Application Note. ShoreTel / Ingate / Verizon Business SIP Trunking. 09 June 2017 Version 1 Issue 2

Transcription

1 Product: ShoreTel Ingate Verizon Business I n n o v a t i o n N e t w o r k A p p N ote TPP Date: October, 2012 System version: ShoreTel 13.x Application e ShoreTel / Ingate / Verizon Business SIP Trunking 09 June 2017 Version 1 Issue 2

2 Table of Contents APPLICATION NOTE... 1 SHORETEL / INGATE / VERIZON BUSINESS SIP TRUNKING OVERVIEW DOCUMENT FEEDBACK DOCUMENT CHANGE HISTORY SIP TRUNKING ARCHITECTURE OVERVIEW INTEROPERABILITY REQUIREMENTS, CERTIFICATION AND LIMITATIONS VERSION SUPPORT LIST OF COMPLIANT HARDWARE ShoreTel System Ingate SIParator or Firewall Voice switches Phones CERTIFICATION RESULTS SUMMARY LIMITATIONS AND OBSERVATIONS SHORETEL SHORETEL UNSUPPORTED FEATURES SHORETEL CONFIGURATION Call Control Settings Sites Settings SITES EDIT SCREEN ADMISSION CONTROL BANDWIDTH SITES EDIT SCREEN INTRA / INTER-SITE CALLS Switch Settings - Allocating Ports System Settings Trunk Groups System Settings Individual Trunks INGATE INGATE PRODUCT INFORMATION INGATE PRODUCT CONFIGURATION Startup Tool INGATE TROUBLESHOOTING Call Flow Examples Startup Tool Ingate Configuration Ingate Troubleshooting Tools INGATE SALES & TECHNICAL SUPPORT Sales Technical Support DOCUMENT AND SOFTWARE COPYRIGHTS TRADEMARKS DISCLAIMER COMPANY INFORMATION... 93

3 ShoreTel tests and validates the interoperability of the Member's solution with ShoreTel's published software interfaces. ShoreTel does not test, nor vouch for the Member's development and/or quality assurance process, nor the overall feature functionality of the Member's solution(s). ShoreTel does not test the Member's solution under load or assess the scalability of the Member's solution. It is the responsibility of the Member to ensure their solution is current with ShoreTel's published interfaces. The ShoreTel Technical Support organization will provide Customers with support of ShoreTel's published software interfaces. This does not imply any support for the Member's solution directly. Customers or reseller partners will need to work directly with the Member to obtain support for their solution. 1 Overview This document provides details for connecting the ShoreTel system though the Ingate SIParator to Verizon Business for SIP Trunking to enable audio communications. The document specifically focuses on the configuration procedures needed to set up these systems to interoperate. SIP Trunking allows the use of Session Initiation Protocol (SIP) communications from an Internet Telephony Service Provider (ITSP) instead of the typical analog, Basic Rate Interface (BRI), T1 or E1 trunk connections. Having the pure IP trunk to the ITSP allows for more control and options over the communication link. This application note provides the details on connecting the ShoreTel IP phone system through an Ingate SIParator which is connected to both the LAN and WAN and acts as a secure gateway to Verizon Business for SIP Trunking. ShoreTel and Ingate have teamed up to build a solid security focused solution, ShoreTel being the IP PBX which sits on the LAN and connects to the Ingate SIParator / Firewall. Providing a solution to allow customers the ability to connect to SIP Trunks offered by Verizon Business in a secure manner is important. The Ingate then is connected to not only the LAN but also the WAN, providing the typical firewall security abilities but also intelligent SIP routing and such SIP features as: Registration Digest Authentication Dial Plan Modification Back to Back User Agent (Terminates SIP messaging on both LAN and WAN side for SIP Protocol Normalization) Transfer conversion of SIP REFER to SIP reinvite messaging Quick configuration templates for each of the certified ITSPs

4 1.1 Document Feedback ShoreTel IP PBX administrators who would like to provide feedback on the contents of this document should send it to 1.2 Document Change History Version 1 Issue 1 Version 1 Issue 2 10/19/2011; Initial Draft 05/07/2012; Configuration Updates 1.3 SIP Trunking Architecture Overview There are a number of different network deployments of SIP Trunking from a Service Provider to an Enterprise. Some Service Providers provide SIP Trunking directly over the Internet and others over provide managed link into their network. Here are two typical deployments examples:

5 2 Interoperability Requirements, Certification and Limitations 2.1 Version Support Products are certified via the Innovation Network Validation Process for the ShoreTel system. The table below contains the matrix of Ingate Firewall and Ingate SIParator versions firmware releases certified on the identified ShoreTel software releases. Ingate Firewall and Ingate SIParator version ShoreTel 13.x 2.2 List of compliant hardware The following hardware while not necessarily used during the validation itself is guaranteed to comply with the testing results of this application note ShoreTel System The ShoreTel system used during the certification process is a ShoreWare Director Enterprise Edition Server consisting of ShoreTel 13.0 version. S P E C I F I C A T I O N S: Minimum Hardware Requirements Pentium E2160 Dualcore 1.8 GHz 1GB RAM 3.5 GB hard disk space for software 30 MB hard disk space per hour of voic storage

6 100 Base-T Ethernet NIC Software Requirements Microsoft Windows Server 2003 R2 SP2 Standard and Enterprise Microsoft Windows Server 2008 SP2 Standard and Enterprise A ShoreTel Small Business Edition Server is also compatible with Verizon Business. Small Business Edition Integrated Server specifications Celeron 1.8 GHz CPU 2 GB RAM 80 GB hard disk or better DVD-ROM drive 10/100 Ethernet NIC Microsoft Windows Server 2003, Telecommunications System Ingate SIParator or Firewall The Ingate system used during validation is a Firewall 1190 consisting of version Ingate products are compatible with communications equipment from other vendors and service providers who support the SIP Protocol. The Ingate products are a security device designed to sit on the Enterprise network edge, an ICSA Labs Certified security product, focused on SIP communications security and network security for the Enterprise. Ingate products are designed to solve the issues related to SIP traversing the NAT (Network Address Translation) which is a part of all enterprise class firewalls. The NAT translates between the public IP address(es) of the enterprise, and the private IP addresses which are only known inside the LAN. These private IP addresses are created to enable all devices to have an IP address, and also provide one of the security layers of the enterprise network. In addition, the Ingate products provide routing rules to assign to SIP traffic flow to ensure only allowed SIP traffic will pass. In the Verizon Business deployment you can use either a SIParator or Firewall product. The following devices are supported Voice switches ShoreGear 30 ShoreGear 50 ShoreGear 50v ShoreGear 90 ShoreGear 90v ShoreGear 90BRI ShoreGear 90BRIv ShoreGear 120 ShoreGear 220T1 ShoreGear 220T1a ShoreGear 220E1

7 2.2.4 Phones Regardless the technology of phones connected to the ShoreTel system, all officially supported phones may place calls over a Verizon SIP trunk & will demonstrate the same behavior (features & limitations) as captured in this application note ShoreTel IP phones ShorePhone IP 110 ShorePhone IP115 ShorePhone IP212k ShorePhone IP230 ShorePhone IP230g ShorePhone IP265 ShorePhone IP560 ShorePhone IP560g ShorePhone IP565 ShorePhone IP655 ShorePhone IP8000 ShorePhone BB SIP phones All officially ShoreTel Innovation Network Certified SIP phones are also compliant with the test results documented in this application note Analog Phones Any analog phones connected to the following ShoreGear voice switches will be compliant with the test results documented in this application note. ShoreGear 30 ShoreGear 50 ShoreGear 50v ShoreGear 90 ShoreGear 90v ShoreGear 90BRI ShoreGear 90BRIv ShoreGear 120 ShoreGear 220T1a 2.3 Certification Results Summary Test Plan for Verizon Business Retail VoIP Trunking Service Table 1: Security Test Case

8 ID Name Description es 1 Layer-2 IPsec Authentication For public Internet customers, IPsec tunnels must be implemented on the IP transport network between Verizon's Retail VoIP Interop lab firewall and the customer. This IPsec tunnel may be supported by a router or firewall on the customer premises. The tunnel can be configured to support either AH (Authenticated Header) or ESP (Encapsulating Security Payload) modes. with the Ingate Firewall only Table 2: DNS-SRV Test Case ID Name Description es 2 DNSSRV Verify the ShoreTel IP-PBX/SBC will Service obey the weights, priorities, and ports of with Ingate SIParator or Protocols/Port the DNS SRV response as defined in Firewall Adherence RFC Table 3: Inbound Call Test Cases (PSTN to ShoreTel) ID Name Description es 3 Inbound Call Loop Avoidance Verification Inbound Call with Originator (PSTN) release Inbound Call with Terminator (CPE) release Inbound Call Hang-up during ring phase (Cancel Call) Verify that an inbound call sent to the ShoreTel IP-PBX will respond with a 4XX, 5XX or 6XX error message, even though the TN is not defined/registered on the IP-PBX. This scenario can occur during initial provisioning and when TN s are ported. Verify that an inbound call connects and terminates properly when the originating device releases call. Verify that an inbound call connects and terminates properly (i.e. the SIP and RTP messaging) when the terminating device releases call. Verify that an inbound call terminates properly when the originating phone hangs up before terminating device answers. The ShoreTel system will terminate the call to the Trunk Group default destination (usually Auto Attendant menus)

9 ID Name Description es 7 Inbound Call Verify that an inbound call to a DID that Customer Phone terminates on the ShoreTel IP-PBX will The ShoreTel system will not terminate with the proper '40X' error terminate the Registered/Online message, however, the DID is not call to the defined/registered on the IP-PBX. Trunk Group default destination (usually Auto Attendant Inbound Calling Line Identification (Caller ID) Inbound Call Waiting Support Inbound G.711 Fax Inbound T.38 Fax Inbound Call from PSTN with Privacy Requested Unscreened ANI using Diversion header Unscreened ANI using P-Asserted- Identity Verify that the ShoreTel IP-PBX can receive the calling number identity in the SIP 'From' header. can provide call waiting. can receive from a PSTN Fax machine using the G.711 codec. Verify the ShoreTel IP-PBX/SBC can receive a fax transmission from a PSTN Fax machine using T.38 negotiation. Verify the ShoreTel IP-PBX/SBC receives no calling information when Blocking is activated on the PSTN (CPN = Restricted) Verify that an outbound call terminate properly with the correct CLI displayed as the Caller ID when the Screened Telephone Number contained in the Diversion Header is different from the CLI contained in the From Header. Verify that an outbound call terminate properly with the correct CLI displayed as the Caller ID when the Screened Telephone Number contained in the P- Asserted-Identity Header is different from the CLI contained in the From Header. menus) Table 4: Outbound Call Test Cases (ShoreTel to PSTN) ID Name Description es Outbound Call with Originator (CPE) Release Outbound Call with Terminator (PSTN) Release Verify that an outbound call terminates properly (i.e., the SIP and RTP messaging) when the originating device releases call. Verify that an outbound call terminates properly (i.e., the SIP and RTP messaging) when the terminating device releases call.

10 ID Name Description es Outbound Hangup During Ring Phase Outbound 1+10 Digit Call Terminates Outbound International Call Outbound 311 Non Emergency Calls Outbound Directory Assistance Outbound 411 Directory Assistance Outbound 1411 Directory Assistance Outbound 711 Telephone Relay Services (Hearing Impaired Services) 911 Emergency Service Support Outbound 511 Information Line Outbound Toll-Free Call Operator Services Call (0+ Local) Operator Assistance Support (0+ Toll) Operator Assistance (0 Minus) Operator Assistance (00 Minus) Verify that an outbound call terminates properly when the originating phone hangs up before terminating device answers. can properly terminate a '1' + ten-digit outbound call to the PSTN. can properly terminate an International outbound call to the PSTN. can properly terminate '311' nonemergency outbound calls to the PSTN. can properly terminate Directory Assistance outbound calls to the PSTN. can properly terminate '411' (directory assistance) outbound calls to the PSTN. can properly terminate 1411 Directory Assistance outbound calls to the PSTN. can properly terminate 711 Telephone Relay Service outbound calls to the PSTN. can properly terminate 911 emergency outbound calls to the PSTN. can properly terminate 511 Information outbound calls to the PSTN. can properly terminate 8XX Toll Free outbound calls to the PSTN. can properly terminate Local Operator Assisted Calls. can properly terminate Long Distance Operator Assisted Calls. can properly terminate Operator Assisted Calls. can properly terminate Operator Assisted Calls.

