1 SIP Trunking Service Definition SD032 V1.2 Issue Date 15 April 2011
2 SIP Trunking Service Definition Introduction InTechnology s SIP Trunking service enables organisations to run voice and data services via a single converged network and maximise on investments made in their existing telephony equipment. SIP Trunking delivers: The ability to make and receive calls to and from the PSTN (Public Switched Telephone Network) via an InTechnology managed IP network Number porting from a number of service providers 24 hour technical support Cost reduction through the reduction or removal of ISDN line rental charges Free on-net calls between the organisation s sites connected using the service Competitive tariffs for external calls
3 1. Service Description Graphical Overview InTechnology Voice Switches The service is delivered from two centrally managed, software based, IP Voice Switches hosted in InTechnology s highly secure and resilient Data Centre environment. The IP Voice Switch technology is based on tried and tested SIP (Session Initiation Protocol) open standards software providing a highly, scalable, cost effective and future proof platform for the provision of IP voice services. InTechnology s IP Voice platform has been deployed in a highly available configuration across InTechnology Data Centre locations in Harrogate and London. The core IP Voice Switches are maintained and supported by InTechnology 24 / 7 and access to the service
4 is delivered across InTechnology s national IP Network (LANnet) and via a range of connection technologies including SDSL, PPC leased lines and 10Mbps, 100Mbps and 1Gbps Ethernet as required. Delivery PBX Connect SIP is delivered over InTechnology s LANnet network. In order to assure call quality and performance across LANnet, InTechnology have implemented QoS systems throughout the LANnet network. A direct connection from the customer site to the nearest LANnet Point of Presence (POP) is required to provide service. These connections are available in bandwidths from 1Mbps to 1Gbps. The PBX Connect SIP service uses the G.729a Codec which results in up to 50kbps of bandwidth usage per call. The customer s PBX must be configured to use G.729a codec. Use of other codecs is not permitted. PSTN Connectivity For external calls, the service includes central, low-cost PSTN breakout from our IP Voice switches. InTechnology s high speed PSTN interconnect enables callers to make and receive external calls, for example other providers landline telephones, mobile telephones and international destinations. PBX Gateways For PBXs that do not have SIP trunk capability (i.e. a PBX that has Q.931 PRI or BRI ports only), InTechnology can provide the service inclusive of a managed gateway which will privide the required conversion. Emergency Calls Emergency 999 calls may be made using the service where supported by the customer s telephony equipment and dial plan. The customer is also responsible for ensuring that InTechnology are provided accurate caller location information. Deployment Process It is necessary to ensure that Customer Premises Equipment (CPE) is compatible with the service and to determine the appropriate gateway requirements where applicable. In order to do this, an audit of the PBX equipment may be necessary. Any reconfiguration required for the PBX to work with the service must be undertaken by the customer or their appointed contractor. InTechnology does not provide PBX reconfiguration or PBX support services.
5 PBX Interoperability Testing Note: PBX interoperability testing relates only to PBXs connecting directly to the service using SIP. For PBXs that are connecting via a gateway, PBX interoperability testing does not apply, however a functional system test still may still be required in this case. Due to the proprietary variations of the SIP standard that exist across different vendor platforms, InTechnology operates an interoperability test programme for each PBX that requires connecting to the service. This process is to ensure that the PBX operates correctly in accordance with an InTechnology service test plan. The testing process is as follows: 1. The customer must configure their PBX to route some non-business critical, test calls via the SIP trunk. The configuration details of the SIP trunk will be provided by InTechnology. (The SIP trunk provided for the purposes of interoperability testing will be with temporary phone numbers only.) 2. The customer may use an InTechnology LANnet connection or a suitable Internet connection of sufficient bandwidth as connectivity for the purposes of the interoperability testing. 3. An InTechnology technical consultant will liaise with an appointed customer contact to carry out the InTechnology service test plan and subject to successful testing, sign off the test plan with the customer. InTechnology will aim to complete the interoperability testing process within 30 days. During the interoperability testing phase: The customer will not be charged any service activation or monthly service charges. The customer will be charged for any chargeable calls made. Whilst every effort will be made, InTechnology offers no assurances that all PBXs will work successfully with the service. In the unlikely event that a PBX cannot successfully operate with the service in accordance with the InTechnology test plan, the customer may cancel the service during the first 30 days by providing InTechnology with written notice. If such notice is not received, the full minimum contract term will become applicable. Note: the customer will be responsible for all configuration works required on the PBX.
