Rohde & Schwarz Topex TOPEX VoisTel August 2011. Content l Why GSM IP PBX? l Target market l VoisTel – GSM IP PBX l IP PBX Services l Compatible SIP phones.

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Presentation transcript:

Rohde & Schwarz Topex TOPEX VoisTel August 2011

Content l Why GSM IP PBX? l Target market l VoisTel – GSM IP PBX l IP PBX Services l Compatible SIP phones l Benefits

Why GSM IP PBX ? l Employee productivity increase by using technology advanced systems l Product which meets major market requirements in terms of voice communications l Companies needs to use an advanced product which gathers features and services otherwise available only through significant investments in infrastructure l VoIP, GSM, PSTN/ISDN networks convergence in a single equipment l Facilitates user access to unavailable resources l Fast Return of Investments

Target market l SMB/SME companies l Large companies with several branches geographically distributed l System integrators who can use TOPEX solution in a complete voice – data solution l Fixed & Mobile Operators – integrated solution for final clients l Companies with legacy PBX system who wants to expand the number of users through hardware low investments

VoisTel – GSM IP PBX l All–in–one GSM IP PBX based on proprietary hardware and software - TOPEX platforms l VoIP, GSM and PSTN networks convergence l Complete solutions with SIP phones ❙ Proprietary operating system based on Linux – allows easier integration and developments with 3 rd party solutions ❙ Award winning solution - Product of The Year 2008 by Internet Telephony – TMC Net

Operating principles l Any – to – any calls between the supported networks l Allows SIP users register l Connecting analogue telephones or fax machines through ATA (Analog Telephone Adapters) ❙ Interconnection with: ❙ Legacy PBX through ISDN BRI or PRI interfaces ❙ GSM/UMTS/HSDPA mobile network through mobile modules ❙ IP network through WAN interface ❙ PSTN / ISDN network through ISDN interfaces

Interconnection diagram TOPEX VoisTel Headquarter LAN SIP Phones Softphone ATA POTSFax 4 BRI / 2 PRI PSTN 2G/3G network 4 ch IP network Branch

Hardware architecture l Small size reliable equipment which includes: l 2 VoIP interfaces (LAN and WAN) l 2 E1 interfaces or 4 BRI interfaces l 4 GSM/UMTS modules l 1 serial interfaces for local management l 4 FXS/4 FXO l Antena for the 4 GSM modules – integrated splitter l Reset button ❙ 1 SIM per module – can be used 4 SIM cards from different mobile operators ❙ Covered frequencies for the mobile networks: ❙ GSM: 850/900/1800/1900 MHz ❙ UMTS/HSDPA: 850/1900/2100 MHz Reset button RS-232 LAN & WAN 4 x BRI Antenna4 SIM slots

Main features I l Customized solution – flexible to the user requirements l 50 SIP subscribers – recommended l Unified Messaging including: l l Voice Mail l FAX l SMS l Advance LCR module – converts fixed to mobile calls into mobile to mobile calls l Routing module for both incoming and outgoing calls l ACL - Access Control List – allows the access onlz for the register users FAX Voice Mail SMS

Main features II l NAT Traversal for voice and signaling l Advanced security and integrated firewall l DISA support l IVR (Interactive Voice Response) l Configurable alarms – through , SMS or voice call l Local configuration (serial or IP) or distance (SSH or IP) l Statistics on different parameters such as ASR, ACD l Root access for root debugging, software updates, capacity extension

Standards and protocols l VoIP protocols supported: l SIP v.1, SIPv.2 l H.323 v.2, H.323v.4 l RTP, RTCP l SRTP, TLS l SIP-H.323 and H.323-SIP protocol conversion l Fax: l T.38 FoIP – according with ITU-T Annex D l pass through (G711u, G711A) l Voice codecs: G.711 A Law and u Law, G.723.1, G729 AB l Protocols E1: ISDN PRI - Euro ISDN, DSS1, CAS, R1, R2

Routing capabilities l Routing depending on: prefix, Caller ID, hour and date, ASR, ACD, load balancing (SIM cards usage) l Routing any to any between VoIP, E1, BRI and GSM VoIP E1BRIGSM VoIP E1BRI

Equipment monitoring l Hardware l 3 optical indicators on the equipment chase (power supply, application loading, memory reading) l 2 LEDs for each RJ45 connector l Software l Through the web based interface allowing: - GSM, ISDN, VoIP ports monitoring - Current calls visualization on the equipment – the number of active calls, details for each call - Subscribers status - 30 sec refresh rate

Administration web based interface l Equipment configuration, administration and equipment monitoring l SIP users add, delete and the modification of SIP users parameters l Routes definition for incoming and outgoing calls l Real time monitoring of the calls l Network parameters setting l Reset to default parameters on two cases – with or without network parameters

SIP Users l SIP users registered in the equipment l A section from the web based interface which allows adding or deleting SIP users from the equipment configuration l Allows parameters modification – every user access is made on personalized username and password l Includes several sections with services available to every registered user: Call Forward, Voic , Hunting, Pick-up, Forking, Do Not Disturb, etc l Can make calls SIP-SIP, SIP-GSM, GSM-SIP, SIP-E1, E1-SIP, SIP-VoIP, VoIP-SIP