11 ID Name Description es Operator Assistance 32 (01+ International) can properly terminate Operator Outbound G.711 Fax Outbound T.38 Fax Outbound Calling Line Identifier (Caller ID) Outbound Fast Answer Outbound Call to PSTN with Privacy Requested Calling Party Number not Provisioned UDP for SIP SDP Support (RFC 2327) RTP and RP (RFC 3550) SIP Headers 18x Behavior Assisted Calls. can terminate outbound calls to a Fax Machine. can send outbound T.38 Fax to a PSTN Fax Machine. Verify that Caller ID is correctly delivered to a PSTN endpoint Verify that a call is handled properly when there is an immediate answer or a '200OK' is received before a '180' or '183' message. Verify the ShoreTel IP-PBX can withhold the calling line's identity from a PSTN user. Verify that the proper failure code is sent when a specified calling party number has not been provisioned in Verizon's application server (BroadSoft). uses UDP transport for SIP. complies with RFC 2327 for SDP. complies with RFC 3550 and RFC 3351 for RTP and RP and to verify that RTP is sent symmetrically. Verizon sends full headers. Verizon receives full and compact headers. ShoreTel IP-PBX/SBC must be able to receive full headers (i.e., the SIP From header can be represented with the full form, From:, or the compact from, f:.) Verizon responds to INVITE requests with 183 Session Progress messages with SDP, or 180 Ringing without SDP. Verizon will not send a 180 message with SDP. For INVITEs sent to the retail customer, the customer may respond with 183 Session Progress and SDP for media, or 180 Ringing without SDP. Retail Customers cannot send 180 with SDP. e 3 Table 5: Protocol Test Cases

12 ID Name Description es 44 UDP for SIP uses UDP transport for SIP. SDP Support (RFC ) complies with RFC 2327 for SDP. RTP and RP 46 Support (RFC complies with RFC 3550 for RTP and ) SIP Headers 18x Behavior RP. Verizon sends full headers. Verizon receives full and compact headers. Retail customers must be able to receive full headers (i.e., the SIP 'From' header can be represented with the full form, 'From:', or the compact form, 'f:'.) Verizon responds to INVITE requests with 183 Session Progress messages with SDP, or 180 Ringing without SDP. Verizon will not send a 180 message with SDP. For INVITEs sent to the retail customer, the customer may respond with 183 Session Progress and SDP for media, or 180 Ringing without SDP. Retail Customers cannot send 180 with SDP. 302 Behavior The objective of this test case is to verify that Verizon will not send a 302 response (move temporarily) to the retail customer. The retail customer should never send a 302 response to Verizon. Diversion Header DTMF (RFC 2833) Outbound DTMF (RFC 2833) Inbound Support for Offer / Answer with SDP (RFC 3264) Verify that in call forwarding scenarios the ShoreTel IP-PBX and Verizon correctly populate the 'Diversion' header and 'From' header to match the redirecting number and calling party number respectively. The ShoreTel IP-IPBX/SBC should support sending DTMF tones per RFC 2833 when using the G.729 codec for an outbound call. The ShoreTel IP-PBX/SBC should support RFC 2833 for DTMF when utilizing the G.729 codec for an inbound call. supports RFC 3264 (Offer/Answer with SDP).

13 ID Name Description es Call Hold (RFC 3264) (ShoreTel to PSTN) Media Inactivity Use of FQDN IP Addressing in SIP Messaging Call hold using RFC 3264 methods (a=sendonly) is supported (and is the [referred hold method). Call hold using RFC 2543 methods (c= ) is supported (but is not preferred). Verify that a call is handled and terminates properly when it is placed on hold for a period of time. When the customer deploys two IP- PBXs/EGWs/SBCs (with associated unique IP addresses), the host portion of the headers containing CLI (Diversion, P-Asserted-ID, and From) must contain the FQDN shared by the two gateways, instead of the gateway s IPv4 address. The customer must also accept inbound calls with FQDN in the host portion of Request-URI in this configuration. The ShoreTel system will first send a=inactive with Ingate SIParator or Firewall Table 7: Media Test Cases ID Name Description es G.711u-law Codec (AVT Payload Type 0) G.729 and G.729a Support Codec Negotiation Early Media Support can support G.711 u-law AVT payload=0 with 20ms packetization. Appendix I and II is not supported. Verizon can receive G.729 or G.729a (AVT payload 18). Verizon sends G.729a. G.729 and G.729a use 20ms packetization. Verify that the ShoreTel IP-PBX selects the correct codec based on what is offered to it. supports Early Media. Table 8: Differentiated Services (DiffServ) Test Cases ID Name Description es 61 RTP Media Marked with Verify that the ShoreTel IP-PBX sets the RTP with the DSCP to EF or CS5 to 62 DSCP EF or CS5 SIP Signaling Marked with DSCP AF32 or CS3 ensure QoS across the network. sets the SIP IP packets with the correct DSCP binary AF32 or CS3 to ensure QoS across the network. with Ingate

14 Table 9: Attended Call Transfer Test Cases ID Name Description es ShoreTel Calls PSTN Attended Transfer to ShoreTel Extension ShoreTel Calls PSTN Attended Transfer to PSTN PSTN Calls ShoreTel Attended Transfer to ShoreTel Extension PSTN Calls ShoreTel Attended Transfer to PSTN Verify that the ShoreTel IP-PBX can properly call the PSTN and then transfer the call to another IP-PBX/SBC phone line. Verify that the ShoreTel IP-PBX can call the PSTN and then transfer the call to another PSTN phone line. phone line can properly receive an inbound call from a PSTN and then transfer the call to another IP-PBX/SBC phone line. phone line can properly receive an inbound call from PSTN and then transfer the call to another PSTN phone. Table 10: Semi-Attended Call Transfer Test Cases ID Name Description es ShoreTel Calls PSTN Semi- Attended Transfer to ShoreTel Extension ShoreTel Calls PSTN Semi- Attended Transfer to PSTN PSTN Calls ShoreTel Semi- Attended Transfer to ShoreTel Extension PSTN Calls ShoreTel Semi- Attended Transfer to PSTN phone line can properly call the PSTN and then transfer (semi-attended) the call to another IP-PBX/SBC phone line. phone line can properly call the PSTN and then transfer (semi-attended) the call to another PSTN phone line. phone line can properly receive an inbound call from PSTN and then transfer (semi-attended) the call to another IPPBX/SBC phone line. phone line can properly receive an inbound call from PSTN and then transfer (semi-attended) the call to another PSTN phone. Table 11: Blind Call Transfer Test Cases ID Name Description es 71 ShoreTel Calls PSTN Blind Transfer to a ShoreTel Extension phone line can properly call the PSTN and then transfer (blind) the call to another IP-PBX/SBC phone line.

15 ID Name Description es ShoreTel Calls PSTN Blind Transfer to PSTN PSTN Calls ShoreTel Blind Transfer to ShoreTel Extension PSTN Calls ShoreTel Blind Transfer to PSTN phone line can properly call the PSTN and then transfer (blind) the call to another PSTN phone line. phone line can properly receive an inbound call from PSTN and then transfer (blind) the call to another IP- PBX/SBC phone line. phone line can properly receive an inbound call from PSTN and then transfer (blind) the call to another PSTN phone. Table 12: Attended Call Transfer Test Cases ID Name Description es ShoreTel IP-PBX calls PSTN attended transfer to ShoreTel IP-PBX extension ShoreTel IP-PBX calls PSTN attended transfer to PSTN PSTN calls ShoreTel IP-PBX attended transfer to ShoreTel IP-PBX PSTN calls ShoreTel IP-PBX attended transfer to PSTN Verify that the ShoreTel IP-PBX can properly call the PSTN and then transfer the call to another IP-PBX/SBC phone line. Verify that the ShoreTel IP-PBX can call the PSTN and then transfer the call to another PSTN phone line. phone line can properly receive an inbound call from a PSTN and then transfer the call to another IP-PBX/SBC phone line. phone line can properly receive an inbound call from a PSTN and then transfer the call to another PSTN phone. Table 13: Semi-Attended Call Transfer Test Cases ID Name Description es ShoreTel Calls PSTN Semi- Attended Transfer to ShoreTel Extension ShoreTel Calls PSTN Semi- Attended Transfer to PSTN phone line can properly call the PSTN and then transfer (semi-attended) the call to another IP-PBX/SBC phone line. phone line can properly call the PSTN and then transfer (semi-attended) the call to another PSTN phone line.

16 ID Name Description es PSTN Calls ShoreTel Semi- Attended Transfer to ShoreTel Extension PSTN Calls ShoreTel Semi- Attended Transfer to PSTN phone line can properly receive an inbound call from PSTN and then transfer (semi-attended) the call to another IPPBX/SBC phone line. phone line can properly receive an inbound call from PSTN and then transfer (semi-attended) the call to another PSTN phone. Table 14: Blind Call Transfer Test Cases ID Name Description es ShoreTel Calls PSTN Blind Transfer to a ShoreTel Extension ShoreTel Calls PSTN Blind Transfer to PSTN PSTN Calls ShoreTel Blind Transfer to ShoreTel Extension PSTN Calls ShoreTel Blind Transfer to PSTN phone line can properly call the PSTN and then transfer (blind) the call to another IP-PBX/SBC phone line. phone line can properly call the PSTN and then transfer (blind) the call to another PSTN phone line. phone line can properly receive an inbound call from PSTN and then transfer (blind) the call to another IP- PBX/SBC phone line. phone line can properly receive an inbound call from PSTN and then transfer (blind) the call to another PSTN phone. Table 15: Conference Call Test Cases ID Name Description es ShoreTel IP-PBX Calls PSTN Conference to ShoreTel Extension ShoreTel IP-PBX Calls PSTN Conference to PSTN PSTN Calls ShoreTel IP-PBX Conference to ShoreTel IP-PBX Extension phone line can properly call the PSTN and conference in another IP-PBX/SBC phone line. phone line can properly call the PSTN and conference in another PSTN phone line. phone line can properly receive an inbound call from a PSTN and conference in another IP-PBX/SBC phone line.

17 ID Name Description es 90 PSTN Calls ShoreTel IP-PBX Conference to PSTN phone line can properly receive an inbound call from a PSTN and conference in another PSTN phone. Table 16: CPE Failover Behavior Test Cases ID Name Description es 91 OPTIONS Method Request and Verify that the ShoreTel CPE devices send proper responses to the OPTIONS Response Round Robin (Load Share 50/50 Between the two CPE s) Primary / Secondary Failover (Hunt) request. Verify proper traffic load balance between 2CPE when the SBC is configured in round-robin mode. Verify proper failover between 2CPE when the SBC is configured in hunt mode and a failure happens on either one of the CPEs. with Ingate Both CPE Fail Verify '2CPE' failure behavior. with Ingate Table 17: Ambient Noise Test Cases ID Name Description es 95 Ambient Noise ShoreTel to PSTN Verify that Re-invites on detection of fax tone are handled properly by the 96 Ambient Noise PSTN to ShoreTel ShoreTel IP-PBX. Verify that Re-invites on detection of fax tone are handled properly by the ShoreTel IP-PBX. Test Plan for Verizon Business IP Enabled Contact Center (IPCC) Service Table 28: Security Test Cases

18 1 Layer-2 IPsec Authentication For non-registering Public Internet customers, IPsec Authentication tunnels must be implemented on the IP transport network between Verizon s IPCC interface proxies and the ShoreTel IP PBX. This IPsec tunnel may be supported by a router or firewall on the customer premises. IPsec tunnels for Verizon Business s IPCC service are configured to use Authentication Header (AH) and Encapsulating Security Payload (ESP) protocols. Customers should have port 500 open for IPsec tunnels and they must be used for SIP messaging only. IPCC supports the IKE v1 key distribution method with preshared keys for IPsec. The IPsec keys are shared out of band as part of a contract agreement. with Ingate Firewall only Table 19: Options Method Request and Response Test Cases ID Name Description es 2 Options Method Request and Response Verify that the ShoreTel IP PBX sends proper responses to the OPTIONS request. Table 20: IP Toll Free Test Cases 3 Simulated CPE Failures Exercise numerous customer system failures to determine the level of network awareness of various failures. Table 21: Inbound Call Test Cases (Verizon Business PSTN to ShoreTel) 4 Inbound Calls with Request-URI Set to Customer s Provisioned URL Address Verify that the ShoreTel IP PBX / SBC will accept inbound calls where the Request-URI is set to the customer s provisioned URL address. The general template definition of a customer s provisioned address is: <user>@<site or instance id>.<customer>.21sip.com To reach a SIP termination the Request- URI can contain various alphanumeric characters, but the user can not be listed in the E.164 format3 which includes a leading +.

