom off er more than PSTN connectivity. Th e SIP Switch uses just that and adds all the rest.

SIP Switch users normally register to the built-in registrar proxy using proper email-like addresses

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Users outside the SIP Switch/Firewall e.g. home workers or travelers, can also register in the SIP

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Th e SIP Switch (or a user) registers to one or several SIP accounts at a VoIP service provider to

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Incoming calls can be forwarded to one or several SIP-phones, or even your mobile!

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The SIP technologymaking it work

Traditional Telephonyallows you to reach everyone and has several useful functions.

VoIP (IP telephony) gives you lower cost.

Live IP Communication brings new ways of communication.

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SIPSwitch

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KEY FEATURES Without With (For Intertex routers, fi rewalls and ADSL modems) SIP Switch SIP Switch

Communicate with the global SIP world YES YES

Multimedia communication (voice, video, instant messaging, etc) YES YES

Presence functionality YES YES

Dial 0 (or 9) for outside line - YES

Call internally using extensions, SIP URIs or phone numbers - YES

Incoming calls can be mapped to individual users - YES

Forwarding and forking of telephone calls - YES

Voice mail routing feature - YES

Restriction of incoming callers (SPAM control) - YES

Convenient dialing between SIP phones, soft SIP clients and PSTN - YES

Flexible Dial Plan to handle routing of outgoing calls - YES

ENUM lookup before sending calls to the PSTN - YES

User accounts and User control pages - YES

Use ordinary SIP accounts for PSTN connectivity - YES

Least cost routing and virtual foreign offi ces - YES

Emergency number is routed to a local PSTN gateway. ENUM lookup before passing the call to a PSTN gateway. International calls can be routed to the best local PSTN gateway so that calls are made at local rates. Call external number (PSTN) by dialing ”0” or ”9”.

INTERTEX DATA AB Rissneleden 45, SE-174 44 Sundbyberg Sweden, phone: +46 8-628 28 28, fax: +46 46 8 6286414, e-mail info@intertex.se, www.intertex.se

Have it ring one SIP address or phone number after another (sequence) or many at the same time (parallel). Send the call to voice mail after ringing for 25 seconds. Forward calls to other phone numbers or SIP addresses.

Outgoing Call Handling – the Dial Plan

PSTN gateway.1

PSTN gateway.2

International calls can be routed to the best local PSTN gateway so that calls are made 3

dialing ”0” or ”9”.4

Have it ring one SIP address or phone number after another (sequence) or 1

Send the call to voice mail after ringing for 25 seconds.2

Forward calls to other phone numbers or SIP addresses.3

Drastically Lower Your Costs for International Calls and Get Virtual Foreign Offi cesYou can use several PSTN connections with the SIP Switch. Get SIP accounts with suitable SIP service providers having good PSTN con-nectivity in the countries you communicate with. Th e long distance part of the call will go over the Internet and you will only have to use PSTN locally, paying only local rates instead of long distance rates. Th is also works the other way around, people calling you from countries where you have a SIP account only pay local fees.

Checkout SIP Service providers that has beentested with the SIP Switch at: www.intertex.se/itsp

Easy Usage and Management of Your Telephony• Convenient dialing between SIP phones, soft SIP clients and regular telephony (PSTN), using URIs, E.164 numbers and internal exten- sions.• Same functionally when working in an offi ce, at home or when traveling, by connecting your SIP client via the Internet.• Flexible Dial Plan with setup wizard, defi nes how outgoing number calls are handled.• User accounts and personal control pages let the individual user control his/her own settings. • Use a SIP account with an ITSP or SIP service provider as a PSTN gateway. Th e SIP Switch will allow all its users to make and receive calls at the same time (unless limited by the provided).

Incoming Call Handling

On line wizard makes setup easy!

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