ShoreTel, Ingate & Gamma Telecom for SIP Trunking

Save this PDF as:
 WORD  PNG  TXT  JPG

Size: px
Start display at page:

Download "ShoreTel, Ingate & Gamma Telecom for SIP Trunking"

Transcription

1 TPP APP NOTE TPP Date: Sept. 19, 2007 Product: ShoreTel Ingate Gamma Telecom System version: ShoreTel 7.5 ShoreTel, Ingate & Gamma Telecom for SIP Trunking 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 box which is connected to both the LAN and WAN, and acts as a gateway to Gamma Telecom for SIP Trunking. Table of Contents ShoreTel, Ingate & Gamma Telecom for SIP Trunking.1 Overview...2 Gamma Telecom Overview and Contact...2 Gamma Telecom...2 Ingate Systems...2 North America...3 EMEA...3 Customers outside North America:...3 What is DIRECT CONNECT?...3 Ingate Product Information...5 Figure 1 - Ingate Firewalls and SIParators...5 Architecture Overview...5 Figure 2 - Architectural Overview...6 Figure 3 Ingate, 3 Possible Options...7 Requirements, Certification and Limitations...7 Version Support...7 ITSP Certification Testing Results Summary...8 Table 1: Initialization and Basic Calls...8 Table 2: Media and DTMF Support...8 Table 3: Performance & Quality of Service...9 Table 4: Enhanced Services and Features...10 Table 5: Security...11 Configuration Overview ShoreTel Unsupported Features ShoreTel Configuration ShoreTel System Settings General:...11 Figure 1 Administration Call Control Options...12 Figure 2 Call Control Options...13 Figure 3 Administration Site...14 Figure 4 Admission Control Bandwidth...14 Sites Edit screen Admission Control Bandwidth...14 Switch Settings - Allocating Ports for SIP Trunks:...14 Figure 5 Administration Switches...15 Figure 6 ShoreGear Switch Settings...15 ShoreTel System Settings Trunk Groups:...16 Figure 7 Administration Trunk Groups...16 Figure 8 Trunk Groups Settings...16 Figure 9 SIP Trunk Group Settings...17 Figure 10 Inbound:...18 Figure 11 Trunk Services:...19 ShoreTel System Settings Individual Trunks:...19 Figure 12 Individual Trunks...20 Figure 13 Trunks by Group...20 Figure 14 Edit Trunks Screen for Individual Trunks...21 Ingate Configuration Alternative A: Ingate Startup Tool...21 Step I Basic Ingate Setup...21 Step II SIP Ingate Trunk Setup (continued)...23 Step III SIP Ingate Trunk Setup...24 Step IV SIP Trunk Setup...25 Alternative B: Configuration through the GUI...25 Ingate Troubleshooting Troubleshooting Outbound Calls...28 Troubleshooting Inbound calls...30 Ingate Technical Support North America Customers:...33 Customers outside North America:...34 Gamma Telecom Functional Overview Access to the Service Public Internet...35 Public Internet Bypass...35 Private Interconnect...35 Gamma Provided Access...36 ADSLMax Solution...36 End User Features Gamma Telecom Support Responsibilities Desk Hours and Contacts...38 Escalation Process...38 Document and Software Copyrights Trademarks Disclaimer Company Information... 40

2 Overview This document provides details for connecting the ShoreTel system through the Ingate SIParator to Gamma Telecom for SIP Trunking to enable audio communications. The document focuses on the configuration procedures needed to set up these systems to interoperate. Gamma Telecom Overview and Contact Gamma Telecom is one of the UK's largest providers of voice services and applications, switching in excess of 7 billion minutes per annum and supplying service via its channel partners to over 50,000 businesses and 400,000 residential customers. Gamma sells highly reliable, traditional and leading edge services through the brand names of over 200 channel partners, which include some of the UK's largest and most successful Systems Integrators, Voice Resale partners, Internet Service Providers (ISPs), Internet Telephony Service Providers (ITSPs) and Specialist Carriers. As an industry leader in IP Telephony with a full UK national network, Gamma's voice services and applications are increasingly provided over a pure IP-based infrastructure, with legacy time-division multiplexing (TDM) interconnects to British Telecom and other carriers. With full ISO9000 accreditation, Gamma is totally committed to a policy of doing it once and getting it right, and is an industry example of the benefits of such an approach. Resellers that are not already Gamma Partners can register their interest in Gamma s IP Trunk services by calling Gamma Customer Services on Alternatively, prospective Resellers can call Gamma Telecom s Head Office on Existing Partners can order Gamma IP Direct Connect Services via the partner portal at Gamma Telecom Home Office 1 Park St, Newbury, Berkshire, RG14 1EA Ingate Systems offers the only fully SIP capable security products offering features important to enterprise adoption of SIP Trunking. The Ingate Firewall offers a single device to protect the network and manage SIP traffic. The Ingate SIParator allows the enterprise to adopt SIP without replacing an existing firewall. Both products include a SIP ALG, proxy and registrar that enable SIP signaling to traverse the firewall, support for dynamic media port management to keep the network safe, encryption for

3 privacy, added routing capabilities to make the installation of SIP Trunks simple and inexpensive, and remote SIP connectivity that enables the enterprise to offer SIP services to their remote workers. North America For general sales questions, please contact your reseller or contact Ingate directly at: Steven Johnson Resellers who want to start selling this solution should contact: Steven Johnson EMEA For general sales questions, please contact your reseller or contact Ingate directly at: Ingate Systems HQ Resellers who want to start selling this solution should contact: Ingate Systems HQ Customers outside North America: Contact your reseller for support. If you don't work with an Ingate Authorized Reseller, you may purchase an Annual Support Agreement from Ingate Systems. For pricing of an Annual Support Agreement, please or phone All support questions and issues should be directed to What is DIRECT CONNECT? IP DIRECT CONNECT is a Gamma Telecom voice termination service that allows Channel Partners to provide inbound and outbound voice telephony calls to their end customers using IP telephony protocols from CPE (customer premises equipment) provided by the Channel partner