19 Inbound Call with Originator (PSTN) Release Inbound Call with Terminator (SIP) Release Inbound Call with Disconnect During Ring Phase (Cancel Call) Inbound Call with Customer Phone Registered with SIP PBX Inbound Call with Ring No Answer Inbound Call with User Busy Inbound Call with CPN Allowed Privacy Null Verify that an inbound call terminates properly (i.e. that appropriate SIP messaging is sent and RTP transmission ends) when the originating PSTN device releases the call. Verify that an inbound call terminates properly (i.e. that appropriate SIP messaging is sent and RTP transmission ends) when the terminating SIP device releases the call. Verify that an inbound call terminates properly when the originating phone disconnects before terminating device answers. Verify when the customer s end device is not registered with the ShoreTel IP PBX, an inbound call routing to the SIP trunk will terminate with the proper response message. Verify proper call termination when there is no answer at the target device. Validate the proper busy message is generated when the target device is busy. Verify that the ShoreTel IP PBX can utilize Caller-ID information from the SIP messages to provide Caller-ID to the ShoreTel IP phone. In this scenario, the calling party will not restrict presentation. In the PSTN, this results in the calling party number (CPN) presentation being Allowed and subsequently the SIP Privacy setting will be Null. (This means the Privacy header may be omitted completely from the message; or if included, the Privacy header field value will be set to none. ) In this case, Verizon Business will pass the Calling Line Identification (CLI) information to the ShoreTel system in the PAsserted-Identity or the From headers. The ShoreTel system will terminate the call to the default Trunk Group destination (usually Auto Attendant Menu) e 1 e 2

20 Inbound Call with CPN Restricted Privacy id Inbound Call with Long Duration Inbound Call with Proprietary Headers Verify that the ShoreTel IP PBX can utilize Caller-ID information from the SIP messages to indicate Calling Party Number (CPN) presentation was restricted and an appropriate message is displayed on the ShoreTel IP phone. In this scenario the calling party will restrict presentation. In the PSTN, this results in the calling party number (CPN) presentation being restricted and subsequently the SIP INVITE will contain a Privacy header with a field value of id. However, because this is a Toll Free call paid for by the called party who subscribes to delivery of calling party information, passing of the calling party number is allowed. In this case, Verizon Business will pass the Calling Line Identification (CLI) information to the customer s equipment in the P-Asserted-Identity header. In addition the Privacy header will be included with a field value of id and the From header will contain anonymous. Verify that an inbound call that has been connected for 20 minutes or more is not inappropriately terminated prior to calling or called party actions. Additionally, this case is to validate that the appropriate SIP messages are sent and RTP is terminated when the calling or called party disconnects the call. When SIP Proprietary Headers have been enabled, the Proprietary Headers will be sent in the INVITE to the ShoreTel IP-PBX. This scenario is used to validate that the CPE can appropriately receive an INVITE with the Proprietary Headers without failing the call.

21 15 Agent to Agent Proprietary Headers The only Agent to Agent Proprietary Header that is supported is the User-to- User Header supported for Avaya CPE. This Header is supported by default and is not dependent on any settings in Toll Free Network Manager (TFNM). When Party B sends a REFER4 or REFER with Replaces to the Service Controller and includes the User-to-User header, it will be sent to Party C in the INVITE or INVITE with Replaces. N/A Table 22: Network Call Redirect (NCR) with no Enhanced Transfer Test Cases ID Name Description es 16 Inbound Call with NCR with Answer Verify that an inbound call terminates properly when Network Call Redirect (NCR) has been enabled and there is no Enhanced Transfer provisioned on the Inbound Call with NCR with Ring No Answer Inbound Call with NCR with User Busy Termination point. Verify that the ShoreTel IP PBX does not accept a call in a Ring No Answer (RNA) situation in which the Agent does not answer the call. In this case the Network Call Redirect (NCR) function will be set to send the call to a secondary location after expiration of a RNA timer provisioned in the application. Validate the proper 486 Busy Here message is generated when the target device is busy and that the SC can take the appropriate NCR actions upon receipt of the message. Table 23: Outbound Call Test Cases (ShoreTel to Verizon Business) e 2

22 ID Name Description es Load Sharing 19 For IDA customers, IPCC interface proxies can be resolved using DNS SRV and DNS A records. Given the DNS SRV record is XXXXX.com5, the IPCC customer resolves the DNS SRV query _sip._udp.xxxxx.com. The customer may cache the DNS response for the duration of the TTL in the DNS response. It is the ShoreTel systems responsibility to determine when a call attempt to a Verizon Business proxy fails. Further, it is the ShoreTel system responsibility to attempt other proxies in the DNS response if the first proxy attempted fails. IPCC customers must obey the weights, priorities, and ports of the DNS SRV response as defined in RFC with Ingate SIParator or Firewall 20 Failover For IDA customers, IPCC interface proxies can be resolved using DNS SRV and DNS A records. Given the DNS SRV record is XXXXX.com6, the IPCC customer resolves the DNS SRV query _sip._udp.xxxxx.com. The customer may cache the DNS response for the duration of the TTL in the DNS response. It is the ShoreTel systems responsibility to determine when a call attempt to a Verizon Business proxy fails. Further, it is the ShoreTel system responsibility to attempt other proxies in the DNS response if the first proxy attempted fails. IPCC customers must obey the weights, priorities, and ports of the DNS SRV response as defined in RFC with Ingate SIParator or Firewall

23 ID Name Description es Outbound Call with CPN Allowed Privacy Null Outbound Call with CPN Restricted Privacy id Verify that the ShoreTel IP PBX sends P-Asserted-Identity information, the Privacy header is either omitted or the field value is absent, and the From header contains Party B s caller-id information. Subsequently the called party should receive the calling party number and if caller-id is enabled the display should display Party B s information. Execution of this scenario should occur after an inbound call is received to ensure the outbound call is allowed. As specified in the IPCC Customer contract, privacy requested on agent to agent calls within the customer domain in advance of an Attended Transfer is not allowed and will not be supported by the service. Table 24: Protocol Test Cases ID Name Description es SIP Methods (RFC 3261) UDP for SIP and Long Message Support SIP Ports CPE Must Accept Receipt of Full and Compact Headers 18x Behavior Verify that the ShoreTel IP PBX supports the INVITE, ACK, BYE, REFER, NOTIFY, CANCEL, OPTIONS, and SUBSCRIBE methods. Verify that the ShoreTel IP PBX uses UDP transport for SIP. Also when necessary, fragmented internet datagrams are supported to handle large signaling messages that exceed the path MTU. Verify that the ShoreTel IP PBX uses the appropriate ports. At this time IPCC infrastructure only sends full headers. However, IPCC customers must be able to receive full and compact headers. Verizon Business responds to INVITE requests with 183 Session Progress messages with SDP, or 180 Ringing without SDP. Verizon Business will not send a 180 message with SDP. For INVITEs sent to the IPCC customer, the CPE may respond with 183 Session Progress and SDP for media, or 180 Ringing without SDP. IPCC customers cannot send 180 with SDP.

24 ID Name Description es Behavior Verify that Verizon Business and the ShoreTel IP PBX never send a Draft-levy-sipdiversion-08 Diversion Header Call Hold Media Inactivity (Call Hold Long Duration) Moved Temporarily response. Verify that the ShoreTel IP PBX does not send a Diversion header. Call Hold using RFC 3264 methods (i.e. a=sendonly, a=recvonly, and a=inactive) is supported (and is the preferred hold method). Call Hold using RFC 2543 methods (c= ) is supported. Customer initiated holds may use either of these methods (RFC 3264 is preferred). Customers must be able to at least handle receipt of RFC 2543 method. Verify which method is being used by the CPE and that it functions correctly. Verify that a call is handled and terminates properly when it is placed on hold for an extended period of time. Configurable via SIP Profile Table 25: Media Test Cases ID Name Description es SDP Support Verify that the ShoreTel IP PBX 32 (RFC 2327) complies with RFC 2327 for SDP Verizon Business SDP Offer Customer SDP Answer Customer SDP Offer Verizon Business SDP Answer Verifying RTP Basic Inbound Call Verify SDP information from previous tests. The ShoreTel IP PBX needs to provide a valid SDP answer with the desired IPCC supported codec. Verify SDP information from previous tests. The ShoreTel IP PBX needs to provide a valid SDP Offer with the desired IPCC supported codec. Specific test cases that will allow the customer to make an SDP offer are call holds, transfers, and calls with NCR capabilities. Verify that the RTP stream matches what was negotiated in the SDP.

25 ID Name Description es Verifying RTP Inbound Call with Enhanced Transfer DTMF (RFC 2833) Verifying RTP Phone on Mute RTP and RP Support (RFC 3550) Early Media Verify that the RTP stream matches what was negotiated in the SDP. With Enhanced Transfer DTMF capabilities, the initial stage of the transferred call to a party without Enhanced Transfer capabilities the call must be G.711 μ- law, after the call is answered, the SDP may be re-negotiated. This will remove the MS from the RTP path and my also result in a codec change. Since ShoreTel IP PBX supports G.729, this codec may be used in the second stage of the call. Verify that the ShoreTel IP PBX supports a configurable AVT payload type for RFC Verify that the ShoreTel IP PBX obeys the annexb=no parameter information. Comfort Noise (CN) packets must not be included Discontinuous Transmission (DTX) must not be performed. Verify that the ShoreTel IP PBX complies with RFC 3550 for RTP and RP. Verify that the ShoreTel IP PBX supports Early Media. Table 26: IP Toll Free Transfer Test Cases ID Name Description es 41 PSTN to SIP UA to SIP UA 42 PSTN to SIP UA to PSTN Verify that the ShoreTel IP PBX can properly receive an inbound Toll Free call from the PSTN and then transfer the call to another SIP PBX using Basic Blind REFER transfer methods. The media stream between the endpoints is through the Network Gateway (NGW). Verify that the ShoreTel IP PBX can properly receive an inbound Toll Free call from the PSTN and then transfer the call to a PSTN phone using Basic Blind REFER transfer methods. The media stream between the endpoints is through the Network Gateway (NGW). with Ingate with Ingate

26 ID Name Description es PSTN to SIP UA 43 to SIP UA PSTN to SIP UA to PSTN PSTN to SIP UA to SIP UA PSTN to SIP UA to PSTN PSTN to SIP UA to SIP UA Verify that the ShoreTel IP PBX phone line can properly receive an inbound Toll Free call from the PSTN and then transfer the call to another SIP PBX phone using Basic Blind REFER transfer methods. The media stream between the endpoints is through the Network Gateway (NGW). Verify that the ShoreTel IP PBX phone line can properly receive an inbound Toll Free call from the PSTN and then transfer the call to a PSTN phone using Basic Blind REFER transfer methods. The media stream between the endpoints is through the Network Gateway (NGW). Verify that a basic blind transfer, REFER method, call terminates properly when Party A, the transferee (PSTN phone), disconnects before the transferred-to device answers. Verify that a basic blind transfer, REFER method, call terminates properly when Party A, the transferee (PSTN phone), disconnects before the transferred-to device answers. Verify that the ShoreTel IP PBX phone line can properly receive an inbound Toll Free call from the PSTN and then transfer the call to another SIP PBX phone using Basic Blind REFER transfer methods. The media stream between the endpoints is through the Network Gateway (NGW). with Ingate with Ingate with Ingate with Ingate with Ingate 48 PSTN to SIP UA to PSTN Verify that the ShoreTel IP PBX phone line can properly receive an inbound Toll Free call from the PSTN and then transfer the call to a PSTN phone using Basic Blind REFER transfer methods. The media stream between the endpoints is through the Network Gateway (NGW). with Ingate

27 ID Name Description es PSTN to SIP UA 49 to SIP UA PSTN to SIP UA to PSTN PSTN to SIP UA to SIP UA PSTN to SIP UA to PSTN PSTN to SIP UA to SIP UA PSTN to SIP UA to PSTN PSTN to SIP UA to SIP UA PSTN to SIP UA to PSTN Verify that the ShoreTel IP PBX phone line can properly receive an inbound Toll Free call from the PSTN and then transfer the call to another SIP PBX phone using Basic Blind REFER transfer methods. The media stream between the endpoints is through the Network Gateway (NGW). Verify that the ShoreTel IP PBX phone line can properly receive an inbound Toll Free call from the PSTN and then transfer the call to a PSTN phone using Basic Blind REFER transfer methods. The media stream between the endpoints is through the Network Gateway (NGW). Verify that a Basic blind transfer, REFER method, call terminates properly when Party A, the transferee (PSTN phone), disconnects before the transferred-to device answers. Verify that a Basic blind transfer, REFER method, call terminates properly when Party A, the transferee (PSTN phone), disconnects before the transferred-to device answers. Verify that the ShoreTel IP PBX phone line can properly receive an inbound call from a Toll Free PSTN caller and then transfer the call to another SIP PBX phone line using basic attended REFER with Replaces transfer method. The media stream between the endpoints is through the Network Gateway (NGW). Basic attended transfer to the PSTN is not supported. Verify that the ShoreTel IP PBX phone line can properly receive an inbound call from a Toll Free PSTN caller and then transfer the call to another SIP PBX phone line using basic attended REFER with Replaces transfer method. The media stream between the endpoints is through the Network Gateway (NGW). Basic attended transfer to the PSTN is not supported. with Ingate with Ingate with Ingate with Ingate with Ingate Blocked with Ingate Blocked

28 ID Name Description es Party A Disconnects Before B Sends REFER with Replaces Party B Disconnects Without Sending REFER with Replaces PSTN to SIP UA to SIP UA PSTN to SIP UA to SIP UA PSTN to SIP UA to SIP UA Verify that the calls will terminate properly if Party A disconnects before Party B sends a REFER with Replaces. Verify that the calls will terminate properly if Party B disconnects without sending a REFER with Replaces. Verify that the customer s SIP PBX phone line can properly receive an inbound call from a Toll Free PSTN caller and then transfer the call to another SIP PBX phone line using Enhanced Transfer function invoked by pressing DTMF digits. The media stream between the endpoints is through the Media Server (MS). Verify that the ShoreTel IP PBX phone line can properly receive an inbound call from a Toll Free PSTN caller and then transfer the call to another SIP PBX phone line using Enhanced Transfer function invoked by pressing DTMF digits. The media stream between the initial endpoints and the initial connection of the final endpoints are through the Media Server and upon completion of the transfer sequence the media stream between the final endpoints does not involve the Media Server. Verify that the ShoreTel IP PBX phone line can properly receive an inbound call from a Toll Free PSTN caller and then transfer the call to another SIP PBX phone line using Enhanced Transfer function invoked by pressing DTMF digits. The media stream between the endpoints is through the Verizon Business Media Server. with Ingate with Ingate