6 Service Components Office Connectivity Overview The PBX Connect SIP service is delivered over an InTechnology LANnet connection. If a LANnet connection is already installed, it is usually possible to supply the service over the existing connection without significant modification provided that there is sufficient bandwidth capacity. Technical Data PBX Connect SIP uses the G.729a codec which consumes approximately 50kb/s per call. Quality of Service (QoS) over DSL and LANnet Ethernet service (LES) connections is implemented using ingress and egress traffic control at both ends of the LANnet connection. As part of the managed service, InTechnology provides a fully managed multiservice router for each customer site to shape traffic and to ensure that voice traffic is not negatively impacted by other traffic (such as Internet traffic) using the LANnet connection. Each customer is provisioned with a private voice network over LANnet. This ensures security and business grade call quality between sites. LANnet is InTechnology s dedicated 2.5Gb/s national MPLS network and is specifically designed to carry high volumes of both voice and data traffic without causing degradation to the voice service. If all available voice channels to a site are in use, the IP Voice platform will prevent an additional call being setup to protect the other active calls from call degradation. An incoming call which is unable to complete because the circuit is congested will be treated as if the person they were calling was on the phone. Note: a call made between two phones on the same site does not consume a channel because it will route internally through the PBX. Suitable LANnet connections for PBX Connect SIP service are InTechnology managed ADSL 1000 and managed ADSL 2000, PPC and Ethernet connections. Gateway Where required, the voice gateway sits between the Q.931 interface of the PBX and the InTechnology managed WAN. This enables the PBX to communicate with InTechnology s IP Voice platform using a SIP trunk across InTechnology s network. InTechnology provides, configures, installs and manages the gateway as part of the service. SIP Trunk The SIP trunk is the service between InTechnology s voice platform and the customer s PBX (or gateway where applicable) that runs across the InTechnology network and enables the PBX to make and receive telephone calls. Numbers and Number Porting In order to facilitate inbound telephone calls, the service may be specified with geographic or non-geographic telephone numbers (DDIs). InTechnology can provide new number
7 ranges to use with the service, as well as port existing numbers and number ranges from existing service providers. Individual Network & Calling Features A range of optional network and calling features can be assigned to the service. Some of these features are chargeable. Some of these features are assigned at group level, meaning they apply to all DDIs within the trunk group. Other services are assigned at a per DDI level and apply only to that specific DDI. Service Calling Line Identity Presentation (CLIP) Feature Description Enables the called party to receive and display the calling party s telephone number before answering the call. The called party will only receive this information if the caller has not restricted the sending of their number (CLI). The called party will require suitable CPE in order to use CLIP. CLIP is assigned at an individual DDI level. Calling Line Identity Restriction (CLIR) Customers can request that their identities (telephone numbers) are not revealed at any time. This service is available free of charge. CLIR is assigned at an individual DDI level. Outgoing Calling Plan Restricts the ability to make outbound calls. Options include blocking all outbound calls and selective blocking based on call type. Call types include: Calls within the group Local calls National calls Freephone calls International calls Directory enquiry calls Premium rate calls including: o 070 o 0871 o 090 Outgoing Calling Plan is assigned at a group level and applies to all DDIs within that group. Incoming Calling Plan Restricts the ability to receive calls. Options include: Allow external inbound calls Partial allow / block external inbound calls only if transferred by another user in the group Block external calls
8 Incoming Calling Plan is assigned at a group level and applies to all DDIs within that group. Call Forwarding Redirects inbound calls to an alternate destination. Call forwarding incorporates three sub-services: Call Forwarding Always redirects all incoming calls to that DDI unconditionally to the specified alternate destination. Call Forwarding Busy redirects incoming calls to the alternate destination if the line associated with the DDI is busy. Call Forwarding No Answer redirects incoming calls to the alternate number if the call remains unanswered within a specified number of rings. Please note: The forwarded leg of the call will be charged to the customer at their pre-agreed tariff. Calls can only be forwarded to telephone numbers permitted by the outgoing calling plan Call Forwarding is assigned at an individual DDI level. Anonymous Call Rejection (ACR) Anonymous Call Rejection enables the rejection of calls from calling parties who have withheld their number. When activated, callers who have withheld their number are informed by an announcement that the person they are calling does not accept calls from anonymous callers. Anonymous Call Rejection is assigned at an individual DDI level. Call Forwarding Advanced Call Forwarding Advanced provides the capability to forward inbound calls intended for a particular DDI to another destination, when the incoming call matches pre-specified criteria. If the incoming call does not match any of the criteria, normal call handling applies. Definable criteria includes: A time schedule e.g. applies all day every day, or outside of business hours Whether the calling party line ID is private or unavailable A list of up to 12 phone numbers
9 The criteria can be combined within predicates (for example, incoming call from this number and within business hours and during work week). Multiple predicates can be defined and the call is forwarded when at least one of the predicates is met. The user can associate a different destination with each predicate, or use the same destination for all predicates. Call Forwarding Advanced is assigned at an individual DDI level. SIP Trunk User Mobility Pack This pack is designed to allow a user to take advantage of the mobility features available on InTechnology s voice platform whilst still utilising their existing PBX. Each DDI may be specified with this optional service pack that provides the user of that DDI with additional functionality delivered from InTechnology s voice platform. This pack provides the user with the Unity PC Assistant Toolbar. This software application sits on the user s Windows PC desktop and integrates into Microsoft Outlook and Internet Explorer. This toolbar provides quick and easy control of the features contain within this pack which comprise of: Call Forwarding Always Call Forwarding Busy Call Forwarding No Answer CommPilot Express Outlook Integration Remote Office Simultaneous Ring Personal Sequential Ring The PC Assistant Toolbar has a set of minimum hardware, software and network requirements. These are defined in the Technical Data section of this service definition. Web Administration Portal The service includes access to a web portal which may be used to administer certain features of the service. Access to this portal should only be provided to appointed personnel who are authorised to administer the organisation s telephony services. Examples of what the portal may be used for include configuring network and calling services.