4 IP DIRECT CONNECT helps transform an end user s site from analog technology to an IP based one, where the customer s ShoreTel endpoint (e.g. IPBX) will be connected to Gamma s network over an IP link. The service offers the following benefits to Resellers / Channel Partners and end users: Lower call costs than traditional TDM-based solutions Zero rated site-to-site calling the ability to offer FREE site-tosite calls Lower connection and rental costs than ISDN Faster provisioning timescales than ISDN Scaleable on a per channel basis Unlocks the IP PBX enabling the creation of a Voice virtual private network (VPN) across multiple sites Offers full emergency services support Complements a converged network with lower costs and simplified architecture IP Direct Connect Service (SIP Trunk) IP DIRECT CONNECT is one of the first true IP transit (SIP Trunk) services on the market, connecting customer sites directly into the Gamma Telecom network using an IP backbone for full PSTN connectivity. IP DIRECT CONNECT is designed for a whole range of companies, including IP PBX distributors, Systems Integrators, Value Added Resellers and Data Networking companies. Key Features IP connectivity at a lower cost than ISDN A scaleable service Lower call costs Daily Call Detail Records (CDRs) Emergency services support - 4 -

5 Ingate Product Information Firewall 1190 & SIParator 19 Possible to SW upgrade Firewall 1450 & SIParator Calls* 120 Mbit/s Firewall & SIParator Calls* 310 Mbit/s Firewall 1600 & SIParator 60 List Price 5100 plus 500 Calls* 385 Mbit/s List Price 6600 plus Firewall 1900 & SIParator Calls* Mbit/s List Price plus 40 Calls* 50 Mbit/s List Price 860 plus options List Price 3000 plus Figure 1 - Ingate Firewalls and SIParators Architecture Overview Providing a good solution to allow customers the ability to connect to SIP Trunks provided by different ITSPs is important. ShoreTel and Ingate have teamed up to build a solid solution, ShoreTel being the IP PBX which sits on the LAN and connects to the Ingate SIParator / Firewall. The Ingate then is connected to not only the LAN but also the WAN, providing not only typical firewall capabilities but also an 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) - Transfer conversion of SIP REFER to SIP reinvite messaging (critical) - Quick configuration templates for each of the certified ITSPs Image below shows a high level drawing of a basic ShoreTel / Ingate / ITSP design. This drawing only represents SIP and Real-time Transfer Protocol - 5 -

6 (RTP) traffic. The next section of this application note covers actual deployment design options. Figure 2 - Architectural Overview Ingate has two products for this solution, the Ingate Firewall and Ingate SIParator. From an SIP functionality point of view they are basically the same. The Ingate Firewall also provides normal data firewalling functionality and is recommended if the enterprise wants to replace their existing firewall. The Ingate SIParator is the solution for those who want to keep an existing firewall when adopting SIP. In this case, the Ingate SIParator will co-exist in parallel with the normal data firewall. The routing of SIP traffic to the Ingate SIParator can be accomplished in three primary ways. The first is the most commonly deployed though each configuration offers its own advantages for the enterprise: Configuration 1: Single leg/dmz only, firewall logs all activity Configuration 2: DMZ/LAN, reduced load on firewall - 6 -

7 Configuration 3: Two legged/standalone, SIP traffic separate from data traffic Figure 3 Ingate, 3 Possible Options Requirements, Certification and Limitations Any Ingate SIParator or Ingate Firewall model will work in this configuration. In a Trunking scenario, it is required to have the Ingate SIP Trunking module installed. A few traversal licenses are included with the Ingate unit at delivery. Typically one traversal license will be needed for each expected concurrent phone call on the SIP Trunk. Additional licenses can be bought via your Ingate reseller. Version Support Products are certified via the Technology Partner Certification Process for the ShoreTel system. The table below contains the matrix of Ingate Firewall and Ingate SIParator version firmware releases certified on the identified ShoreTel software releases. Ingate Firewall and Ingate SIParator version with the patch ig- patch shoretel-2 applied ShoreTel

8 ITSP Certification Testing Results Summary Table 1: Initialization and Basic Calls ID Name Description Notes 1.0 Configuration TPP Lab will use the configuration Pass Application Note application note provided by the vendor to configure the vendor s product to work with the ShoreTel system. 1.1 Setup and Verify successful setup and Pass initialization initialization of the Server under Test. 1.2 Outbound Call Verify calls outbound placed through the Pass (Domestic) 1.3 Inbound Call (Domestic) 1.4 Device restart Power Loss 1.5 Device restart Network Loss 1.6 All Trunks Busy Inbound Callers 1.7 All Trunks Busy Outbound Callers 1.8 Incomplete Inbound Calls SUT reach the external destination. Verify calls received by the SUT are routed to the default trunk group destination. Verify that the SUT recovers after power loss to the SUT. Verify the SUT recovers after loss of network link to the SUT. Verify an inbound caller hears busy tone when all channels/trunks are in use. Verify an outbound caller hears busy tone when all channels/trunks are in use. Verify proper call progress tones are provided and proper call teardown for incomplete inbound calls. Pass Pass Pass Pass Pass Pass Table 2: Media and DTMF Support ID Name Description Notes 2.1 Media Support ShoreTel Phone to SUT Verify call connection and audio path from a ShoreTel phone to an external destination through the service provider using all supported codes with both Pass 2.2 Media Support SIP Reference to SUT 2.3 Codec Negotiation 2.4 DTMF Transmission Out-of-Band / In- Band sides set to a common codec Verify call connection and audio path from an SIP Reference phone to an external destination through the service provider using all supported codes with both sides set to a common codec. Verify codec negotiation between the SUT and the calling device with each side configured for a different codec. Verify transmission of in-band and outof-band digits per RFC 2833 for various devices connected to the SUT. Pass Pass Pass - 8 -

9 Table 3: Performance & Quality of Service ID Name Description Notes 3.1 Speech quality Minimal Impairment Verify acceptable voice quality between two parties with minimal network impairment condition. Not Tested 3.2 Speech quality Moderate Impairment 3.3 Speech quality High Impairment 3.4 Voice Quality Service Levels Verify acceptable voice quality between two parties with low-tomoderate artificial network impairment condition. Verify acceptable voice quality between two parties with moderate-tohigh artificial network impairment condition. Verify the SUT can provide a voice quality service-level agreement (SLA) across the WAN from the customer premises to the SUT SIP gateway. 3.5 Capacity Test Verify the service provider interface can sustain services through period of heavy outbound and inbound load. 3.6 Post Dial Delay Verify that post dial delay is within acceptable limits. 3.7 Billing Verify that all test calls made are Accuracy accurately reflected in the SUT s CDR and billing reports. Not Tested Not Tested Not Tested Gamma Telecom s lab setup was configured for a maximum of four simultaneous calls. Pass Pass Used ShoreTel CDRs - 9 -