29 ID Name Description es 62 PSTN to SIP UA to SIP UA Verify that the ShoreTel IP PBX phone line can properly receive an inbound call from a Toll Free PSTN caller and then transfer the call to another SIP PBX phone line using Enhanced Transfer function invoked by pressing DTMF digits. The media stream between the initial endpoints and the initial connection of the final endpoints are through the Media Server and upon completion of the transfer sequence the media stream between the final endpoints does not involve the Media Server. Table 27: IVR Test Case ID Name Description es 63 Simulated CPE Failures Exercise numerous customer system failures to determine the level of network awareness of various failures. Table 28: Inbound Calls (Verizon Business PSTN to ShoreTel) Test Cases ID Name Description es Inbound Call with Originator (PSTN) Release Inbound Call with Terminator (SIP) Release Inbound Call with Disconnect During Ring Phase (Cancel Call) Inbound Call with Customer Phone Registered with SIP PBX Inbound Call with Ring No Answer Timer Expire Inbound Call with User Busy Verify that an inbound call terminates properly (i.e. that appropriate SIP messaging is sent and RTP transmission ends) when the originating PSTN device releases the call. Verify that an inbound call terminates properly (i.e. that appropriate SIP messaging is sent and RTP transmission ends) when the terminating SIP device releases the call. Verify that an inbound call terminates properly when the originating phone disconnects before terminating device answers. Verify when the customer s end device is not registered with ShoreTel IP PBX, an inbound call routing to the SIP trunk will terminate with the proper response message. Verify proper call termination when there is no answer at the target device. Verify the proper busy message is generated when the target device is busy. e 1 e 2

30 ID Name Description es Inbound Call with CPN Allowed Privacy Null Inbound Call with CPN Restricted Privacy id Verify that the ShoreTel IP PBX can utilize Caller-ID information from the SIP messages to provide Caller-ID to the customer s phone. In this scenario, the calling party will not restrict presentation. In the PSTN, this results in the calling party number (CPN) presentation being Allowed and subsequently the SIP Privacy setting will be Null. (This means the Privacy header may be omitted completely from the message; or if included, the Privacy header field value will be set to none. ) In this case, Verizon Business will pass the Calling Line Identification (CLI) information to the customer s equipment in the PAsserted-Identity or the From headers. The Privacy header will either be omitted or included. If the Privacy header is included it will have a field value of none. Verify that the ShoreTel IP PBX can utilize Caller-ID information from the SIP messages to indicate Calling Party Number (CPN) presentation was restricted and an appropriate message is displayed on the customer s phone. In this scenario the calling party will restrict presentation. In the PSTN, this results in the calling party number (CPN) presentation being restricted and subsequently the SIP INVITE will contain Privacy header with a field value of id. However, because this is a Toll Free call paid for by the called party who subscribes to delivery of calling party information, passing of the calling party number is allowed. In this case, Verizon Business will pass the Calling Line Identification (CLI) information to the customer s equipment in the P-Asserted-Identity header. In addition the Privacy header will be included with a field value of id and the From header will contain anonymous.

31 ID Name Description es 72 Inbound Call with Long Duration Verify that an inbound call that has been connected for 20 minutes or more is not inappropriately terminated prior to calling or called party actions. Additionally, this case is to validate that the appropriate SIP messages are sent and RTP is terminated when the calling Agent to Agent Proprietary Headers Inbound Call with NCR with Answer Inbound Call with NCR with Ring No Answer Inbound Call with NCR with User Busy or called party disconnects the call. IP IVR can support various Agent to Agent Proprietary Headers based on customer specific scripting. One example of this is the User-to-User Header supported for Avaya CPE. If appropriately scripted in IP IVR, this Header is supported. When Party B sends a REFER or REFER with Replaces to the Service Controller and includes the User-to-User header, it will be sent to Party C in the INVITE or INVITE with Replaces. When the customer has IP IVR with Network Call Redirect (NCR) enabled, an inbound call will need to be tested to validate the CPE can appropriately handle the SIP signaling. Verify that the ShoreTel IP PBX does not accept a call in a Ring No Answer (RNA) situation in which the Agent does not answer the call. In this case the Network Call Redirect (NCR) function will be set to send the call to a secondary location after expiration of a RNA timer provisioned in the application. Verify that the proper 486 Busy Here message is generated when the target device is busy and that the IP IVR can take the appropriate NCR actions upon receipt of the message. with Ingate e 1 e 2

32 ID Name Description es 77 Inbound Call with Release Link Trunking (RLT) Release Link Trunking (RLT) is a function that allows IP IVR to remove the Media Server from the RTP path, even though the SIP signaling continues to pass through IP IVR. This capability is based on a function in the exit Service Independent Building Block (SIBB). IP IVR generally is limited to use of G.711 because the Media Server is engaged. However, in this particular case when codec re-negotiation occurs to drop out the Media Server it is possible to use G.729. Table 29: Outbound Calls (ShoreTel to Verizon Business) Test Cases ID Name Description es SIP Methods (RFC 3261) UDP for SIP and Long Message Support Verify that the ShoreTel IP PBX supports the INVITE, ACK, BYE, REFER, NOTIFY, CANCEL, OPTIONS, and SUBSCRIBE methods. IPCC customers with either PIP access or IDA must always use UDP to transport SIP signaling. (P is not allowed.) To support large signaling messages (i.e. message sizes that exceed the path maximum transmission unit [MTU]), fragmented packets are supported. The objective of this test is to verify that the ShoreTel IP PBX uses UDP transport for SIP. Also when necessary, fragmented internet datagrams are supported to handle large signaling messages that exceed the path MTU.

33 ID Name Description es 80 SIP Ports Verizon Business Session Border Controllers (SBCs) listen on UDP port 5070 or above. This port value is assigned by Verizon Business at the time of customer provisioning for IPCC customers utilizing PIP access. Verizon Business IPCC SIP servers listen on UDP port 5060 which would apply to IPCC customers utilizing IDA. At a minimum customer devices must accept SIP signaling on port It is also recommended that the customer sends SIP signaling from port The objective of this test is to verify the customer s SIP PBX uses the CPE Must Accept Receipt of Full and Compact Headers 18x Behavior appropriate ports. At this time IPCC infrastructure only sends full headers. However, IPCC customers must be able to receive full and compact headers. Verizon Business responds to INVITE requests with 183 Session Progress messages with SDP, or 180 Ringing without SDP. Verizon Business will not send a 180 message with SDP. For INVITEs sent to the IPCC customer, the customer may respond with 183 Session Progress and SDP for media, or 180 Ringing without SDP. IPCC customers cannot send 180 with SDP Behavior Verify that Verizon Business and the IPCC customer never send a 302 Moved Temporarily response. Draft-levy-sipdiversion-08 Diversion Header Call Hold Verify that the ShoreTel SIP PBX does not send a Diversion header. Call Hold using RFC 3264 methods (i.e. a=sendonly, a=recvonly, and a=inactive) is supported (and is the preferred hold method). Call Hold using RFC 2543 methods (c= ) is supported. Customer initiating holds may use either of these methods (RFC 3264 is preferred). Customers must be able to at least handle receipt of RFC 2543 method. The objective of this test case is to verify which method is being used by the CPE and that it functions correctly.

34 ID Name Description es 86 Media Inactivity (Call Hold Long Duration) Verify that a call is handled and terminates properly when it is placed on hold for an extended period of time. Table 30: Media Test Cases ID Name Description es SDP Support (RFC Verify the customer s SIP PBX ) complies with RFC 2327 for SDP Verizon Business SDP Offer Customer SDP Answer Customer SDP Offer Verizon Business SDP Answer Verifying RTP Basic Inbound Call Verifying RTP Inbound Call with NCR Verify SDP information from previous tests. The ShoreTel SIP PBX needs to provide a valid SDP answer with their desired IPCC supported codec. Verify SDP information from previous tests. The ShoreTel SIP PBX needs to provide a valid SDP Offer with their desired IPCC supported codec. Specific test cases that will allow the customer to make an SDP offer are call holds, Transfers, and calls with NCR capabilities. Verify that the RTP stream matches what was negotiated in the SDP. Verify that the RTP stream matches what was negotiated in the SDP. With NCR capabilities, the initial stage of the call must be G.711 μ-law, after the call is answered, the SDP may be renegotiated. This will remove the MS from the RTP path and may also result in a codec change. If the customer supports G.729, this codec may be used in the second stage of the call. If the customer s SIP PBX is G.711 μ-law only, the codec will not change. The RTP steam needs to be checked in order to verify that the correct codec is being used at both stages of the call DTMF (RFC 2833) Verifying RTP Phone on Mute Verify that the ShoreTel SIP PBX supports a configurable AVT payload type for RFC Verify that the ShoreTel SIP PBX obeys the annexb=no parameter contained in the SDP information. Comfort Noise (CN) packets must not be included in the G.729 RTP payload and Discontinuous Transmission (DTX) must not be performed.

35 ID Name Description es RTP and RP Verify that the ShoreTel SIP PBX 94 Support (RFC complies with RFC 3550 for RTP and ) Early Media RP. Verify that the ShoreTel SIP PBX supports Early Media PSTN to SIP UA to SIP UA PSTN to SIP UA to PSTN PSTN to SIP UA to SIP UA PSTN to SIP UA to PSTN PSTN to SIP UA to SIP UA Verify that the ShoreTel SIP PBX phone line can properly receive an inbound Toll Free call from the PSTN and then transfer the call to another SIP PBX phone using Blind REFER transfer methods. The media stream between the endpoints is through the Network Gateway (NGW) and anchored in the Verizon Business Media Server. Verify that the ShoreTel SIP PBX phone line can properly receive an inbound Toll Free call from the PSTN and then transfer the call to a PSTN phone using Blind REFER transfer methods. The media stream between the endpoints is through the Network Gateway (NGW) and anchored in the Verizon Business Media Server. Verify that the ShoreTel SIP PBX phone line can properly receive an inbound Toll Free call from the PSTN and then transfer the call to another SIP PBX phone using Blind REFER transfer methods. The media stream between the endpoints is through the Network Gateway (NGW) and anchored in the Verizon Business Media Server. Verify that the ShoreTel SIP PBX phone line can properly receive an inbound Toll Free call from the PSTN and then transfer the call to another PSTN phone using Blind REFER transfer methods. The media stream between the endpoints is through the Network Gateway (NGW) and anchored in the Verizon Business Media Server. Verify that a basic blind transfer, REFER method, call terminates properly when Party A, the transferee (PSTN phone), disconnects before the transferred-to device answers. with Ingate with Ingate with Ingate with Ingate with Ingate

36 ID Name Description es PSTN to SIP UA 101 to PSTN PSTN to SIP UA to SIP UA PSTN to SIP UA to PSTN PSTN to SIP UA to SIP UA PSTN to SIP UA to PSTN PSTN to SIP UA to SIP UA PSTN to SIP UA to PSTN Verify that a basic blind transfer, REFER method, call terminates properly when Party A, the transferee (PSTN phone), disconnects before the transferred-to device answers. Verify that the ShoreTel SIP PBX phone line can properly receive an inbound call from a Toll Free PSTN caller and then transfer the call to another SIP PBX phone line using REFER transfer method. The media stream between the endpoints is through the Verizon Business s IP IVR/MS proxy. Verify that the ShoreTel SIP PBX phone line can properly receive an inbound call from a Toll Free PSTN caller and then transfer the call to another PSTN phone line using REFER transfer method. The media stream between the endpoints is through the Verizon Business s IP IVR/MS proxy. Verify that the ShoreTel SIP PBX phone line can properly receive an inbound Toll Free call from the PSTN and then transfer the call to another SIP PBX phone using Blind REFER transfer methods. The media stream between the endpoints is through the Network Gateway (NGW) and anchored in the Verizon Business Media Server. Verify that the ShoreTel SIP PBX phone line can properly receive an inbound Toll Free call from the PSTN and then transfer the call to another PSTN phone using Blind REFER transfer methods. The media stream between the endpoints is through the Network Gateway (NGW) and anchored in the Verizon Business Media Server. Verify that a basic blind transfer, REFER method, call terminates properly when Party A, the transferee (PSTN phone), disconnects before the transferred-to device answers. Verify that a basic blind transfer, REFER method, call terminates properly when Party A, the transferee (PSTN phone), disconnects before the transferred-to device answers. with Ingate with Ingate with Ingate with Ingate with Ingate with Ingate with Ingate