10 Product Ordering Information Where no PBX Gateway is required: Partcode N-PBX-111 Description PBX Connect SIP (per channel) Where an ISDN2e PBX Gateway is required: Partcode N-PBX-101 N-PBX-112 N-PBX-113 N-PBX-114 Description PBX Connect ISDN2e (2 channel) PBX Connect ISDN2e (4 channel) PBX Connect ISDN2e (6 channel) PBX Connect ISDN2e (8 channel) Where an ISDN30e PBX Gateway is required: Partcode N-PBX-102 N-PBX-103 N-PBX-104 N-PBX-105 Description PBX Connect ISDN30e (8 channel) PBX Connect ISDN30e (15 channel) PBX Connect ISDN30e (30 channel) PBX Connect ISDN30e (60 channel) Individual Network & Calling Features: Partcode N-PBX-120 N-PBX-121 N-PBX-122 N-PBX-123 N-PBX-124 N-PBX-125 N-PBX-126 Description Calling Line Identity Presentation (CLIP) Calling Line Identity Restriction (CLIR) Outgoing Calling Plan Incoming Calling Plan Call Forwarding Call Forwarding Advanced Anonymous Call Rejection SIP Trunk User Mobility Pack: Partcode N-PBX-130 Description SIP Trunk User Mobility Pack
11 Service Capacities and Limitations Capacities The maximum number of simultaneous calls that may be made using the service is defined by the number of channels that the service is specified with. PBX Connect SIP may be specified with a minimum of 2 channels and a maximum of 1,000 channels. PBX Connect ISDN2e may be specified with the following number of channels: PBX Connect ISDN30e may be specified with the following number of channels: Limitations The maximum number of simultaneous calls that can be made using the service is limited to the amount of bandwidth available on the customer WAN connection The services can only be delivered via managed InTechnology network circuits of adequate bandwidth The service cannot be delivered over ADSL Max, ADSL 512kbps, SDSL or any form of bonded xdsl circuit The service is delivered subject to successful completion of interoperability testing, where applicable The ISDN30e version of the service only supports ISDN30e Q.931 interfaces The ISDN2e version of the service only supports ISDN2e Q.931 interfaces The following ISDN type Network & Calling Features are not supported: o Connected Line Identity Presentation (COLP) o Call Deflection o Caller Redirect o Customer controlled call forwarding of calls (using the PBX) o Malicious call indication ISDN2 DASS is not supported ISDN30 DASS is not supported Features supported by Q.SIG that are not supported within Q.931 are not supported by the service DPNSS is not supported Data services are not supported via the service Signalling between PBXs is not supported using the service Fax is not supported; this includes analogue or ISDN fax machines connected via the PBX or directly to the gateway Analogue devices such as modems, bank machines and franking machines are not supported Limitations with Transferring and Forwarding Calls
12 Call Quality on Transferred and Forwarded Calls PBX Connect SIP PBX Connect SIP enables a PBX with capable of using a SIP trunk to communicate with InTechnology s IP voice platform using the G.729a codec. Where a customer PBX transcodes between G.729a and any other codec (for example if the handsets are using a codec other than G.729a), call quality can become degraded during certain call scenarios. InTechnology will not provide support for any call quality issues where a PBX is transcoding between codecs. Call Quality on Transferred and Forwarded Calls PBX Connect ISDN2e / ISDN30e PBX Connect ISDN2e / ISDN30e enables a PBX with an ISDN2e / ISDN30e (Q.931) interface which uses the G.711 (PCM) codec, to communicate with InTechnology s IP voice network which uses the G.729a codec. The gateway provided as part of the service performs this conversion between G.711 (PCM) and G.729a. The conversion from G.711 (PCM) to G.729a (also called encoding) compresses the call, reducing the bandwidth required across the network to 50kbps per call. In certain call scenarios, calls can end up being encoded from G.711 (PCM) to G.729a more than once. In this event, call quality will become degraded. InTechnology cannot provide support for call quality issues on calls that have been encoded more than once. Please see the Supported and Unsupported Call Scenarios section of this document for more information on these limitations. Number Presentation on Forwarded Calls On forwarded calls, the presented telephone number will be the primary number associated with the trunk group. Should the PBX present the originating number (the number of person who s made the call), the service will not present this when the call is forwarded. The party to whom the call is forwarded will still be presented with the primary telephone number associated with the PBX Connect group. This is not applicable to transferred calls.