10 Table 4: Enhanced Services and Features ID Name Description Notes 4.1 Caller ID Name Verify that Caller ID name and number Fail, and Number - is received from SIP endpoint device. intermittent Inbound failures where CID is IP address of ITSP 4.2 Caller ID Name and Number - Outbound 4.3 Hold from SUT to SIP Reference 4.4 Call Forward - SUT 4.5 Call Transfer Blind 4.6 Call Transfer Consultative 4.7 Conference Ad Hoc 4.8 Inbound DID/DNIS Verify that Caller ID name and number is sent from SIP endpoint device. Verify successful hold and resume of connected call. Verify outbound calls that are being forwarded by the SUT are redirected and connected to the appropriate destination. Verify a call connected from the SUT to the ShoreTel phone can be transferred to an alternate destination. Verify a call connected from the SUT to the ShoreTel phone can be transferred to an alternate destination. Verify successful ad hoc conference of three parties. Verify the SUT provides inbound dialed number information and is correctly routed to the configured destination. 4.9 Outbound 911 Verify that outbound calls to 911 are routed to the correct Public Safety Answering Point (PSAP) for the calling location and that caller ID information is delivered Operator Assisted Verify that 0+ calls are routed to an operator for calling assistance. (Gamma) Pass / Fail to Cingular mobile number always shows up as Unknown Pass Pass Pass Pass Pass Pass Not Tested Not Tested

11 Table 5: Security ID Name Description Notes 5.1 Digest Authentication Configuration Overview Verify the SUT supports the use of digest authentication for service access for inbound and outbound calls. Gamma Telecom authenticate s based on the IP address provided. The configuration information below shows examples for configuring the ShoreTel, Ingate and Gamma Telecom. Configuration requirements can vary from setup to setup though the information provided in these steps, along with the Planning and Installation Guide and documentation provided by Ingate and the Gamma Telecom, should prove to be sufficient. However, every design can vary and some may require more planning then others. ShoreTel Unsupported Features At the time of this writing, the following are currently not supported, though support will be added in an upcoming future release - Fax redirect not supported today via SIP Trunks (though direct DID to fax endpoint is supported) - Office Anywhere 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. Note: ShoreTel basically just points its Individual SIP Trunks to the Ingate SIParator. ShoreTel System Settings General: 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

12 into ShoreWare Director and select Administration then Call Control followed by Options (Figure 1). Figure 1 Administration Call Control Options The Call Control Options screen will then appear (Figure 2)

13 Figure 2 Call Control Options Within the Call Control Options screen, confirm that the appropriate settings are made for the Enable SIP Session Timer, Intra-Site Calls, Inter-Site Calls and Always Use Port 5004 for RTP fields. The first step is to make sure that the box is checked for the Enable SIP Session Timer. Next the Session Interval Timer needs to be set. The recommended setting for Session Interval is 1800 seconds. The last item to select is the appropriate refresher (from the pull down menu) for the SIP Session Timer. The Refresher field will be set either to Caller (UAC) [User Agent Client] or to Callee (UAS) [User Agent Server]. If the Refresher field is set to Caller (UAC), the Caller s device will be in control of the session timer refresh. If Refresher is set to Callee (UAS), the device of the person called will control the session timer refresh. The next settings to verify are the Intra-Site Calls and the Inter-Site Calls settings under the Voice Encoding and Quality of Service prompt. For the Intra-Site Calls, verify that the desired audio bandwidth is selected for the CODEC for calls within the system. The settings should then be confirmed for the desired audio bandwidth CODEC for Inter-Site calls (calls between sites). Note: SIP uses both G.711 and G.729 CODECs. The CODEC setting will be negotiated to the highest CODEC supported (fax requires G.711 at minimum). Un-checking the box for Always Use Port 5004 for RTP is required for implementing SIP on the ShoreTel system. For SIP configurations, Dynamic User Diagram Protocol (UDP) must be used for RTP Traffic. If the box is unchecked, Media Gateway Control Protocol (MGCP) will no longer use UDP port 5004; MGCP and SIP traffic will use dynamic UDP ports. Once this parameter is unchecked, ensure 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 Shore Ware Director by selecting Administration then Sites (Figure 3)

14 Figure 3 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. The only change required to the Edit Site screen is to the Admission Control Bandwidth field (Figure 4). Figure 4 Admission Control Bandwidth Note: Bandwidth of 1024 is just an example. Please see the 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 devices will be counted against the site bandwidth. Bandwidth needs to be set appropriately based on site setup and configuration with the Gamma Telecom SIP Trunking. See the ShoreTel Planning and Installation Guide for more information. Switch Settings - Allocating Ports for SIP Trunks: The final general settings to input are the ShoreGear switch settings. These changes are modified by selecting Administration, then Switches in ShoreWare Director (Figure 5)

15 Figure 5 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 will be displayed. Within the Edit ShoreGear Switch screen, select the desired number of SIP Trunks from the ports available (Figure 6). Figure 6 ShoreGear Switch Settings Each port designated as an SIP Trunk enables the support for 5 individual trunks

16 ShoreTel System Settings Trunk Groups: ShoreTel Trunk Groups support both Dynamic and Static SIP endpoint Individual Trunks. Note: A ShoreGear switch can only support one Trunk Group with Dynamic IP addressing. In trunk planning the following need to be considered. 1. Are the SIP devices using DHCP or Static IP? 2. Are the SIP devices endpoints (like ATAs, Conference Phone or WiFi handset) or non-endpoint devices like an ITSP? 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 (Figure 7). Figure 7 Administration Trunk Groups This selection brings up the Trunk Groups screen (Figure 8). Figure 8 Trunk Groups Settings From the pull down menus on the Trunk Groups screen, select the site desired and select the SIP trunk type to configure, then click on the Go link from Add new trunk group at site:. The Edit SIP Trunk Group screen will appear (Figure 9)

17 Give the Trunk Group a meaningful name. Example: HQ-SIP-Ingate-ITSP Figure 9 SIP Trunk Group Settings For the Ingate SIP Trunking, the trunks need to be configured as inter-site trunks (trunks between sites). The trunks will also be configured as static. The next step within the Edit SIP Trunks Group screen is to input the name for the trunk group. In the example in Figure 9, the name SIP has been created. The next step is to verify the setting of the Teleworker check box. The Teleworker check box needs to be checked since the trunk groups have been configured as inter-site. Since this box is checked, it will count against the site bandwidth. The Enable Digest Authentication field is not required when connecting to an Ingate box. 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 (Figure 10)