SIP users services l IP PBX services: ● Mobility Extension ● Call Forward Offline ● Call Forward Busy ● Call Forward No Answer ● Call Forward Always ● Call Waiting ● Call Back ● Do Not Disturb ● Reject Anonymous Calls ● Multiple alias for each SIP user ● CLIP/CLIR ● Voice Mail Offline ● Voice Mail Busy ● Voice Mail No Answer ● Voice Mail Always ● Voic 2 ● Conference ● DCR – Dynamic CLIP Routing ● Missed Call 2 ● SIP Presence ● Call Hold Automatic/Consultation ● Music on Hold ● Call Transfer Attended/Unattended ● Call Hunting ● Call Pick-up ● Call Forking ● Forking pe Mobil ● Grupuri de Centrex ● Web 2 SMS ● SMS 2 Mail ● Mail 2 SMS ● IVR ● Auto Attendant ● SIP 2 SIP video call

SMS sending and receiving l SMS sending from the web based interface l Destination GSM number l SMS content l SMS 2 – send the received SMS on the installed SIM cards to a specified address Mobile phone GSM network TOPEX VoisTel SMS to SMS content transformed in the body

Mail 2 SMS l Mail 2 SMS – send an content received at a certain address through a SMS message: l subject must include the GSM number of the destination l SMS sending is made on username and password l The must be only in text format – without images or attachements SMS from If: - Subject = GSM number - = text - Username = ok - Password = ok Mobile phone GSM network TOPEX VoisTel

Voic services l Voice messaging services for each user l The voic service can be activated in several cases: l User is Offline l User is Busy – on call l User is not answering on the phone l For all the received calls Routes the call to 414 extension GSM network TOPEX VoisTel Call to No answer from 414 user Voic Message recording

Voic 2 l Service which allows sending voice messages to a specified address l The voice messages are registered in the IP PBX and if the option Voic 2 is checked then the messages will be sent as an attachment (.wav file) to the respective address GSM network TOPEX VoisTel Routes the call to 414 extension Call to Voic Message recording No answer from 414 subscriber to

Missed Call 2 l Allows a “mobile” user to check the missed calls from the SIP phone even when he is not in the office l The info with the missed calls details are sent through the TOPEX VoisTel equipment to an address of the user Routes the call to 414 extension Call to TOPEX VoisTel GSM network 414 User is not in the office Info missed call on the

Call Hunting/Pick- up/Forking l Groups based services – in order to activate these services must be defined hunting groups, pick-up groups and forking groups including several SIP users: l Hunting – directing the calls to another user from the same hunting group in the case when the SIP user is not answering or is busy. The call diverting is made according to each user priority l Pick-up – call pick-up of a subscriber from the same pick-up group in the case when the initial user is not in the office l Forking – a call to a subscriber from a forking group is distributed to all the subscribers from the group. The first user who will pick-up the phone will answer the call No answer No answer Picked up call IP network The same hunting group

Forking on mobile l Call Forking service can be used in two cases: l SIP users from the forking group in the same location l The telephones are registered on a single user account – SIP phone from the office and the mobile phones. Both phones will ring simultaneously. The first who will answer will take over the call IP network 414 GSM network Received call X

Mobility Extension l Use the mobile phone as an IP PBX extension: l Allows the transfer of the calls received on the mobile phone to another SIP user registered in the IP PBX – the transfer is made through DTMF codes l The service can be used by initial activation of mobile forking on the mobile and the transfer from the mobile to another extension IP network 414 GSM network Answered call X *415 – Transfer to Call session

DISA l Available service for the calls made from the mobile network to a subscriber registered in the IP PBX l For a call made on the SIM number from the equipment – the source party will receive DISA tone when the mobile user can insert the extension Call to 0723 xxx xxx DISA tone Insert extension 414 from the mobile phone Call session 414 GSM network 415 TOPEX VoisTel Inserted SIM with no 0723 xxx xxx

Dynamic CLIP Routing l Through Dynamic CLIP Routing, incoming calls from GSM to SIP extension are always routed to the right extension according to the CLIP Routing table SIP Phone – Extension 222 IP PBX TOPEX VoisTel – SIM Mobile Network Missed Call From Call to Call to Dynamic CLIP Routing to extension 414

Auto Attendant l Service which allows the transfer of an outgoing call to a certain department / extension from a company without human intervention l Uses and IVR (Interactive Voice Response) for recording welcome messages l Contains a menu such as – for sales press 1, for marketing press 2, etc l Allows direct transfer to a certain extension l Allows the registering of the welcome message for each user

Compatible SIP phones l Capability to provide a complete solution with SIP phones or softphones l TOPEX l Snom 300, 320, 360, 370, 820, m3 l Linksys SPA 921, 922, 941, 942, 962 l Alcatel 8 Series l Softphone: TOPEX, SJPhone, XLite

TOPEX VoisTel – licenses l Hardware platform cost including: l VoIP interfaces l E1 / BRI interface l GSM/UMTS interfaces l Sofware licenses for the IP users – 10 licenses package l SIP phones cost + +

Benefits l Low investments – fast savings – allows significant reduction of the telephone bill – LCR module l Integration of the mobile technology (GSM/UMTS) in an IP PBX solution l Secured, flexible, reliable l Fully redundant solution and technologically advanced l Easy to use, customized and manage l Solution which combines all the interconnection possibilities in a company l Rich set of services for the SIP users

Thank you!