13 Supported and Unsupported Call Scenarios InTechnology will not provide support for call quality issues for call scenarios that are not stated as support in this section of the service definition. Call scenarios that are stated as being unsupported and any other call scenarios not documented in this section that may arise are also unsupported. This section applies to: PBX Connect ISDN2e PBX Connect ISDN30e PBX Connect SIP where transcoding between G.729a and another codec occurs on the customer PBX (i.e. the handsets are not running G.729a) Supported use of the service The PBX may make calls to the PSTN, to Unity IP Voice* users and to another PBX connected to the service The PBX may receive calls from the PSTN, from Unity IP Voice users and from another PBX connected to the service The PBX may receive calls from the PSTN that have been transferred or forwarded by a Unity IP Voice user Calls made by the PBX to Unity IP Voice users may be transferred or forwarded by the Unity IP Voice user to the PSTN, to other Unity IP Voice users or to another PBX connected to the service *Unity IP Voice is InTechnology s hosted IP Telephony service Exceptions All inbound calls may not be transferred or forwarded by the PBX, except to other internal telephone extensions connected to the same PBX Where more than one PBX is connected to the service, calls made between the PBXs may not then be transferred or forwarded Overview of Supported and Unsupported Call Scenarios Call Scenario Description Call Quality Supported PBX to PSTN A call is made from the PBX to the PSTN Yes PSTN to PBX A call is made from the PSTN to the PBX Yes PBX to Unity A call is made from the PBX to a Unity IP Voice Yes user Unity to PBX A call is made by a Unity IP Voice user to the Yes PBX PSTN to PBX to PSTN A call from the PSTN is transferred or forwarded No by the PBX to another party on the PSTN PSTN to PBX to Unity A call from the PSTN is transferred or forwarded No by the PBX to a Unity IP Voice user PSTN to Unity to PBX A call from the PSTN is transferred or forwarded Yes by a Unity IP Voice user to the PBX PBX to Unity to PSTN A call from the PBX is transferred or forwarded Yes by a Unity IP Voice user to another party on the PSTN Unity to PBX to PSTN A call from a Unity IP Voice user is transferred or forwarded by the PBX to another party on the PSTN No
14 Unity to PBX to Unity PBX to PBX PSTN to PBX to PBX PBX to PBX to PSTN PBX to PBX to Unity Unity to PBX to PBX PBX to Unity to PBX PSTN to PBX to PBX to PSTN A call from a Unity IP Voice user is transferred or forwarded by the PBX to another Unity IP Voice user A call is made from a PBX connected via the service to another PBX connected via the service A call from the PSTN is transferred or forwarded by the PBX to another PBX also connected via the service A call made from a PBX connected via the service, to another PBX connected via the service, is transferred by the 2 nd PBX to another party on the PSTN A call made from a PBX connected via the service, to another PBX connected via the service, is transferred by the 2 nd PBX to a Unity IP Voice user A call made from a Unity IP Voice user to a PBX connected via the service is transferred or forwarded to another PBX connected via the service A call made from a PBX connected via the service is transferred or forwarded by a Unity IP Voice user to another PBX connected via the service A call made from the PSTN is transferred or forwarded by a PBX connected via the service to another PBX connected to the services. This call is then transferred or forwarded by the 2 nd PBX to another party on the PSTN No Yes No No No No Yes No
15 Technical Data Hardware, software and connectivity requirements for the PC Assistant Toolbar software application are detailed below. Internet connectivity requirements Software requirements All Platforms Access to ipt.intechnology.co.uk on TCP / UDP port 80, 2208, 443 Access to portal.ipt.intechnology.co.uk on TCP / UDP port 80, 2208, 443 Access to bsews1.ipt.intechnology.co.uk on TCP / UDP port 80, 2208, 443 Access to bsews2.ipt.intechnology.co.uk on TCP / UDP port 80, 2208, 443 Access to ecom.intechnology.co.uk on TCP port 80 Windows 2000 with SP4 (or higher), Windows XP, Windows Vista, or Citrix Presentation Server 3 or 4 Windows Installer 2.0 Internet Explorer 6.0, 7.0, or 8.0 (required for IE toolbar edition) Mozilla Firefox 2.0 or 3.0 (required for Firefox toolbar edition) Outlook 2000 SP3, 2002/XP SP2, 2003, 2007 (required for Outlook toolbar edition) Citrix Presentation Server Platform The application can be published on a Citrix server via the Management Console for MetaFrame. Citrix ICA Client Workstation Platform No additional software requirements. Microsoft Windows Platform 2 GHz Intel Pentium 4 or equivalent CPU 512MB of RAM 60MB of free hard disk space 8MB Video Graphics Card SVGA monitor (15 or larger) 800 x 600 minimum screen resolution Hardware requirements Citrix Presentation Server Platform 2 GHz Intel Pentium 4 or equivalent CPU Minimum 2GB of RAM 60MB of free hard disk space Citrix ICA Client Workstation The hardware requirements for a Citrix ICA client workstation include: 1.2 GHz or higher, Pentium 3, or compatible CPU 128 megabytes (MB) of RAM Video graphics card with 8 MB of RAM minimum 800 x 600 screen resolution minimum Network connection of minimum 56 Kbps speed