37 ID Name Description es 108 PSTN to SIP UA to SIP UA Verify that the ShoreTel SIP PBX phone line can properly receive an inbound Toll Free call from the PSTN and then transfer the call to another SIP PBX phone using DTMF digits to invoke the transfer. Depending on the negotiated SDP, the customer may send the DTMF digits either encoded using RFC 2833 or PSTN to SIP UA to PSTN PSTN to SIP UA to SIP UA PSTN to SIP UA to PSTN PSTN-SIP UA-SIP UA PSTN to SIP UA to PSTN Party A Disconnects Before B Sends REFER with Replaces Party B Disconnects Without Sending REFER with Replaces in the RTP 0 media stream. Verify that the ShoreTel SIP PBX phone line can properly receive an inbound Toll Free call from the PSTN and then transfer the call to another PSTN phone using DTMF digits to invoke the transfer. Depending on the negotiated SDP, the customer may send the DTMF digits either encoded using RFC 2833 or in the RTP 0 media stream. Verify that the ShoreTel SIP PBX phone line can properly receive an inbound call from a Toll Free PSTN caller and then transfer the call to another SIP PBX phone line using Attended REFER with Replaces transfer method. Attended transfer to the PSTN is not supported. Verify that the ShoreTel SIP PBX phone line can properly receive an inbound call from a Toll Free PSTN caller and then transfer the call to another SIP PBX phone line using Attended REFER with Replaces transfer method. Attended transfer to the PSTN is not supported. Verify that the calls will terminate properly if Party A disconnects before Party B sends a REFER with Replaces. Verify that the calls will terminate properly if Party B disconnects without sending a REFER with Replaces. with Ingate Blocked with Ingate Blocked with Ingate with Ingate

38 ID Name Description es 116 PSTN to SIP UA to SIP UA Verify that the ShoreTel SIP PBX phone line can properly receive an inbound Toll Free call from the PSTN and then transfer the call to another SIP PBX phone using DTMF digits to invoke the Attended Transfer. This test may be run two different ways depending on the customer s configuration, encoding the DTMF digits per RFC 2833, or sending 117 PSTN to SIP UA to PSTN then in the RTP 0 media stream. Verify that the ShoreTel SIP PBX phone line can properly receive an inbound Toll Free call from the PSTN and then transfer the call to another PSTN phone using DTMF digits to invoke the Attended Transfer. Depending on the negotiated SDP, the customer may send the DTMF digits either encoded using RFC 2833 or in the RTP 0 media stream. Test Plan for Verizon Business EMEA Retail VoIP Trunking Service Table 31: Security Test Case A Layer-2 IPSec Authentication (Mandatory) Table 32: DNS-SRV Test Case B DNS-SRV Service Protocols/Port Adherence For public Internet customers, IPSec tunnels must be implemented on the IP transport network between Verizon's Retail VoIP Interop lab firewall and the customer. This IPSec tunnel may be supported by a router or firewall on the customer premises. The tunnel can be configured to support either AH (Authenticated Header) or ESP (Encapsulating Security Payload) modes. will obey the weights, priorities, and ports of the DNS SRV response as defined in RFC with Ingate Firewall only with Ingate SIParator or Firewall

39 Table 33: Inbound Call Test Cases (Verizon Business PSTN to ShoreTel) ID Name Description es 1 Inbound Call Loop Avoidance Verification Verify that an inbound call sent to the ShoreTel IP-PBX will respond with a 4XX, 5XX or 6XX error message, even though the TN is not defined/registered on the IP-PBX. This scenario can occur during initial provisioning and when 2 Inbound - Hang Up During Ring Phase (Call Canceled) 3 Inbound Call Customer Phone not Registered/Online 4 Inbound - PSTN to Customer CPN Presentation = Allowed, Terminator Release TN s are ported. Verify that an inbound call terminates properly when the originating phone hangs up before terminating device answers. Verify that an inbound call to a DID that terminates on the ShoreTel IP-PBX will terminate with the proper '40X' error message, however, the DID is not defined/registered on the IP-PBX. Verify that the ShoreTel IP PBX can utilize Caller-ID information from the SIP messages to provide Caller-ID to the ShoreTel IP phone. In this scenario, the calling party will not restrict presentation. 5 Inbound - Fax can terminate an inbound fax to a DID number. 6 7 Inbound Call from PSTN with Privacy Requested Inbound Busy Line 8 Inbound Call - Ring No Answer (RNA) 9 Inbound - Long Duration Call with Originator Release Verify the ShoreTel IP-PBX/SBC receives no calling information when Blocking is activated on the PSTN (CPN = Restricted) Verify the proper busy message is generated when the target device is busy. Verify proper call termination when the there is no answer at the target device. Verify that an inbound call, which has been connected for at least 20 minutes, terminates properly. The ShoreTel system will terminate the call to the Trunk Group default destination (usually Auto Attendant menus) e 2 e 1

40 ID Name Description es Inbound - G.711 CODEC Negotiation Inbound - G.729 CODEC Negotiation Inbound - Customer is Offhook Inbound - DTMF (RFC2833) Inbound - Ambient Noise PSTN to CPE Inbound Call Forward to CPE Using SIP Diversion Header Verify that an inbound call call from the PSTN to the EMEA Retail VoIP phone terminates properly and that the customer s preferred G.711 CODEC is negotiated in the SDP regardless of what is preferred by Verizon s PSTN gateway and that there is voice path in both directions. Verify that an inbound call from the PSTN to the EMEA Retail VoIP phone terminates properly and that the customer s preferred G.729 CODEC is negotiated in the SDP regardless of what is preferred by Verizon s PSTN gateway and that there is voice path in both directions. Verify that an inbound call from PSTN to the EMEA Retail VoIP phone that is off-hook returns the proper busy message. Verify that the ShoreTel IP-PBX supports RFC 2833 for DTMF when utilizing the G.729 codec for an inbound call. Verify that the ShoreTel IP-PBX will accept a SIP Re-Invite sent from VZB upon detecting of a high frequency tone after call establishment and properly renegotiate the codec to G.711ulaw. This Test Case apply only to customer s using G.729 as the preferred codec. Verify that in call forwarding scenarios with the ShoreTel IP-PBX and Verizon correctly populate the 'Diversion' header and 'From' header to match the redirecting number and calling party number respectively. e 2 Table 34: Outbound Call Test Cases (ShoreTel to PSTN) ID Name Description es Outbound Hangup During Ring Phase (Call Canceled) Outbound - Local Geographic National PSTN Call Originator Release Verify that an outbound call terminates properly when the originating phone hangs up before terminating device answers. Verify that an outbound local call terminates properly (i.e., the SIP and RTP messaging) when the originating device releases call.

41 ID Name Description es Outbound - Geographic National PSTN Call Terminator Release Outbound - National Cell Call Outbound - International PSTN Call Outbound - Short Dial Number Calls Outbound - Emergency Services Call (Police, EMS/Fire) Outbound - Freephone Call (080X) Outbound - Business Rate Services (08XX) Outbound Fax Outbound - Fast Answer Call Outbound - Call with Privacy Asserted (VoIP to PSTN) Outbound - CPN Provisioned in Verizon Proxy Verify that an outbound national call terminates properly (i.e., the SIP and RTP messaging) when the terminating device releases call. Verify that an outbound national cellular call terminates properly (i.e., the SIP and RTP messaging) when the originating device releases call. can properly terminate an International outbound call to the PSTN, when the originating device releases call. can properly terminate Short Dial nonemergency outbound calls to the PSTN. Short Dialed test numbers will depend on the specific country of the customer. Verify that the ShoreTel IP-PBX can properly terminate emergency outbound calls to the PSTN. Emergency Service test numbers will depend on the specific country of the customer. Verify that the ShoreTel IP-PBX can properly terminate an outbound 080X Toll Free call to the PSTN, when the originating device releases call. Verify that the ShoreTel IP-PBX can properly terminate an outbound 08XX call to the PSTN and is terminated to the Operator or ARU, when the originating device releases call. can terminate outbound calls to a Fax Machine. Verify that a call is handled properly when there is an immediate answer or a '200OK' is received before a '180' or '183' message. Verify the ShoreTel IP-PBX can withhold the calling line's identity from a PSTN user. Verify that the proper failure code is sent when a specified calling party number has not been provisioned in Verizon's proxy server.

42 ID Name Description es Outbound - Premium Rate 09XX Outbound - Long Duration Call with INFO Method Call Audit Outbound - Call Origination with 183 Session Progress (with SDP) Outbound - Call Forward to PSTN using SIP Diversion Header Outbound - Call Hold Outbound - DTMF (RFC2833) NTE Payload Negotiation Verify that the ShoreTel IP-PBX can properly terminate an outbound 09XX call to the PSTN, when the originating device releases call. Verify that an outbound national call terminates properly (i.e., the SIP and RTP messaging), and the call is active for more than 20 minutes, when the originating device releases call. Verify that an outbound national call terminates properly to the PSTN with 183 Session Progress messages when the originating device releases call. Verify that an inbound call to a user configured to always forward externally, that the outbound call s SIP Req-URI contains final destination PSTN number, the Diversion Header contains DID of SIP UA device forwarding the call and the From: header contains the Calling Party number of the Originating caller. Verify that an outbound national call to PSTN terminates properly and when placed on hold, utilizing RFC3264 methods (a=sendonly) is preferred, although RFC2543 methods (C= ) is supported, but not preferred. Then verify that Music on Hold is not activated and the media stream has stopped between the two SIP endpoints, when call is on Hold. Retrieve the call to re-activate the media path, then verify voice path in both directions. Verify that outbound calls terminate properly to the PSTN which terminates to an interactive response unit requiring DTMF input such as a bank, voic , reservation system, etc. Verify the call is answered and the DTMF menu is received, enter the DTMF digits to execute menu instructions. Verify the menus can be accessed. The receiving endpoint must reply with NTE payload type sent by initiator. For example, if the initiating UAC sends payload type 101, the receiving UAS must reply with payload type 101.

43 ID Name Description es Outbound - G711 Verify that the ShoreTel IP-PBX can 35 CODEC properly negotiate G711 codec on Negotiation Outbound - G729 CODEC Negotiation Outbound - Ring No Answer (RNA) Outbound - Ambient Noise CPE to PSTN outbound calls. Verify that the ShoreTel IP-PBX can properly negotiate G729 codec on outbound calls. Verify proper call termination when the there is no answer at the terminating device Verify that the ShoreTel IP-PBX will accept a SIP Re-Invite sent from VZB upon detecting of a high frequency tone after call establishment and properly renegotiate the codec to G.711ulaw. This Test Case apply only to customer s using G.729 as the preferred codec. Table 35: Protocol Test Cases ID Name Description es 39 UDP for SIP uses UDP transport for SIP. SDP Support (RFC ) RTP and RP (RFC 3550) SIP Headers 18x Behavior complies with RFC 2327 for SDP. complies with RFC 3550 and RFC3351 for RTP and RP and to verify that RTP is sent symmetrically. Verizon sends full headers. Verizon receives full and compact headers. Retail customers must be able to receive full headers (i.e., the SIP 'From' header can be represented with the full form, 'From:', or the compact form, 'f:'.) Verizon Business responds to INVITE requests with 183 Session Progress messages with SDP, or 180 Ringing without SDP. Verizon Business will not send a 180 message with SDP. For INVITEs sent to the retail customer, the CPE may respond with 183 Session Progress and SDP for media, or 180 Ringing without SDP. Retail customers cannot send 180 with SDP. 302 Behavior Verify that Verizon will not send a 302 response (move temporarily) to the retail customer. Support for Offer / Answer with SDP (RFC3264) should support RFC 3264 (Offer/Answer with SDP).

44 ID Name Description es 46 Media Inactivity Verify that a call is handled and terminates properly when it is placed on 47 Use of FQDN IP Addressing in SIP Messaging hold for an extended period of time. When the customer deploys two IP- PBXs/EGWs/SBCs (with associated unique IP addresses), the host portion of the headers containing CLI (Diversion, P-Asserted-ID, and From) must contain the FQDN shared by the two gateways, instead of the gateway s IPv4 address. The customer must also accept inbound calls with FQDN in the host portion of Request-URI in this configuration. with Ingate Table 36: Differentiated Services (DiffServ) Test Cases ID Name Description es 48 RTP Media Marked with Verify that the ShoreTel IP-PBX sets the RTP with the DSCP to EF or CS5 to 49 DSCP EF or CS5 SIP Signaling Marked with DSCP AF32 or CS3 ensure QoS across the network. sets the SIP IP packets with the correct DSCP binary AF32 or CS3 to ensure QoS across the network. with Ingate Table 37: Attended Call Transfer Test Cases ID Name Description es ShoreTel IP-PBX calls PSTN attended transfer to ShoreTel IP-PBX extension ShoreTel IP-PBX calls PSTN attended transfer to PSTN PSTN calls ShoreTel IP-PBX attended transfer to ShoreTel IP-PBX PSTN calls ShoreTel IP-PBX attended transfer to PSTN Verify that the ShoreTel IP-PBX can properly call the PSTN and then transfer the call to another IP-PBX/SBC phone line. Verify that the ShoreTel IP-PBX can call the PSTN and then transfer the call to another PSTN phone line. phone line can properly receive an inbound call from a PSTN and then transfer the call to another IP-PBX/SBC phone line. phone line can properly receive an inbound call from a PSTN and then transfer the call to another PSTN phone.