18 Figure 10 Inbound: Within the Inbound: settings, ensure that the Number of Digits from CO is set to 10 and the DNIS or DID box is checked along with the Extension parameter (see Planning and Installation Guide for further information on configuration). Tandem Trunking is not required unless you plan on routing incoming SIP trunk calls out other ShoreTel trunks. Note: This section is configured in the same way as any normal Trunk Group

19 Figure 11 Trunk Services: On the Trunk Services: screen, make sure that the appropriate services are checked or unchecked based on what the ITSP supports and what features are needed from this Trunk Group. The last checkbox 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 of the SIP Trunk group. After these settings are made to the Edit SIP Trunk Group screen, press the Save button to input the changes. This completes the settings needed to set up the trunk groups on the ShoreTel system. ShoreTel 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 (Figure 12)

20 Figure 12 Individual Trunks The Trunks by Group screen, used to change the individual trunks settings, then appears (Figure 13). Figure 13 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 Headquarters and the trunk group is SIP. Click on the Go button to bring up the Edit Trunk screen (Figure 14). IP Address of Ingate LAN interface

21 Figure 14 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 recommendation 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 ITSP Trunk select Use IP Address button and input an IP address of the Ingate SIParator product. The last step is to select the number of individual trunks desired (each one supports one audio path example if 5 is input, then 5 audio paths can be up at one time). Once these changes are complete, select the Save button. Note: 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 SIParator in the same way as if primary rate interface (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. Ingate Configuration The Ingate product can be configured using two alternative methods, either the Ingate Start-up Tool, a wizard for a complete first time configuration, or the traditional configuration via the GUI. The latter is more suitable if you already have your Ingate configured and operational in your network. Select one of these methods for configuration of the Ingate unit (the option most suitable for you). Alternative A: Ingate Startup Tool When you have received your Ingate device, unpack it and connect it to the network according to the picture below (in the screen shot). Install the Start-up Tool on a Windows PC and start the tool. Make sure that your Ingate device is turned on. Step I Basic Ingate Setup Accomplish basic network setup by following steps A-D in the picture below. Note that if you already provided your Ingate unit an IP address, you don t need to do the MAC address part, but all your old settings will be replaced by the tool

22 A. Connect Ingate according to picture B. Type MAC address and IP address of the Ingate unit. The MAC address is found on the label of the Ingate. C. Enter a password. No password is set by default. D. Press Contact Status information

23 Step II SIP Ingate Trunk Setup (continued) In the second step, the SIP Trunking tool is launched. Once contact has been established, step two is to configure the Network Topology by clicking on Launch

24 Step III SIP Ingate Trunk Setup (continued) In the third step, the SIP Trunk itself is configured. Follow steps A-C in the picture below. If you don t have the Trunking module installed, the tool will prompt you for license information and automatically install the Trunking module before continuing to the trunk configuration. A. Configure netmask for internal network B. Configure external interface using DHCP or static IP C. Click Launch again

25 Step IV SIP Trunk Setup (continued) In the fourth step, the SIP Trunk itself is configured. Follow steps A-D in the picture below. If you don t have the Trunking module installed the tool will prompt you for license information and automatically install the Trunking module before continuing to the trunk configuration. A. Select trunking provider from the drop-down menu and provide necessary account information. B. Select ShoreTel from the drop-down menu and IP address on your network. C. Check Use number manipulation and put a + in the Add Prefix Box. D. When all settings are entered, the tool will generate a configuration based on your input, and you will automatically be redirected to the Ingate. You only need to apply the configuration and then start using it! Once the settings have been entered in the tool and the configuration has been applied in the Ingate GUI, the SIP trunk is operational. Alternative B: Configuration through the GUI 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 if 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 (domain name 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

26 Control) First make these basic settings and apply the configuration to have the unit working in the network environment. Then proceed with the following settings to get SIP Trunking to work with the ITSP. NETWORK NETWORK AND COMPUTERS - Add a network for the Service Provider (ITSP IP). If you don t know the IP addresses used, 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). BASIC CONFIGURATION SIParator TYPE (SIPARATOR ONLY) Use the appropriate SIParator configuration for your deployment. SIP SERVICE BASIC SIP Module: On. SIP Traffic Filtering Under Proxy Rules, change the Default Policy for SIP Requests to Process All. SIP TRAFFIC USER DATABASE Configure an account with details as provided from the ITSP. Notice that the account type must be XF/Register as shown in the picture. The Ingate will register to the ITSP using this account information so that the ITSP knows where to send incoming calls and it also works as authentication of outgoing calls. SIP TRAFFIC DIAL PLAN Configure the Dial Plan according to the picture below

27 SIP TRAFFIC ROUTING Local REFER handling: Always check handle REFER locally. The configuration of the Ingate is now done and the changes must be applied on the Administration page to take effect

28 Ingate Troubleshooting Troubleshooting Outbound Calls Symptom: When trying to make a call from an internal ShoreTel extension to PSTN, there is no ringing signal on the PSTN phone. Note: If you get a ringing signal on the PSTN phone, these troubleshooting steps will not help you to find the problem. Please contact your sales representative for support. Outbound traffic troubleshooting overview Get a log for the failing call First try to make a call to a PSTN number from a ShoreTel phone and notice the behavior on the ShoreTel phone as well as on the PSTN phone

29 Next step is to search the log on the Ingate. Log into the Ingate box and navigate to the Display Log page. Make necessary settings on this page according to the picture below. Especially make sure that you have the highlighted checkboxes in the correct state. Then press Display log further down on the same page. You will now see a log of all SIP packets received and sent by the Ingate, with the newest log entry on the top. Ensure the signaling is received from the ShoreTel Localize the call initiation from the ShoreTel by searching for invite sip in your browser. You should look for the first packet coming from the ShoreTel system that starts with a recv from <IP address of the ShoreGear switch> as you can see in the example below (only the first lines of the log messages are shown here)