16 PBX Connect ISDN2e and ISDN30e Service Level Agreement Service Availability The time for service restoration (TSR) of a fault is defined as the time period between fault detection and service restoration. The TSR starts with pro-active fault detection by the InTechnology service management centre (SMC) or when InTechnology acknowledges the Customer's reported fault. Service Element PBX Connect ISDN2e and ISDN30e Service Availability Time for Service Restoration Next working day Service Credits InTechnology shall pay the Customer by way of Service Credits, a sum equal to one hour's Service Charge (exclusive of VAT) for each full-completed hour in excess of the TSR based on the following calculation. Service Credit = (Monthly Service Charge) x Full completed hr(s) in excess of SLA / Number of hours in a month The maximum Service Credit available in respect of any one month will be equal to and no more than one month s Service Charge in any one-month. Please note: One or more instances in any 24 hour period will only result in one service credit being paid. The maximum Service Credit available in respect of any one month will be equal to and no more than one Month Service Charge in any one-month. Any downtime relating to the LANnet Services will also incur Service Credits as defined in the LANnet SLA. Any reduced charges under this Service Level Agreement will be confirmed by credit note issued by InTechnology to Customer, confirming the adjustment to be made to the following monthly charge. InTechnology shall not be liable to pay any more to the Customer by way of service credits in any one month than a sum that is equal to (the VAT exclusive amount of) one-month service charge regardless of the number of network outages in that particular month. Service Availability Limitations The Customer should acknowledge that the warranties provided above shall not apply in the event that any failure or suspension of the Services arising as a result of a failure of the Customer Equipment, Customer Internet connection or is caused by any action or omission of the Customer, its employees, agents, sub-contractors or invitees.
17 InTechnology shall not be liable for any failure to comply with the service levels defined where the Customer is in breach of any warranties set out in the contract. In calculating Service availability the following circumstances are excluded: Service unavailable or affected as a result of Service suspension pursuant to the Service Agreement. Service unavailable or affected due to faults on the Customer's side of the service. Service unavailable or affected due to faults on the Customer s equipment. Service unavailable or affected due to incorrect configuration of customer equipment. Service unavailable or affected due to unsupported use of the service. Service unavailable or affected due to circumstances created by the customer. Service unavailable or affected due to planned maintenance. Service unavailable or affected due to emergency maintenance. Service unavailable or affected due to Force Majeure. Call completion not possible due to busy signal fully utilised Trunk line or network capacity for example. Planned maintenance can involve a temporary suspension of part or all of the service in order to enable InTechnology to undertake vital remedial/maintenance or upgrade work. Planned maintenance and controlled outages will always be notified to the customer at least 7 days in advance and be planned in such a way to have minimum impact on the customer's operations. Controlled outages will not be classified as faults as the core systems are fully redundant and availability will be unaffected. Controlled outages to the service for the planned maintenance take place at 4am - 6am on Monday and Thursday. Emergency maintenance required as a result of identifying a problem through ongoing monitoring and management, that could potentially cause an outage or failure of the service, will be notified to the Customer at the earliest possible time and be managed in such a way to have minimum impact on the Customer's operation.