45 Table 38: Semi-Attended Call Transfer Test Cases ID Name Description es ShoreTel Calls PSTN Semi- Attended Transfer to ShoreTel Extension ShoreTel Calls PSTN Semi- Attended Transfer to PSTN PSTN Calls ShoreTel Semi- Attended Transfer to ShoreTel Extension PSTN Calls ShoreTel Semi- Attended Transfer to PSTN phone line can properly call the PSTN and then transfer (semi-attended) the call to another IP-PBX/SBC phone line. phone line can properly call the PSTN and then transfer (semi-attended) the call to another PSTN phone line. phone line can properly receive an inbound call from PSTN and then transfer (semi-attended) the call to another IPPBX/SBC phone line. phone line can properly receive an inbound call from PSTN and then transfer (semi-attended) the call to another PSTN phone. Table 39: Blind Call Transfer Test Cases ID Name Description es ShoreTel Calls PSTN Blind Transfer to a ShoreTel Extension ShoreTel Calls PSTN Blind Transfer to PSTN PSTN Calls ShoreTel Blind Transfer to ShoreTel Extension PSTN Calls ShoreTel Blind Transfer to PSTN phone line can properly call the PSTN and then transfer (blind) the call to another IP-PBX/SBC phone line. phone line can properly call the PSTN and then transfer (blind) the call to another PSTN phone line. phone line can properly receive an inbound call from PSTN and then transfer (blind) the call to another IP- PBX/SBC phone line. phone line can properly receive an inbound call from PSTN and then transfer (blind) the call to another PSTN phone.

46 Table 40: Attended Call Transfer Test Cases ID Name Description es ShoreTel IP-PBX calls PSTN attended transfer to ShoreTel IP-PBX extension ShoreTel IP-PBX calls PSTN attended transfer to PSTN PSTN calls ShoreTel IP-PBX attended transfer to ShoreTel IP-PBX PSTN calls ShoreTel IP-PBX attended transfer to PSTN Verify that the ShoreTel IP-PBX can properly call the PSTN and then transfer the call to another IP-PBX/SBC phone line. Verify that the ShoreTel IP-PBX can call the PSTN and then transfer the call to another PSTN phone line. phone line can properly receive an inbound call from a PSTN and then transfer the call to another IP-PBX/SBC phone line. phone line can properly receive an inbound call from a PSTN and then transfer the call to another PSTN phone. Suppprted Table 41: Semi-Attended Call Transfer Test Cases ID Name Description es ShoreTel Calls PSTN Semi- Attended Transfer to ShoreTel Extension ShoreTel Calls PSTN Semi- Attended Transfer to PSTN PSTN Calls ShoreTel Semi- Attended Transfer to ShoreTel Extension PSTN Calls ShoreTel Semi- Attended Transfer to PSTN phone line can properly call the PSTN and then transfer (semi-attended) the call to another IP-PBX/SBC phone line. phone line can properly call the PSTN and then transfer (semi-attended) the call to another PSTN phone line. phone line can properly receive an inbound call from PSTN and then transfer (semi-attended) the call to another IPPBX/SBC phone line. phone line can properly receive an inbound call from PSTN and then transfer (semi-attended) the call to another PSTN phone.

47 Table 42: Blind Call Transfer Test Cases ID Name Description es ShoreTel Calls PSTN Blind Transfer to a ShoreTel Extension ShoreTel Calls PSTN Blind Transfer to PSTN PSTN Calls ShoreTel Blind Transfer to ShoreTel Extension PSTN Calls ShoreTel Blind Transfer to PSTN phone line can properly call the PSTN and then transfer (blind) the call to another IP-PBX/SBC phone line. phone line can properly call the PSTN and then transfer (blind) the call to another PSTN phone line. phone line can properly receive an inbound call from PSTN and then transfer (blind) the call to another IP- PBX/SBC phone line. phone line can properly receive an inbound call from PSTN and then transfer (blind) the call to another PSTN phone. Table 43: Conference Call Test Cases ID Name Description es ShoreTel IP-PBX Calls PSTN Conference to ShoreTel Extension ShoreTel IP-PBX Calls PSTN Conference to PSTN PSTN Calls ShoreTel IP-PBX Conference to ShoreTel IP-PBX Extension PSTN Calls ShoreTel IP-PBX Conference to PSTN phone line can properly call the PSTN and conference in another IP-PBX/SBC phone line. phone line can properly call the PSTN and conference in another PSTN phone line. phone line can properly receive an inbound call from a PSTN and conference in another IP-PBX/SBC phone line. phone line can properly receive an inbound call from a PSTN and conference in another PSTN phone. Table 44: CPE Failover Behavior Test Cases ID Name Description es 78 OPTIONS Method Request and Response Verify that the ShoreTel CPE devices send proper responses to the OPTIONS request.

48 ID Name Description es Round Robin (Load Share 50/50 Between the two CPE s) Primary / Secondary Failover (Hunt) Verify proper traffic load balance between 2CPE when the SBC is configured in round-robin mode. Verify proper failover between 2CPE when the SBC is configured in hunt mode and a failure happens on either one of the CPEs. with Ingate Both CPE Fail Verify '2CPE' failure behavior. with Ingate e 1: ShoreTel user extension s Call Handling Mode must be configured to Never forward incoming calls in order for the system to not answer an incoming call. e 2: ShoreTel user extension must be configured for Extension Only, have a Call Stack of 1 and be on an actual call in order to generate a busy tone. e 3: ShoreTel system supports T.38 faxing, but for outbound calls it requires that the answering device re-negotiate for T Limitations and Observations TBD 3 ShoreTel The configuration information below shows examples for configuring the ShoreTel, Ingate and Verizon Business network. Even though configuration requirements can vary from setup to setup, the information provided in these steps, along with the ShoreTel Planning and Installation Guide (or the Administration Guide) and documentation provided by Ingate and the Verizon Business network should prove to be sufficient. However every design can vary and some may require more planning than others. 3.1 ShoreTel Unsupported Features Please refer to the ShoreTel Administration Guide, Chapter 18 Session Initiation Protocol, for supported and unsupported features via SIP Trunks. Following are some feature limitations via SIP Trunks: General Feature Limitations Fax redirect not supported via SIP Trunks using G.711 (though Direct Inward Dialing (DID) to fax endpoint is supported) ShoreTel supports Music On Hold (MOH) over SIP trunks. The maximum number of music on hold (MOH) streams that a SIP-enabled switch can

49 support varies with the switch model. The range of such streams across all the voice switch models is Limitation: MOH source needs be on SIP trunk switch. If the ShoreTel server has a conference bridge 4.2 installed, you should not enable SIP. The conference bridge is not compatible with a ShoreTel system that has SIP enabled due to the dynamic RTP port required for SIP. ShoreTel supports the Service Appliance (SA-100) conferencing / IM system from Release -12. SIP trunk calls from / to the SA-100 is supported. The SA- 100 accepts access codes in DTMF RFC2833 only. 4 to 6 party conferences, when a SIP trunk is involved, utilize Make Me conference ports. Silent Monitoring, Barge-In, Silent Coach, Park/Unpark, Call recording features are supported on a SIP trunk call only if SIP trunk configured with Default ITSP SIP profile and the trunk is on a half-width switch. Silence detection on trunk-to-trunk transfers is not supported, it requires a physical trunk. There may be other feature limitations when using SIP Trunks. Please consult the ShoreTel Administration Guide for further details.

50 3.2 ShoreTel Configuration This section describes the ShoreTel system configuration to support SIP Trunking. The section is divided into general system settings and trunk configurations (both group and individual) needed to support SIP Trunking. e: ShoreTel basically just points its Individual SIP Trunks to the Ingate SIParator. The first settings to address within the ShoreTel system are the general system settings. These configurations include the Call Control, the Site and the Switch settings. If these items have already been configured on the system, skip this section and go on to the ShoreTel System Settings Trunk Groups section below Call Control Settings The first settings to configure within ShoreWare Director are the Call Control Options. To configure these settings for the ShoreTel system, log into ShoreWare Director and select Administration then Call Control followed by Options. Administration Call Control Options The Call Control Options screen will then appear.

51 Call Control Options Within the Call Control Options parameters; confirm that the appropriate settings are made for the DTMF Payload Type, Enable SIP Session Timer, DiffServ / ToS Byte (0-255) and Always Use Port 5004 for RTP parameters. The DTMF Payload Type (96-127) parameter defaults to a value of 102, you will need to ensure that you change this parameter to a value of 101 to interoperate with Verizon Business SIP Trunking Service. Once you modify this parameter you will need to reboot all of the ShoreTel IP Phones (and any Service Appliances), not rebooting the ShoreTel IP phones will cause the default value (102) to be utilized. The next step is to verify that the Enable SIP Session Timer box is disabled (unchecked). Configure the DiffServ / ToS Byte parameter to a setting of 184, this configures DiffServ appropriately for the Verizon network.

52 The last item is to verify that the Always Use Port 5004 for RTP is not enabled, if this is a new installation the option will be grayed out, do not modify this parameter if it is disabled. e: Disabling (un-checking) the parameter Always Use Port 5004 for RTP is required for implementing SIP on the ShoreTel system. For SIP configurations, Dynamic User Datagram Protocol (UDP) must be used for RTP Traffic. If the box is unchecked, MGCP will no longer use UDP port 5004; MGCP and SIP traffic will use dynamic UDP ports. Once this parameter is unchecked, make sure that everything (IP Phones, ShoreGear Switches, ShoreWare Director, Distributed Voice Services / Remote Servers, Conference Bridges and Contact Centers) are fully rebooted this is a one time only item. By not performing a full system reboot, one way audio will probably occur during initial testing Sites Settings The next settings to address are the administration of sites. These settings are modified under the ShoreWare Director by selecting Administration, then Sites. Administration Site This selection brings up the Sites screen. Within the Sites screen, select the name of the site to configure. The Edit Site screen will then appear.

53 e: Bandwidth of 2046 is just an example. Please refer to the ShoreTel Planning and Installation Guide for additional information on setting Admission Control Bandwidth. Sites Edit screen Admission Control Bandwidth The Admission Control Bandwidth defines the bandwidth available to and from the site. This is important as SIP trunk calls may be counted against the site bandwidth. Bandwidth needs to be set appropriately based on site setup and configuration with Verizon network. Please refer to the ShoreTel Planning and Installation Guide for additional information. Sites Edit screen Intra / Inter-Site Calls By default ShoreTel 13.0 has 11 built-in codecs, these codecs can be grouped as Codec Lists and defined in the sites page for Inter-site and Intra-site calls. Configure the "Intra-Site Calls" option to a Codec List that contains the desired codecs and save the change. When establishing calls with Verizon Business SIP Trunking, the preferred codec choice is G.729 (this means that you will need to modify or create an existing codec list and move the G.729 codec higher in the priority list). The site that the SIP Trunk Group belongs to will determine which Intra-Site Codec List will be utilized, be sure to move the desired codec up the list for higher priority. Please refer to the ShoreTel Planning and Installation Guide for additional information Switch Settings - Allocating Ports The final general settings to input are the ShoreGear switch settings. These changes are modified by selecting Administration, then Platform Hardware, then Voice Switches / Service Appliances followed by Primary in ShoreWare Director. Administration Switches This action brings up the Switches screen. From the Switches screen simply select the name of the switch to configure. The Edit ShoreGear Switch screen

54 will be displayed. Within the Edit ShoreGear Switch screen, select the desired number of SIP Trunks from the ports available. ShoreGear Switch Settings In this example we are using a ShoreGear 220T1 switch, which has Built-in Capacity, notice how we allocated 20 SIP trunk ports, by defining the value of 20 in the SIP Trunk area. Please be sure that the total equals 100, you can make all of the ports SIP trunks if needed. If you use a different model ShoreGear switch, some switches do not have the Built-in Capacity and you must designate the Port Type for 5 SIP Trunks each port designated as a SIP Trunk enables the support for 5 individual trunks. e: If you would like Music On Hold (MOH) to be played when calls are on hold, then the MOH source needs to be the same ShoreGear switch that is also hosting the SIP Trunks. ShoreTel 13 adds an additional option to the Port Type of half-width ShoreGear switches (the ShoreGear 220T1 has a parameter named Dedicate to SIP Trunk Media Proxy that will allow the entire DSP resources on the switch to be utilized for SIP media proxies). The new selection is SIP Media Proxy, it ensures that the ShoreTel system that is using SIP Trunks to have feature parity with PRI trunks. These include RFC 2833 DTMF detection for Office Anywhere External or Simultaneous Ring calls,

55 three party mesh conferencing (without needing to configure MakeMe conference ports), call recording, Silent Monitoring, Barge-In, Whisper Page, and Invites with no SDP when there s no common codec between ITSP and the local extension. For further information on SIP Media Proxy please refer to Chapter 18 of the ShoreTel 13 System Administration Guide System Settings Trunk Groups If the SIP Trunk Groups have already been configured on the system, skip down to the ShoreTel System Settings - Individual Trunks section. The settings for Trunk Groups are changed by selecting Administration, then Trunks followed by Trunk Groups within ShoreWare Director. Administration Trunk Groups This selection brings up the Trunk Groups screen. Trunk Groups Settings