30 >>> Info: sipfw: recv from :5060 via UDP connection 12746: INVITE SIP/2.0 If you cannot find a packet like the one above, the problem is in the communication from Shoregear to the Ingate. Follow these steps: 1. Make sure the Ingate SIP module is turned on, SIP Services SIP Module On. Retest if you change any setting. 2. Make sure the ShoreTel configuration is correct. Check the IP address pointing at Ingate one extra time. Retest if you change any setting. 3. Make sure there is IP connectivity between the ShoreTel and Ingate. Contact your network administrator for assistance if needed. If none of the steps above solves the problem, contact your sales representative for support. Ensure the signaling to the ITSP works If you find the incoming packet, you should find a similar packet leaving the Ingate just above (just after in time) the incoming packet. The first rows of the outgoing packet will look something like this: >>> Info: sipfw: send sf (0x ) to :5060 via UDP connection 12748: INVITE SIP/2.0 If you don t see the outgoing packet, something is probably wrong with the Ingate configuration or you lack Internet connectivity: 1. Make sure the Ingate is configured correct. 2. Make sure the IP connectivity between the Ingate and the ITSP is working. Contact your network administrator for assistance if needed. If you see a packet sent from the Ingate verify that it is sent to the IP address provided by the ITSP. If not, correct your configuration and retest. If none of the steps above solves the problem contact your sales representative for support. Troubleshooting Inbound calls Symptom: When trying to make an inbound call to a ShoreTel phone via the SIP Trunk there is no ringing signal on the ShoreTel phone

31 Note: If you get a ringing signal on the ShoreTel phone this troubleshooting will not help you to find the problem. Please contact your sales representative for support. Inbound troubleshooting overview Get a log for the failing call First try to make a call to a ShoreTel phone from a PSTN phone and notice the behavior on the ShoreTel phone as well as on the PSTN phone. Next step is to search the log on the Ingate. Log in to the Ingate box and navigate to the Display Log page. Make necessary settings on the logging page according to the picture below. Especially make sure that you have the highlighted checkboxes in the correct state

32 Then press Display log further down on the same page. You will now see a log of all SIP packets received and sent by the Ingate, with the newest log entry on the top. Ensure the signaling is received from the ITSP Localize the call initiation from the Trunking provider by searching for invite sip in your browser (use Ctrl-F). You should look for the first packet coming from the ITSP system that starts with a recv from <IP address of the ITSP> as you can see in the example below (only the first lines of the log are shown below). >>> Info: sipfw: recv from :5060 via UDP connection 12748: INVITE SIP/

33 If you cannot find a packet like the one above, the problem is in the communication from the ITSP to the Ingate. Follow these steps: 1. Make sure you have IP connectivity between the Ingate and your ITSP. Contact your network administrator for assistance if needed. 2. Make sure the Ingate SIP module is turned on, SIP Services SIP Module On. Retest if you change any setting. If you still don t see any packets in the log, contact your ITSP for further troubleshooting. Ensure correct signaling to the ShoreTel PBX If you find the incoming packet, you should find a similar packet leaving the Ingate just above (just after in time) the incoming packet. The first lines of the outgoing packet will look something like this: >>> Info: sipfw: send sf (0x ) to :5060 via UDP connection 12746: INVITE SIP/2.0 If you don t see the outgoing packet, something is probably wrong with the Ingate configuration or you may lack connection to your LAN where the ShoreTel is located: 1. Make sure that you have IP connectivity between ShoreTel and the Ingate. Contact your network administrator for assistance if needed. 2. Make sure your Ingate is configured correctly. If you see the outgoing packet, make sure the IP address it is sent to is the one used by the Shore Gear switch. If the call still fails after executing the steps described above, please contact your sales representative for support. Ingate Technical Support North America Customers: Contact your reseller for support. If you don't work with an Ingate Authorized Reseller, you may purchase an Annual Support Agreement from Ingate Systems. For pricing of an Annual Support Agreement, please or phone Steve Johnson at All support questions and issues should be directed to

34 * 8 # * 8 # * 8 # Customers outside North America: Contact your reseller for support If you don't work with an Ingate Authorized Reseller, you may purchase an Annual Support Agreement from Ingate Systems. For pricing of an Annual Support Agreement, please or phone All support questions and issues should be directed to Gamma Telecom Functional Overview IP DIRECT CONNECT Reseller 1 IP Direct Connect or other Gamma Telecom IP Product Customer 2 Site 1 Reseller 2 IP Direct Connect or other Gamma Telecom IP Product Public/Private Access LLU / ADSL / Private Interconnect Analogue phone (optional) ShoreTel IP Phones x n Fax PC Gamma Softswitch* (Endpoint x) Reseller 1 IP Direct Connect Customer 1 Site 1 Gamma Marconi or Nortel (or OLO) Marconi/Nortel CDRs Softswitch CDRs Customer equipment - only IPBX (Endpoint) is of relevance to Gamma Customer 1 belonging to Reseller 1 has an ShoreTel IPBX (the endpoint) connected to Gamma s SIP Softswitch via a Private Interconnect, Gamma derived access or other DSL transport The Customer may connect a number of Generic devices to an endpoint, and will be charged based on the number of concurrent channels requested. The Customer can call any other Customer belonging to the same Reseller; these calls will be zero rated to the Reseller with all other calls charged at the contracted rates. x

35 Calls to another Gamma IP endpoint on a different Reseller account will be charged as an off-net call. The service is a per site service and as such does not allow the channels to be dynamically allocated between two originating IP addresses (sites/endpoints). For connections via the public Internet, communication with the Gamma network must be made using an SIP proxy server (for example an SIP trunk card in a PBX) and a static public IP address. For connections using the Gamma Provided Access Solution ADSLMax, the customer will be provided with a private IP address range for use with the service. Access to the Service The IP DIRECT CONNECT service can be accessed by a number of methods, with the means of access selected by the Reseller to meet their and the end user s needs. Gamma provides the connectivity and authentication to the service via SIP Registration (IPDC V3) or IP Authentication (IPDC V1 & V2). To access the IPDC SIP Registration address or fixed IP address, Gamma channel partners can use one of the following transport options: Public Internet Public Internet bypass (i.e. connecting directly to an ISP or access provider that Gamma already peers with) Private Interconnect Gamma Provided Access Solutions Public Internet It is the responsibility of the Channel Partner who chooses to connect to Gamma via the public Internet to provide the IP transport to the customer s site. The Gamma service boundary for this connectivity option is the IP address of the Gamma network at the Gamma Session Border Controller. Public Internet Bypass As for standard Public Internet connections, it is the Channel Partner s responsibility to source the IP transport. The benefit of using Public Internet Bypass is that it provides a single hop transit between the originating network and the Gamma IPAN. The Gamma peering arrangements are regularly being updated. For details of Gamma s current peering agreements, please contact your account manager for advice. Private Interconnect Gamma is currently able to provide both electrical and optical Ethernet interfaces at a meet me point situated in the following locations: Rack Telecity2, 8/9 Harbour Exchange Square Rack TFM 51, Telehouse East