18 PBX Connect SIP Service Level Agreement This Service Level Agreement (SLA) sets out terms on which InTechnology shall provide the Service to the customer. This Service Level Agreement is subject to and shall be interpreted in accordance with the other terms of the Agreement. The purpose of this Service Level Agreement in conjunction with the Service Description is to set out the agreed supply, operation and management of the PBX Connect SIP Service by InTechnology. Please refer to the InTechnology Service Descriptions for further details of the Network Services. No changes may be made to this Service Level Agreement, except by written agreement dated and signed by both parties. Voice Specific SLA Parameter Core Voice System Availability %, TSR 10 minutes The Core Voice Systems from which the service is delivered are located in two, geographically separated, InTechnology managed data centre facilities and are built in a fully resilient and redundant fashion. Each system provides failover to the other system and therefore a total failure can only occur when both systems are simultaneously unavailable. InTechnology has implemented HP OpenView and uses this system to manage and report on these platforms. The HP OpenView system periodically probes each element of the Voice systems and raises the alert if total service availability is lost. In the highly unlikely event of this happening the measure of core system availability and associated TSR will be measured from the point of total system failure. Service Availability Limitations The Customer should acknowledge that the warranties provided above shall not apply in the event that any failure or suspension of the Services arising as a result of a failure of the Customer Equipment, or is caused by any action or omission of the Customer, its employees, agents, sub-contractors or invitees. InTechnology shall not be liable for any failure to comply with the service levels defined where the Customer is in breach of any warranties set out in the contract. In calculating Service availability the following circumstances are excluded: Service unavailable or affected as a result of Service suspension pursuant to the Service Agreement. Service unavailable or affected due to faults on the Customer's side of the service. Service unavailable or affected due to faults on the Customer s equipment. Service unavailable or affected due to incorrect configuration of customer equipment. Service unavailable or affected due to unsupported use of the service. Service unavailable or affected due to the use of non-accredited PBXs, PBX hardware or PBX software. Service unavailable or affected due to circumstances created by the customer.
19 Service unavailable or affected due to planned maintenance. Service unavailable or affected due to emergency maintenance. Service unavailable or affected due to Force Majeure. Call completion not possible due to busy signal fully utilised Trunk line or network capacity for example. Planned maintenance can involve a temporary suspension of part or all of the service in order to enable InTechnology to undertake vital remedial/maintenance or upgrade work. Planned maintenance and controlled outages will always be notified to the customer at least 7 days in advance and be planned in such a way to have minimum impact on the customer's operations. Controlled outages will not be classified as faults as the core systems are fully redundant and availability will be unaffected. Controlled outages to the service for the planned maintenance take place at 4am - 6am on Monday and Thursday. Emergency maintenance required as a result of identifying a problem through ongoing monitoring and management, that could potentially cause an outage or failure of the service, will be notified to the Customer at the earliest possible time and be managed in such a way to have minimum impact on the Customer's operation. Service Credits InTechnology shall pay to the Customer Service Credits for each proven instance of failure to comply with the SLA metrics defined above. An instance shall be defined as: Core System Availability - Failure to restore service within the stated TSR. One Service Credit will be equal to one day s Service Charges based on the average daily charge of the previous month s total PBX Connect SIP Service Charges invoiced for the site(s) affected. Please note: One or more instances in any 24 hour period will only result in one service credit being paid. The maximum Service Credit available in respect of any one month will be equal to and no more than one Month Service Charge in any one-month. Any reduced charges under this Service Level Agreement will be confirmed by credit note issued by InTechnology to Customer, confirming the adjustment to be made to the following monthly charge. InTechnology shall not be liable to pay any more to the Customer by way of service credits in any one month than a sum that is equal to (the VAT exclusive amount of) one-month service charge regardless of the number of network outages in that particular month.
20 Responsibilities Customer Responsibilities PBX telephone system; availability, operation, configuration and maintenance. PBX telephone handsets; operation, configuration and maintenance. Ongoing adds / moves / changes associated with the PBX. All fixed cabling including data and voice; suitability for use with PBX, operation, maintenance. All LAN configuration, operation and maintenance. Any re-configuration or upgrades (hardware or software) to the PBX that may be required in order for the PBX to operate with the service. Provision and ongoing supply of mains power to any InTechnology hardware required in order to provide the service. Implementation of suitable security policies to prevent fraudulent use of the PBX and the service. InTechnology Responsibilities Service provision and availability. Provision of analogue telephone lines to service analogue devices where requested by the customer.
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Section I: Introduction and Service Description. OPERATIONS MANUAL Voice Over IP 1. Voice over IP. Voice over IP services (collectively "VoIP") IP Integrated Access works with existing key systems or PBX
Delivering effective voice services over your IP Network Why duplicate costs? Voice and data networks have evolved independently of each other and consequently most organisations operate two separate networks,
SIP trunking, simply put, is a way for you to accomplish something that you already do, for less money, with equal or better quality, and with greater functionality. A Guide to SIP V4 An Introduction to
Lines and Calls Service Definition SD013 v2 Issue Date 15 April 2013 Lines and Calls Service Definition... 4 Introduction... 4 Service Description... 4 Overview... 4 Lines... 4 Calls... 5 Order Types...