56 From the pull down menus on the Trunk Groups screen, select the site desired and select the SIP trunk type to configure and click on the Go link from Add new trunk group at site:. The Edit SIP Trunk Group screen will appear. SIP Trunk Group Settings Within the Edit SIP Trunks Group screen define a name for the trunk group, in this example we chose Verizon Business. The Profile: parameter is drop down selection, select Verizon. The Enable Digest Authentication field is not required when connecting to an Ingate device. The Enable SIP Info for G.711 DTMF Signaling box should not be checked. Enabling SIP info is currently only used with tie trunks between ShoreTel systems. The next item to change in the Edit SIP Trunks Group screen is to make the appropriate settings for the Inbound: fields. Inbound

57 Within the Inbound: settings ensure the Number of Digits from CO is set to 10 for Verizon. Configure the DNIS or DID parameters as needed, we recommend that you enable them. The Extension parameter can be enabled or disabled, it doesn t make a difference with SIP Trunks (see Planning and Installation Guide for further information on configuration). Enable the Tandem Trunking parameter, this will allow the ShoreTel system to transfer calls to external parties via the SIP trunks. Select an appropriate User Group that has access to this Trunk Group being configured. The last item to define is the Destination parameter, this will determine where an inbound call is routed if there isn t a DNIS, DID or extension match, we chose the default Auto Attendant menu. Outbound and Trunk Services

58 If this trunk group is to be used for outbound calls, be sure to enable (check) the Outbound: parameter, then define an Access Code:, Local Area Code: and the Billing Telephone Number. Please refer to the ShoreTel Planning and Installation Guide for additional information. On the Trunk Services: section, make sure the appropriate services are enabled (checked) or disabled (unchecked) based on what the Verizon network supports and what features are needed from this Trunk Group. The one parameter that needs to be enabled is Enable Original Called Information, be sure to check the box to the left of this parameter. The last parameter Caller ID not blocked by default determines if the call is sent out as <unknown> or with caller information (Caller ID). User DID etc. will impact how information is passed out to the SIP Trunk group, this parameter needs to be enabled (checked). Trunk Digit Manipulation The parameter Remove leading 1 from 1+10D can be enabled or disabled. Verizon Business network accepts calls with our without a leading 1. After these settings are made to the Edit SIP Trunk Group screen, press the Save button to save the changes.

59 Trunk Group Dialing Rules Logout out of ShoreWare Director, then login using the Support Entry mode by holding down the CTRL + Shift keys and clicking on the U of the User ID: field. You should see the *** Support Entry *** appear on the page: ShoreWare Director Support Entry Log into ShoreWare Director with the normal administrator credentials. After a successful login, you will need to edit the SIP Trunk Group, created above, by selecting Administration, then Trunks followed by Trunk Groups within ShoreWare Director. Administration Trunk Groups This selection brings up the Trunk Groups screen.

60 Trunk Groups Settings Click on the SIP Trunk Group desired, in this case it was HQ Verizon SIP. The Edit SIP Trunk Group screen will appear. Scroll to the bottom of the page, to the Trunk Group Dialing Rules: section: SIP Trunk Group Dialing Rules Click on the Edit button to the right of the Custom: parameter. This action brings up the Trunk Group Dialing Rules pop-up window. Within the pop-up window enter the following string: ;10E This entry is case sensitive and must be entered exactly as noted, example is as follows: Trunk Group Dialing Rules Webpage Dialog

61 Click on the Save button, this completes the settings needed to set up the trunk groups on the ShoreTel system for operation with Verizon Business network System Settings Individual Trunks This section covers the configuration of the individual trunks. Select Administration, then Trunks followed by Individual Trunks to configure the individual trunks. Individual Trunks The Trunks by Group screen that is used to change the individual trunks settings then appears. Trunks by Group Select the site for the new individual trunk(s) to be added and select the appropriate trunk group from the pull down menu in the Add new trunk at site area. In this example, the site is HQ Verizon Labs and the trunk group is HQ Verizon SIP. Click on the Go link to bring up the Edit Trunk screen.

62 Edit Trunks Screen for Individual Trunks From the individual trunks Edit Trunk screen, input a name for the individual trunks, select the appropriate switch, select the SIP Trunk type and input the number of trunks. When selecting a name, the best practice is to name the individual trunks the same as the name of the trunk group so that the trunk type can easily be tracked. Select the switch upon which the individual trunk will be created. For the Verizon Business Network, select Use IP Address button and input the IP address of the Ingate product. The last step is to select the number of individual trunks desired (each one supports one audio path example if 30 is input, then 30 audio paths can be up at one time). Once these changes are complete, press the Save button to input the changes. e: Individual SIP Trunks cannot span networks. SIP Trunks can only terminate on the switch selected. There is no failover to another switch. For redundancy, two trunk groups will be needed with each pointing to another Ingate device just the same as if PRI were being used. After setting up the trunk groups and individual trunks, refer to the ShoreTel Product Installation Guide to make the appropriate changes for the User Group settings. This completes the settings for the ShoreTel system side

63 4 Ingate Ingate Systems AB is a Stockholm, Sweden based high-tech company that designs, develops, manufactures and markets leading data communications products for trusted Unified Communications. Ingate designed the world s first Session Initiation Protocol (SIP)-capable firewalls and SIParators, products that enable Unified Communications over the Internet. Unified Communications, with applications such as Internet telephony, presence indication, instant messaging, and audio/video conferencing, are modern and powerful business tools that enable enterprises to maintain reliable IP-communications internally and externally. As more businesses utilize these applications, service providers are offering SIP trunks to connect Local Area Networks to the outer world via Internet and/or dedicated, managed IP-lines. The enterprise Session Border Controller (Firewall) needs to manage all incoming and outgoing traffic securely. Authorized traffic based on SIP needs to pass through the Session Border Controller in a controlled manner reaching SIP units inside and outside the LAN. Ingate's Session Border Controllers are compatible with existing networks, and allow businesses to utilize the cost and time saving benefits of IP-based real-time communications with minimum investment. Ingate s leading products are marketed through world leading distributors, Value Added resellers and OEM s on all continents. 4.1 Ingate Product Information Ingate SIParator and Firewall products are compatible with communications equipment from other vendors and service providers who support the SIP Protocol. The Ingate products are a security device designed to sit on the Enterprise network edge, an ICSA Labs Certified security product, focused on SIP communications security and network security for the Enterprise. Ingate products are designed to solve the issues related to SIP traversing the NAT (Network Address Translation) which is a part of all enterprise class firewalls. The NAT translates between the public IP address(es) of the enterprise, and the private IP addresses which are only known inside the LAN. These private IP addresses are created to enable all devices to have an IP address, and also provide one of the security layers of the enterprise network. In addition, the Ingate products provide routing rules to flexibility in SIP traffic flows and ensure only allowed SIP traffic will pass. This provides an ability to route any call to any destination in a secure manner. The Ingate products also contain SIP Protocol normalization tools to assist in the interoperability of all of the different SIP vendors and service providers. Features

64 such as a B2BUA and other advanced customization tools allow integration with any other vendor. 4.2 Ingate Product Configuration The following section will briefly describe the configuration of the Ingate products. Further configuration of the Ingate products can be found under the Account Login page. Including a Configuration Guide for the Ingate with a ShoreTel when using SIP Trunking Startup Tool The Ingate Startup Tool is an installation tool for Ingate Firewall and Ingate SIParator products, facilitates the out of the box set up of SIP Trunking solutions with ShoreTel and various Internet Telephony Service Providers. Designed to simplify SIP trunk deployments, the tool will automatically configure a user s Ingate Firewall or SIParator to work with ShoreTel and the SIP Trunking service provider of your choice. With the push of a button, the configuration tool will automatically create a SIP trunk deployment designed to the user s individual setup. Users can select ShoreTel from a drop-down menu and the Internet Telephony Service Provider (ITSP) they use; the configuration tool will automatically apply the correct settings to the Ingate Firewall or SIParator to work seamlessly with that vendor or service provider. A list of SIP Trunking service providers that have demonstrated interoperability with the Ingate products is incorporated into the interface. Please note that not all SIP Trunking service providers listed in this interface have been certified by ShoreTel. Consult the ShoreTel Certified Technology Partner list of vendors for a current list. ( The configuration tool is available now as a free download for all Ingate Firewalls and SIParators. It can be found at Also available here is a Startup Tool Getting Started Guide to assist in using the Startup Tool.

65 Contacting the Ingate Unit The examples below were taken using the Ingate Trunk Group Startup Tool. When you first launch the Startup Tool TG you will need to select the Product Type: Select Ingate Firewall/SIParator as the model and press the Next button. There are three main options to keep in mind. 1) Is this an Out of the Box installation, if so select Configure the unit for the first time. 2) If the Ingate has a configuration already, then select Change or update configuration of the unit. And 3) Select Configure SIP Trunking to have the available options for SIP Trunking.

66 Select one or the other Assign IP Addresses and MAC the tool will use to config the Ingate Select Configure SIP Trunking Assign Password the tool will use to config the Ingate Select the appropriate network interface Status Information, helpful for troubleshooting

67 Startup Tool Configuration Network Topology The Network Topology tab is about defining how the Ingate product will be deployed and the Network Topology required around it. The Product Type defines the deployment and the rest define the IP Addresses and Masks and DNS Servers. Assign IP Addresses, the tool will config the Ingate Select the deployment according to the picture Status Information, helpful for troubleshooting

68 Startup Tool Configuration IP-PBX Selecting ShoreTel as the IP-PBX Type will ensure the Ingate Startup Tool propagates the necessary configuration into the Ingate SIParator or Firewall to ensure correct operation. This configuration is based off of extensive TTP Interop Testing as well as experience in various deployments. Simply assign the IP Address of the ShoreTel ShoreGear Switch. Assign the ShoreTel ShoreGear as the IP- PBX Type Assign the IP Address of the ShoreGear SIP Trunk Switch Status Information, helpful for troubleshooting

69 Startup Tool Configuration ITSP Ingate has gone out to verify and test with a large number of Carriers and Service Providers. With every ITSP verification, Ingate records the individual setup and deployment characteristics of each Service Provider. With a simple selection of the Service Provider the knowledge and configuration of each deployment is propagated to the Ingate SIParator or Firewall. Select ITSP Vendor Assign the ITSP IP Address User Account Information, DID Assignment and Registration Authentication Status Information, helpful for troubleshooting 1. In the Name: drop down option select Generic (no register). 2. In the Provider address: enter the IP address provided by Verizon Business or enable (check) the Use domain name option. 3. In the Account information: define the Domain: provided by Verizon Business.

70 Upload Configuration At this point the Startup Tool has all the information required to push a database into the Ingate unit. The Startup Tool can also create a backup file for later use. 1. Press the Upload button. If you would like the Startup Tool to create a Backup file also select Backup the configuration. Upon pressing the Upload button the Startup Tool will push a database into the Ingate unit. 2. When the Startup has finished uploading the database a window will appear and once pressing OK the Startup Tool will launch a default browser and direct you to the Ingate Web GUI.

71 3. Although the Startup Tool has pushed a database into the Ingate unit, the changes have not been applied to the unit. Press Apply Configuration to apply the changes to the Ingate unit. 4. A new page will appear after the previous step requesting to save the configuration. Press Save Configuration to complete the saving process.

72 Additional Configuration The Startup Tool addresses the majority of the configuration on the Ingate SIParator, the remaining configuration step involves creating a regular expression in the dial plan to perform B2BUA functions. Log into the Ingate Web UI, then go to SIP Trunks. Select the appropriate SIP Trunk by selecting the Goto SIP Trunk page button or the Trunk 1 tab. Once in the actual Trunk Group page, scroll down to the SIP Trunking Service parameter section:

73 Configure the Restrict to calls from: parameter, using the drop down link to Generic (no register). Scroll down to the From header domain: parameter section: Set the From header domain parameter to as entered: and in the From domain: section enter the public IP address of the Ingate SIParator/Firewall. Scroll down to the PBX Lines parameter section: There will be two entries in this area, in the second entry go to the Outgoing Calls section, then in the User Name column, to the right of the $1 enter the following:

74 ?P-Asserted-Identity=%3csip%3a$(P-Asserted- Identity.user)% %3e&Diversion[1]=sip:$(diversion[1].user)% :5060 This entry is case sensitive and be sure that there aren t any spaces. e, the P/IP address defined here ( ) should be replaced with the Ingate s WAN / Eth1 (public) IP address, the IP address is right after the %40, which must be left intact. After doing this the entry will now look as follows: You can see the entire entry by selecting it and scrolling to the right, it should now look as follows: $1?P-Asserted-Identity=%3csip%3a$(P-Asserted- Identity.user)% %3e&Diversion[1]=sip:$(diversion[1].user)% :5060 Scroll down to the Setup for the PBX parameter section: In the Match From Number/User in field: drop down list select P-Asserted-Id. URI, then in the To header field: verify that it is configured for Same as Request- URI.