36 SYST RPS STRT UTIL DUPLXSPEED SYST MODE RPS STRT UTIL DUPLXSPEED SYST MODE RPS STRT UTIL DUPLXSPEED SYST MODE RPS STRT UTIL DUPLXSPEED MODE Catalyst 2950SERIES Catalyst 2950SERIES Catalyst 2950SERIES Catalyst 2950SERIES Rack Floor 8 Redbus, 6/7 Harbour Exchange Square These meet me points are supported via a resilient Ethernet over fiber ring as shown in the picture below. 10/100/1000 Mb/s to Customer Gigabit Ethernet to Gamma Core Network Ethernet Meet Me Switches 10/100/1000 Mb/s to Customer Gigabit Ethernet to Gamma Core Network Ethernet Meet Me Switches Connection type is 10/100/1000 Cat5/6 only. Bandwidth/Port requirements (per customer) must be known prior to agreeing to service. Channel Partners requiring resilient connections must notify Gamma in advance of their requirements. Channel Partners are responsible for the costs and installation/delivery of the cables to Gamma racks using the appropriate resources at these sites. All cables must have at least 2m spare and be terminated correctly. All cables must be clearly labeled / tagged with the Customer Name, Source Rack/Port and Destination Rack/Port. Gamma Provided Access Gamma can provide access to the end user site as an optional service to Channel Partners. This service can be provided in one of two ways depending on the location of the end user s premises. ADSLMax Solution For end user sites that are not within reach of the local Exchanges that can provide LLU service, Gamma will provide an ADSLMax solution providing the potential for 832kbps Upstream and 8Mbps Downstream Bandwidth. This service is provided at a contention ratio of 10:1. Gamma will provide and install (optional charge) a router for use with the service. The router will be configured using VPN tunnels and private IP addressing, thus

37 providing a secure and managed solution from the end user site to the Gamma SBC for service termination. Following installation of the router, test calls will be made directly through the router. These tests are designed to ensure that the router configuration is working correctly; it is not an end-to-end test involving the customer s CPE. This will happen later in the provisioning process. Note: For both Gamma provided access solutions, the Channel Partner is responsible for the Analog line provision at the end user s site that will be required to support the LLU or ADSLMax connection. End User Features End users are served by the IP telephony device connected to the IP DIRECT CONNECT (IPDC) service. They will enjoy the features provided by the IP-PBX or similar device connected. IPDC will provide call control and signaling messages to the IP Telephony device attached Gamma Telecom Support Responsibilities The channel partner is responsible for first line support. Gamma Telecom is responsible for second line support. The second line support process is outlined in the document overleaf. It is completely integrated into existing support processes for existing Gamma services, and as such, has. Familiar support processes Integration into trouble ticket Single front desk Defined escalation procedure Visibility of product elements via separate systems in R1 versions FAULT REPORTS FRONT DESK STATUS CHECK WEB ACCESS TO ISP Reporting Faults on-line

38 Desk Hours and Contacts 8am 10pm Monday through Saturday 8am 6pm Sundays and Public Holidays Escalation Process Single destination number fault: Initial Contact immediate Level 1 after 8 working hours Level 2 after 16 working hours Level 3 after 24 working hours Serious service fault: Initial contact immediate Level 1 after 4 working hour Level 2 after 8 working hours Level 3 after 16 working hours Initial Contact Front Desk Level One FD Supervisor, Joe Mundy Level Two Allan Mauchline, Head of Network Management Gamma Telecom Product FAQs Q1. What bandwidth is required at the customer site? A1. There are a number of factors that determine the amount of bandwidth required at a customer site, namely the number of concurrent calls, the codec employed, and the packetization of the data. For guidance on the maximum number of calls recommended on different access types, please use the Access Selector Tool on the Gamma portal. Q2. Can the access be shared with Internet access? A2. Yes, though this will be provided independently by your ISP, for information the suggested maximum number of calls indicated by the Access

39 Selector Tool assumes the IP transport is used solely for voice calls. Gamma strongly recommends that the ADSL Connection used from the end user site is dedicated for voice and not shared with any other medium, to provide the highest quality voice solution. Q3. What codecs and packetization are supported? A3. G711, and G729 codecs are supported. Currently the most common packetization is 20ms though it is likely that 10ms will become the standard in the near future. The IPDC Service supports both. Q4. Does the service support cross-language information processing (CLIP) and cross-language information retrieval (CLIR)? A3. Yes Q5. Can the number of concurrent calls be changed? A5. Yes, the number of concurrent calls can be changed once a month without charge. Gamma reserves the right to charge for more than one change per month. Q6. Do I need to configure a STUN/outbound proxy server? A7. No, the IPDC Session Border Controller will carry out Network Address Translation (NAT) traversal. Q7. What signaling protocols are supported? A7. SIP v1 and v2. Q8. How do I register the SIP phone end users? A8. The onsite customer PBX provides the user registrar. The Gamma IPDC solution will provide registrar support for the IPBX or concentration device onsite but not the individual IP Phones Q9. I have a dynamic IP address, how do I configure my equipment? A9. At present we can only accept static IP addresses or alternatively, we can use the SIP registrar functionality. Q10. Can I have value mobile? A10. No, IP Direct Connect is only configured on the premium mobile product. Q11. What customer equipment is supported? A11. The Gamma IPDC solution can support all devices that conform to the SIP v1 and v2 RFCs. To ensure compatibility of equipment and ease of installation, Gamma is continually undertaking conformance testing with equipment vendors. For the devices that have been tested, Gamma Telecom can provide configuration information and examples to assist in the Channel Partner s installation of this equipment. If connection is required for a device that has not previously been connected to the Gamma Network, we will first connect the customer s equipment to a lab machine. For the latest information on tested devices, please refer to the Gamma Channel Partners Portal. Q12. Do you support call parking, call hold and conferencing?