TXI Telephony Product Book 2012 2012 PRICING, HOW TO This section details the common pricing scenarios for the following: Receptionist Call Centre SIP Trunking Fax to Email Messaging Receptionist: Receptionist
VitalPBX Hosted Voice That Works For You Vital Voice & Data s VitalPBX VVD Hosted PBX solutions provide you with the stability of a traditional telephone PBX system and the flexibility that only a next
Integrated Voice Service Guide Save money and maximize bandwidth efficiency Version 201009 TABLE OF CONTENTS TABLE OF CONTENTS...2 PRODUCT OVERVIEW...3 WHAT IS INTEGRATED VOICE?...3 HOW IT WORKS...4 BASE
SIP Trunking DEEP DIVE: The Service Provider Larry Keefer, AT&T Consulting UC Practice Director August 12, 2014 2014 AT&T Intellectual Property. All rights reserved. AT&T, the AT&T logo and all other AT&T
NSW Government Telecommunications: SIP (Session Initiation Protocol) Standard October 2014 CONTENTS 1. Context 3 2. Required NSW Government business outcomes 3 3. Additional business outcomes for agency
Service Guide Save money and maximize bandwidth efficiency Learn More: Call us at 877.634.2728. www.megapath.com Table of Contents Product overview... 3 What is integrated voice?... 3 How it works... 4
Voice over IP Basics for IT Technicians White Paper Executive summary The IP phone is coming or has arrived on desk near you. The IP phone is not a PC, but does have a number of hardware and software elements
ADSL Aux Bearer BIS Block Terminal Care Level Agreement CDTA CLIP CLIR Channels Contact CPE Asymmetric Digital Subscriber Line. Commonly referred to as broadband, this is a type of broadband used to connect
Unity IP Voice Service Definition SD009 v2.5 Issue Date 20 th August 2010 Unity IP Voice Service Definition Unity IP Voice Specific Terms and Conditions The following clauses are in addition to the InTechnology
AAPT BUSINESS SIP VOICE Service Schedule An AAPT Business Voice Solution This Service Schedule forms part of the Agreement between Us and You and cannot be used as a stand-alone agreement. Any terms defined
VoIP Bandwidth Considerations - design decisions When calculating the bandwidth requirements for a VoIP implementation the two main protocols are: a signalling protocol such as SIP, H.323, SCCP, IAX or
AT&T Connect Video Conferencing Functional and Architectural Overview v9.5 October 2012 Video Conferencing Functional and Architectural Overview Published by: AT&T Intellectual Property Product: AT&T Connect
SIP Matters at 360 Solutions Flexible, resilient and cost effective call routing for your voice traffic 360 Solutions What is SIP Trunking? Our SIP Trunking solution will connect your site directly into
Whitepaper: Microsoft Office Communications Server 2007 R2 and Cisco Unified Communications Manager Integration Options Document Summary This document provides information on several integration scenarios
A cost effective alternative to ISDN that gives your business flexibility and business continuity What is Genius SIP? Genius SIP is a market-leading SIP Trunking solution, connecting your site directly
North Central Texas Council of Governments Addendum RFP # NCT-2010-7 Issued April 09, 2010 Section #1 to RFP: # NCT-2010-7 Addendum RFP NCT # 2010-7 Questions & Answers 1. The following is assumed; please
SIP TRUNKING SERVICES SERVICE SCHEDULE (October 2013) 1. About this document 1.1 This document is part of the Commander Standard Form of Agreement ( SFOA ) under section 479 Telecommunications Act 1997,
MANAGED PBX SERVICE SCHEDULE 1. APPLICABILITY This Service Schedule is applicable only to the COF for the purchase of Managed PBX Services which has been signed by the Customer and Neotel. 2. DEFINITIONS
Mistral Hosted PBX The future of business phone systems Tel: 0870 751 6300 Web: www.mistral.net Mistral HOSTED PBX is a revolutionary futureproof way of implementing your office or enterprise phone system.
Receptionist-Small Business Administrator guide What is it? Receptionist-Small Business works with your desk phone, soft phone, or mobile device so you can control calls, monitor the lines of employees,
Contents 1 Introduction... 2 2 PBX... 3 3 IP PBX... 4 3.1 How It Works... 4 3.2 Functions of IP PBX... 5 3.3 Benefits of IP PBX... 5 4 Evolution of IP PBX... 6 4.1 Fuelling Factors... 6 4.1.1 Demands from
What is ViP? Voice internet Phone ViP is a complete package for your hosted/voip telephone solution. The ViP package has been created to keep things simple for you. We bundle in all the hardware, software,
ehealth and VoIP Overview Voice over IP (VoIP) configurations can be very complex. Your network could contain a variety of devices, applications, and configuration capabilities to support voice traffic.