75 Scroll to the bottom of the page and click on the Save button. Be sure to apply and save the configuration, as described below in section Ingate Applying and Saving Changes Ingate QoS Configuration Verizon Business requires that SIP signaling have precedence over other traffic. You will need to purchase the Quality of Service module for the Ingate device and configure it as follows. Log into the Ingate Web UI, and then select the Quality of Service tab.

76 This action will take you the QoS and SIP parameters, select the QoS Classes tab. In the QoS Classes area, verify that No. 1 has a Class Name of SIP Signaling and in the SIP area the drop down list is set to Signaling, then save the change. Select the TOS Modification tab. In the TOS/DSCP Modification area, click on the Add new rows button, then in the Class field use the drop down and select SIP Signaling, in the TOS Octet area, under the DSCP field enter 28. Save the change. Be sure to apply and save the configuration as described below in section Ingate Applying and Saving Changes Ingate Options Configuration ShoreTel 13 adds the ability to determine whether the SIP trunks are in service or not, it does so via the SIP OPTIONS message. By default Ingate responds to the OPTIONS message, which should be sufficient, but is not optimal since Ingate will be operational for the most part. Instead we recommend that you configure Ingate to pass the OPTIONS message onto Verizon Business, this way if there s a

77 connectivity issue between Ingate and Verizon Business, ShoreTel can properly take the SIP trunks out of service. In addition Verizon Business also sends OPTIONS messages to the ShoreTel system, by default Ingate responds with a 403 Forbidden message, which should be okay, but we recommend that you configure Ingate to forward the OPTIONS message to the ShoreTel IP-PBX instead, for a proper 200 OK response. Log into the Ingate Web GUI, select the SIP Traffic tab, followed by the Dial Plan page. Scroll down to the Matching From Header section and click on the Add new rows button. In the Name column, define something appropriate (we chose Verizon), in the Username and Domain columns define an asterisk (*), then in the Transport column use the drop down list to select Any, finally in the Network column, use the drop down list and select Generic (no register). Scroll down to the Matching Request-URI section and click on the Add new rows button, but change the value in the rows section to 2: This action will add two empty rows, in the first row define a name (this will be for OPTIONS messages coming from Verizon), we chose OPTIONS-Ping. Then in the Tail column, use the drop down list and select nothing, finally in the Domain column define the IP address that Verizon is sending the messages to, this will usually be the Ingate WAN interface. In the second row, define a name (this will be for OPTIONS messages coming from ShoreTel), we chose OPTIONS-ST, in the Tail column, use the drop down list and once again select nothing, finally in the Domain column define Ingate s LAN IP address. Scroll down to the Forward To section and click on the Add new rows button.

78 In the Name field define a name, we chose Verizon, then in the Or This / Replacement Domain, Port and Transport field enter the domain name provided by Verizon Business, in the Port enter the port provided by Verizon Business (in our example it is 5131) and finally in the Transport use the drop down to select UDP. Scroll down to the bottom of the page and click on the Save button. Scroll down to the Dial Plan section and click on the Add new rows button, but change the value in the rows section to 2. You will now have two empty rows. The No. field will automatically increment, modify the number to be one above the entry that contains WAN, in our example we changed the number to 2 and 3. In the first empty row, in the From Header field, use the drop down to select ShoreTel ShoreGear, then in the Request- URI field, use the drop down to select the Request-URI created earlier (in our example it is OPTIONS-ST ), then in the Action field use the drop down to select Forward. Finally in the Forward To field, use the drop down to select Forward To selection created earlier (in our example it is Verizon ). In the second empty row, in the From Header field, use the drop down to select Verizon (which was created earlier in the Matching From Header section), then in the Request-URI field, use the drop down to select the Request-URI created earlier (in our example it is OPTIONS-Ping ), then in the Action field use the drop down to select Forward. Finally in the Forward To field, use the drop down to select Generic (no register). Scroll down to the bottom of the page and click on the Save button. Be sure to apply and save the change as noted in the following section.

79 Ingate Applying and Saving Changes To apply and save any configuration changes, log into the Ingate Web GUI and select the Administration tab: Then in the Save/Load Configuration tab select the Apply configuration button. This action brings up the following page: Select the Save configuration button. This action will apply and save the configuration changes. 4.3 Ingate Troubleshooting Call Flow Examples Incoming Call

80 Incoming calls will always originate from the Verizon Business network and be addressed directly to the Ingate SIParator IP Address. The Ingate in turn will route the call to the ShoreGear switch Outgoing Call Outgoing calls will always originate from the ShoreTel Phones, then the ShoreGear switch makes a call directly to the Ingate IP address. The Ingate in turn will route the call to the Verizon Business network Startup Tool Status Bar Located on every page of the Startup Tool is the Status Bar. This is a display and recording of all of the activity of the Startup Tool, displaying Ingate unit information, software versions, Startup Tool events, errors and connection information. Please refer to the Status Bar to acquire the current status and activity of the Startup Tool.

81 Configure Unit for the First Time Right Out of the Box, sometimes connecting and assigning an IP Address and Password to the Ingate Unit can be a challenge. Typically, the Startup Tool cannot program the Ingate Unit. The Status Bar will display The program failed to assign an IP address to eth0. Possible Problems Ingate Unit is not Turned On. Ethernet cable is not connected to Eth0. Incorrect MAC Address An IP Address and/or Password have already been assigned to the Ingate Unit Ingate Unit on a different Subnet or Network Possible Resolution Turn On or Connect Power (Trust me, I ve been there) Eth0 must always be used with the Startup Tool. Check the MAC address on the Unit itself. MAC Address of Eth0. It is possible that an IP Address or Password have been already been assigned to the unit via the Startup Tool or Console The Startup Tool uses an application called Magic PING to assign the IP Address to the Unit. It is heavily reliant on ARP, if the PC with the Startup Tool is located across Routers, Gateways and VPN Tunnels, it is possible that MAC addresses cannot be found. It is the intension of the Startup Tool when configuring the unit for the first time to keep the network simple. See Section 3. Despite your best efforts 1. Use the Console Port, please refer to the Reference Guide, section Installation with a serial cable, and step through the Basic Configuration. Then you can use the Startup Tool, this time select Change or Update the Configuration 2. Factory Default the Database, then try again.

82 Change or Update Configuration If the Ingate already has an IP Address and Password assigned to it, then you should be able use a Web Browser to reach the Ingate Web GUI. If you are able to use your Web Browser to access the Ingate Unit, then the Startup should be able to contact the Ingate unit as well. The Startup Tool will respond with Failed to contact the unit, check settings and cabling when it is unable to access the Ingate unit. Possible Problems Possible Resolution Ingate Unit is not Turned On. Turn On or Connect Power Incorrect IP Address Check the IP Address using a Web Browser. Incorrect Password Check the Password. Despite your best efforts 1. Since this process uses the Web (http) to access the Ingate Unit, it should seem that any web browser should also have access to the Ingate Unit. If the Web Browser works, then the Startup Tool should work. 2. If the Browser also does not have access, it might be possible the PC s IP Address does not have connection privileges in Access Control within the Ingate. Try from a PC that have access to the Ingate Unit, or add the PC s IP Address into Access Control.

83 Network Topology There are several possible error possibilities here, mainly with the definition of the network. Things like IP Addresses, Gateways, NetMasks and so on. Possible Problems Error: Default gateway is not reachable. Error: Settings for eth0/1 is not correct. Error: Please provide a correct netmask for eth0/1 Error: Primary DNS not setup. Possible Resolution The Default Gateway is always the way to the Internet, in the Standalone or Firewall it will be the Public Default Gateway, on the others it will be a Gateway address on the local network. IP Address of Netmask is in an Invalid format. Netmask is in an Invalid format. Enter a DNS Server IP address IP-PBX The errors here are fairly simple to resolve. The IP address of the IP-PBX must be on the same LAN segment/subnet as the Eth0 IP Address/Mask. Possible Problems Error: The IP PBX IP does not seem to be on the LAN. Error: You must enter a SIP domain. Error: As you intend to use RSC you must enter a SIP domain. Alternatively you may configure a static IP address on eth1 under Network Topology Possible Resolution The IP Address of the IP-PBX must be on the same subnet as the inside interface of the Ingate Eth0. Enter a Domain, or de-select Use Domain Enter a Domain or IP Address used for Remote SIP Connectivity. e: must be a Domain when used with SIP Trunking module.

84 ITSP The errors here are fairly simple to resolve. The IP address, Domain, and DID of the ITSP must be entered. Possible Problems Error: Please enter a domain name for your provider Error: Please enter number, name and domain. Possible Resolution Enter a Domain, or de-select Use Domain Enter a DID and Domain, or de-select Use Account Apply Configuration At this point the Startup Tool has pushed a database to the Ingate Unit, you have Pressed Apply Configuration in Step 3) of Section 4.7 Upload Configuration, but the Save Configuration is never presented. Instead after a period of time the following webpage is presented. This page is an indication that there was a change in the database significant enough that the PC could no longer web to the Ingate unit. Possible Problems Eth0 Interface IP Address has changed Access Control does not allow administration from the IP address of the PC. Possible Resolution Increase the duration of the test mode, press Apply Configuration and start a new browser to the new IP address, then press Save Configuration Verify the IP address of the PC with the Startup Tool. Go to Basic Configuration, then Access Control. Under Configuration Computers, ensure the IP Address or Network address of the PC is allowed to HTTP to the Ingate unit.

85 4.3.3 Ingate Configuration Configure your Ingate Firewall or Ingate SIParator to get basic network connectivity on all applicable interfaces. Please refer to the Reference Guide and other documentation as needed. Remember to configure the following: Assign IP addresses on the inside and outside interface. For DMZ SIParators, use one interface only. (Network -> All Interfaces) Assign a default gateway. (Network -> Default Gateway) Assign a DNS server address. (Basic Configuration -> Basic Configuration) Define the IP subnet allowed to configure the Ingate and the interfaces to use for configuration. (Basic Configuration -> Access Control) First make these basic settings and apply the configuration to have the unit working in your network environment. Then proceed with the following settings to get SIP Trunking to work with your service provider Network and Computers This is an example of the Network Networks and Computers page with an Ingate SIParator in a Stand-alone configuration. The Networks and Computer page is a IP Table List or Route List, providing the Ingate knowledge of its surrounding networks and what interface they are connected too. Also, the table provides identification of specific IP Addresses for later use in providing filter and identification of source IP addresses in the Dial Plan and other locations. Add a network for the Service Provider (ITSP IP). If you don t know the IP addresses used for the ITSP, you can put in as lower limit and as upper limit. In this way, requests from any IP address will be accepted. Add a network for the LAN (inside IP range). Add an IP Address of the ShoreTel ShoreGear

86 Basic Configuration SIParator Type (SIParator Only) Use the appropriate SIParator configuration for your deployment SIP Traffic Filtering Under Proxy Rules, change the Default Policy for SIP Requests to Process All. Content Type: Add */* and Allow - ON

87 Interoperability There are some general Interop settings required for use with ShoreTel. Configuration Steps: URI Encoding Use shorter, encrypted URI Signaling Order of Re-INVITEs Send response before re-invite are forwarded Allow Large UDP Packets - Allow Large UDP Packets

88 Dial Plan This is an example of the SIP Traffic - Dial Plan on the Ingate SIParator. There are three main parameters that need to be defined to create the Dial Plan. Matching From Header, Matching Request URI and Forward To are parameters that when combined together form the Dial Plan. The key difference in the ShoreTel integration is the use of the B2BUA as the SIP Normalization Tool, thus the Forward To section Regular Expression has sip:$1@ ;b2bua for the ShoreTel ShoreGear destination as well as sip:$1@ ;b2bua for the Verizon Business network destination.

89 4.3.4 Ingate Troubleshooting Tools Display Logs Here is the internal logging of the Ingate. The Display Logs show all SIP Signaling and also TLS (SSH) certificate exchange and setup. Press Display Log to see internal logs Always create a Support Report for Ingate Support Show newest log on top Filter on SIP specific fields Filter on SIP traffic only

90 Packet Capture The Packet Capture capability of the Ingate allows for the capture and export of all traffic on any one or ALL interfaces simultaneously. Then export to your PC where it can be viewed in Wireshark or Ethereal. Select All Interfaces to cook multiple captures from multiple interfaces into one PCAP Filter on Port, Transport and other criteria Download PCAP File Start Capture, reproduce the problem, then Stop Capture

91 Check Network Standard PING and Trace Route feature for simple network checks. PING and Trace Route 4.4 Ingate Sales & Technical Support Sales North America For general sales questions or resellers who want to start selling this solution, please contact; Steven Johnson or EMEA For general sales questions or resellers who want to start selling this solution please contact; Ingate Systems HQ or Technical Support North America Customers: The Ingate Authorized Reseller should always be your first contact for support. The ShoreTel TAC is a part of the Ingate Authorized Resellers. If you don't work with an Ingate Authorized Reseller, you may purchase an Annual Support Agreement from Ingate Systems. All support questions and issues should be directed to: support@ingate.com Phone: Operational Hours: 8:00am to 6:00pm EST