40 A12. These features are supported for most manufacturers, however we cannot guarantee inter-working with all vendors due to differing vendor implementations. Q13. Is fax supported? A13. G711 pass-through is supported for the provision of fax services. Please refer to your equipment vendor s specification for the correct configuration of this facility. Due to the differing implementations of T38 fax by equipment vendors, Gamma does not provide support for this service across the IP Direct Connect service. Document and Software Copyrights Copyright 2007 by ShoreTel, Inc., Sunnyvale, California, U.S.A. All rights reserved. Printed in the United States of America. Contents of this publication may not be reproduced or transmitted in any form or by any means, electronic or mechanical, for any purpose, without prior written authorization of ShoreTel Communications, Inc. ShoreTel, Inc. reserves the right to make changes without notice to the specifications and materials contained herein and shall not be responsible for any damage (including consequential) caused by reliance on the materials presented, including, but not limited to typographical, arithmetic or listing errors. Trademarks The ShoreTel logo, ShoreTel, ShoreCare, ShoreGear, ShoreWare and ControlPoint are registered trademarks of ShoreTel, Inc. in the United States and/or other countries. ShorePhone is a trademark of ShoreTel, Inc. in the United States and/or other countries. All other copyrights and trademarks herein are the property of their respective owners. Disclaimer To be ShoreTel Certified means that Technology Partner's product will inter-operate with the ShoreTel system, but ShoreTel does not certify that the features or functionality of Technology Partner's product will perform as specified by Technology Partner nor that Technology Partner's product will meet your specific application needs or requirements. To interoperate means that Technology Partner's product is able to exchange, use and share information with the ShoreTel system. Company Information ShoreTel, Inc. 960 Stewart Drive

ShoreTel, Ingate & BandTel for SIP Trunking

ShoreTel, Ingate & BandTel for SIP Trunking TPP APP NOTE TPP-10066 Date: 1/25/08 Product: ShoreTel Ingate BandTel System version: ShoreTel 7.0 & 7.5 ShoreTel, Ingate & BandTel for SIP Trunking SIP Trunking allows the use of Session Initiation Protocol

More information

ShoreTel, Ingate & AireSpring for SIP Trunking

ShoreTel, Ingate & AireSpring for SIP Trunking TPP APP NOTE TPP-10102 Date: September 2008 Product: ShoreTel Ingate AireSpring System version: ShoreTel 8.0 ShoreTel, Ingate & AireSpring for SIP Trunking SIP trunking allows the use of Session Initiation

More information

Skype Connect Getting Started Guide

Skype Connect Getting Started Guide A P P N O T E TPP-10251 Date : September 2010 Product: ShoreTel Ingate Skype System version: ShoreTel 10.x Skype Connect Getting Started Guide SIP Trunking allows the use of Session Initiation Protocol

More information

ShoreTel, Ingate & XO for SIP Trunking

ShoreTel, Ingate & XO for SIP Trunking I n n o v a t i o n N e t w o r k A p p N o t e TPP- 13027 Date : March 2013 Product: ShoreTel Ingate XO System version: ShoreTel 13.x ShoreTel, Ingate & XO for SIP Trunking SIP Trunking allows the use

More information

ShoreTel, Ingate & Broadvox for SIP Trunking

ShoreTel, Ingate & Broadvox for SIP Trunking I n n o v a t i o n N e t w o r k A p p N o t e IN- 13056 Date : Sept, 2013 Product: ShoreTel Ingate Broadvox System version: ShoreTel 13.x ShoreTel, Ingate & Broadvox for SIP Trunking SIP Trunking allows

More information

Application Note. IP8000 Conference Phone Configuration Guide. Table of Contents. Overview. Requirements. ST-10010 October 25, 2007

Application Note. IP8000 Conference Phone Configuration Guide. Table of Contents. Overview. Requirements. ST-10010 October 25, 2007 Application Note ST-10010 October 25, 2007 IP8000 Conference Phone Configuration Guide This application note provides the details on adding the IP8000 conference phone to the ShoreTel IP phone system.

More information

TelePacific I n n o v a t i o n

TelePacific I n n o v a t i o n TelePacific I n n o v a t i o n N e t w o r k A p p N o t e TPP- 10468 Date : January, 2013 Product: ShoreTel Ingate TelePacific System version: ShoreTel 13.x ShoreTel, Ingate & TelePacific for SIP Trunking

More information

ShoreTel & AMTELCO Infinity Console via SIP Trunking (Native)

ShoreTel & AMTELCO Infinity Console via SIP Trunking (Native) Product: ShoreTel AMTELCO Infinity Console I n n o v a t i o n N e t w o r k A p p N o t e IN-15063 Date : October, 2015 System version: ShoreTel 14.2 ShoreTel & AMTELCO Infinity Console via SIP Trunking

More information

TPP- 10421 Date: May, 2012 Product: ShoreTel Ingate VoIP Unlimited System version: ShoreTel 11.2

TPP- 10421 Date: May, 2012 Product: ShoreTel Ingate VoIP Unlimited System version: ShoreTel 11.2 I n n o v a t i o n N e t w o r k A p p N o t e TPP- 10421 Date: May, 2012 Product: ShoreTel Ingate VoIP Unlimited System version: ShoreTel 11.2 ShoreTel, Ingate & VoIP Unlimited for SIP Trunking SIP trunking

More information

Multi-Tech FaxFinder IP

Multi-Tech FaxFinder IP Multi-Tech FaxFinder IP I n n o v a t i o n N e t w o r k A p p N o t e TPP-10324 Date: June, 2011 Product: ShoreTel FaxFinder IP System Version: ShoreTel 11.X Abstract By integrating FaxFinder IP with

More information

ThinkTel. IN - 13024 Date : May 2013 Product: ShoreTel Ingate ThinkTel System version: ShoreTel 13.x

ThinkTel. IN - 13024 Date : May 2013 Product: ShoreTel Ingate ThinkTel System version: ShoreTel 13.x ThinkTel Innovation Network App Note IN - 13024 Date : May 2013 Product: ShoreTel Ingate ThinkTel System version: ShoreTel 13.x ShoreTel, Ingate & ThinkTel for SIP Trunking SIP Trunking allows the use

More information

TPP-10096 Date: December, 2008 Product: ShoreTel EtherSpeak SureTrunk System version: ShoreTel 8.1

TPP-10096 Date: December, 2008 Product: ShoreTel EtherSpeak SureTrunk System version: ShoreTel 8.1 TPP APP NOTE TPP-10096 Date: December, 2008 Product: ShoreTel EtherSpeak SureTrunk System version: ShoreTel 8.1 Abstract SIP Trunking allows the use of Session Initiation Protocol (SIP) communications