VoIP in the Enterprise Date: March. 2005 Author: Sonia Hanson Version: 1.1 1 1 Background Voice over IP In the late 1990s Voice over IP (VoIP) was seen as a disruptive new technology that had the potential
Management Summary for Unified Communications IP PBX Prepared By for YOU of General: The Unified Communication Internet Protocol Private Branch Exchange (UCIPPBX) is a fully realised 3 rd generation office
SIP Trunking voice services A cost effective alternative to ISDN that gives your business flexibility and business continuity WHAT IS IP DIRECTCONNECT? IP DirectConnect is the UK s market-leading SIP Trunking
Whitepaper - Guide to migrating to SIP There are both cost savings and productivity benefits to be had by moving from the old circuit switched telephony world to SIP. This short guide describes the steps
INTRODUCTION VoIP, Hosted Telephony, SIP, Unified Communications or IP Centrex can all be used as terms to define Hosted VoIP, but what are the main business reasons for implementing Hosted Telephony in
Section 1 The VoIP Service Schedule 2 VoIP Service The Voice-over-IP (VoIP) Service supplied by ORION /REACHNET under this Agreement is only offered to Customers of ORION /REACHNET s Satellite Internet
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5. DEPLOYMENT ISSUES Having described the fundamentals of VoIP and underlying IP infrastructure, let s address deployment issues. 5.1 LEGACY INTEGRATION In most cases, enterprises own legacy PBX systems,
APPENDIX 1 LOT 1: IP TELEPHONY SERVICES A. TECHNICAL SPECIFICATIONS EXISTING SYSTEMS IP based phone service and phone sets, rented: IP based Phone service for 35 phone numbers/users; Centrex PBX solution
NSN s SIP trunking is the fastest growing telephony service. It connects your site directly into NSN s network via an IP connection to carry and terminate your inbound and outbound voice calls across the
Phone 1800 800 723 email@example.com switchtelecom.com.au May2014-Version 1 Welcome to a new kind of telecommunications company Best Products Mobile Offering an extensive range of smartphones
Allstream Converged IP Telephony SIP Trunking Solution An Allstream White Paper 1 Table of contents Introduction 1 Traditional trunking: a quick overview 1 SIP trunking: a quick overview 1 Why SIP trunking?
SPRINT SIP TRUNKING SERVICE PRODUCT ANNEX The following terms and conditions in this Sprint SIP Trunking Service Product Annex ( Annex ), together with the applicable Sprint service agreement ( Agreement
Voice over IP (VoIP) Basics for IT Technicians VoIP brings a new environment to the network technician that requires expanded knowledge and tools to deploy and troubleshoot IP phones. This paper provides
Cisco IP Communicator (Softphone) Compatibility Cisco IP Communicator is Windows based and works on both XP and Vista The minimum PC requirements for use with Microsoft Windows XP are: Microsoft Windows
WINSCRIBE HARDWARE SPECIFICATIONS Technology Overview proposes centralization of resources by providing a networked solution that fits into the existing framework of your server environment with minimal
Verizon Unified Communications and Collaboration as a Service Service Level Agreement ( SLA ) 1. Overview. This SLA provides performance metrics and provisions for Unified Communications and Collaboration
Guide to migrating to SIP Whitepaper August 2011 Trefor Davies, CTO of Timico http://www.trefor.net/ Session Initiation Protocol or SIP is the VoIP standard used by most telecommunications service providers
SIP Trunking Guide: Get More For Your Money 07/17/2014 WHITE PAPER Overview SIP trunking is the most affordable and flexible way to connect an IP PBX to the Public Switched Telephone Network (PSTN). SIP
Xorcom IP-PBX Software s Based on the Elastix Asterisk i distribution, Xorcom s entire family of IP-PBX appliances provide all the standard telephone functionality supported by Asterisk at no extra cost,
ESI SIP Trunking Installation Guide 0450-1227 Rev. B Copyright 2009 ESI (Estech Systems, Inc.). Information contained herein is subject to change without notice. ESI products are protected by various U.S.
Information Crib Sheet Internet Access Service Agreement 1. Definitions and Interpretation This Service Agreement is to be read in conjunction with the Conditions for Communications Services (the Conditions
CTS White Paper Page 1 of 11 Converged Telephony Solution Technical White Paper ٠ May 2004 CTS White Paper Page 2 of 11 Converged Telephony Solution White Paper The focus of this white paper is to explain
White Paper Voice over IP Networks: Ensuring quality through proactive link management Build Smarter Networks Table of Contents 1. Executive summary... 3 2. Overview of the problem... 3 3. Connectivity
SIP TRUNKING THE COST EFFECTIVE AND FLEXIBLE ALTERNATIVE TO ISDN A cost-effective alternative to ISDN that provides flexibility and continuity Reliable voice services SIP trunking is the fastest-growing
Receptionist Small Business Administrator Guide Revision 1.0 GCI Ltd Global House 2 Crofton Close Lincoln Lincolnshire LN3 4NT www.gcicom.net Copyright GCI 2012 GCI Receptionist Small Business - Administrator