More information

Configuration Guide For Use with tipicall s SIP Trunking Service

Configuration Guide For Use with tipicall s SIP Trunking Service Product: ShoreTel Ingate tipicall SIP I n n o v a t i o n A p p N o t e N e t w o r k IN-15005 Date: February 2015 System version: ShoreTel 14.2 Ingate SIParator 5.0.3 Configuration Guide For Use with

More information

ShorePhone IP 8000 Conference Phone Configuration Guide

ShorePhone IP 8000 Conference Phone Configuration Guide Application Note ST-10070 April 16, 2008 ShorePhone IP 8000 Conference Phone Configuration Guide This application note provides the details on adding the IP 8000 conference phone to the ShoreTel Unified

More information

Application Notes for Avaya IP Office 7.0 Integration with Skype Connect R2.0 Issue 1.0

Application Notes for Avaya IP Office 7.0 Integration with Skype Connect R2.0 Issue 1.0 Avaya Solution & Interoperability Test Lab Application Notes for Avaya IP Office 7.0 Integration with Skype Connect R2.0 Issue 1.0 Abstract These Application Notes describe the steps to configure an Avaya

More information

TPP: 10096 Date: September, 2011 Product: ShoreTel EtherSpeak System version: ShoreTel 11.x

TPP: 10096 Date: September, 2011 Product: ShoreTel EtherSpeak System version: ShoreTel 11.x I n n o v a t i o n N e t w o r k A p p N o t e TPP: 10096 Date: September, 2011 Product: ShoreTel EtherSpeak System version: ShoreTel 11.x Abstract In 2008, EtherSpeak certified the SureTrunk for ShoreTel

More information

Application Notes for Configuring Cablevision Optimum Voice SIP Trunking with Avaya IP Office - Issue 1.1

Application Notes for Configuring Cablevision Optimum Voice SIP Trunking with Avaya IP Office - Issue 1.1 Avaya Solution & Interoperability Test Lab Application Notes for Configuring Cablevision Optimum Voice SIP Trunking with Avaya IP Office - Issue 1.1 Abstract These Application Notes describe the procedures

More information

Avaya IP Office 8.1 Configuration Guide

Avaya IP Office 8.1 Configuration Guide Avaya IP Office 8.1 Configuration Guide Performed By tekvizion PVS, Inc. Contact: 214-242-5900 www.tekvizion.com Revision: 1.1 Date: 10/14/2013 Copyright 2013 by tekvizion PVS, Inc. All Rights Reserved.

More information

nexvortex Setup Template

nexvortex Setup Template nexvortex Setup Template ZULTYS, INC. April 2013 5 1 0 S P R I N G S T R E E T H E R N D O N V A 2 0 1 7 0 + 1 8 5 5. 6 3 9. 8 8 8 8 Introduction This document is intended only for nexvortex customers

More information

Application Notes for BT Wholesale/HIPCOM SIP Trunk Service and Avaya IP Office 8.0 Issue 1.0

Application Notes for BT Wholesale/HIPCOM SIP Trunk Service and Avaya IP Office 8.0 Issue 1.0 Avaya Solution & Interoperability Test Lab Application Notes for BT Wholesale/HIPCOM SIP Trunk Service and Avaya IP Office 8.0 Issue 1.0 Abstract These Application Notes describe the procedures for configuring

More information

IP PBX. SD Card Slot. FXO Ports. PBX WAN port. FXO Ports LED, RED means online

IP PBX. SD Card Slot. FXO Ports. PBX WAN port. FXO Ports LED, RED means online 1 IP PBX SD Card Slot FXO Ports PBX LAN port PBX WAN port FXO Ports LED, RED means online 2 Connect the IP PBX to Your LAN Internet PSTN Router Ethernet Switch FXO Ports 3 Access the PBX s WEB GUI The

More information

Using FaxFinder with ShoreTel. Application Notes

Using FaxFinder with ShoreTel. Application Notes Using FaxFinder with ShoreTel Application Notes Using FaxFinder with ShoreTel Application Notes ShoreTel Part Number IN-14045, Document Version 0.08 Copyright This publication may not be reproduced, in

More information

Application Notes for Configuring Broadvox SIP Trunking with Avaya IP Office - Issue 1.0

Application Notes for Configuring Broadvox SIP Trunking with Avaya IP Office - Issue 1.0 Avaya Solution & Interoperability Test Lab Application Notes for Configuring Broadvox SIP Trunking with Avaya IP Office - Issue 1.0 Abstract These Application Notes describe the procedures for configuring

More information

Product: ShoreTel EtherSpeak System version: ShoreTel 14.1

Product: ShoreTel EtherSpeak System version: ShoreTel 14.1 Innovation Network App Note for ShoreTel Native & Ingate 5.x (Physical/Virtual) IN10449 Date: June, 2014 Product: ShoreTel EtherSpeak System version: ShoreTel 14.1 Abstract EtherSpeak certified the SureTrunk

More information

Application Note Configuring the Synapse SB67070 SIP Gateway for Broadvox GO! SIP Trunking

Application Note Configuring the Synapse SB67070 SIP Gateway for Broadvox GO! SIP Trunking Configuring the Synapse SB67070 SIP Gateway for Broadvox GO! SIP Trunking 2012 Advanced American Telephones. All Rights Reserved. AT&T and the AT&T logo are trademarks of AT&T Intellectual Property licensed

More information

AT&T IP Flex Reach/ IP Toll Free Configuration Guide IC 3.0 with Interaction SIP Proxy

AT&T IP Flex Reach/ IP Toll Free Configuration Guide IC 3.0 with Interaction SIP Proxy INTERACTIVE INTELLIGENCE AT&T IP Flex Reach/ IP Toll Free Configuration Guide IC 3.0 with Interaction SIP Proxy Version 1.7 9/2/2009 TABLE OF CONTENTS 1 AT&T... 5 1.1 Introduction... 5 1.2 Product Descriptions...

More information

Application Note Startup Tool - Getting Started Guide

Application Note Startup Tool - Getting Started Guide Application Note Startup Tool - Getting Started Guide 1 April 2012 Startup Tool Table of Contents 1 INGATE STARTUP TOOL... 1 2 STARTUP TOOL INSTALLATION... 2 3 CONNECTING THE INGATE FIREWALL/SIPARATOR...

More information

SHORETEL APPLICATION NOTE

SHORETEL APPLICATION NOTE SHORETEL APPLICATION NOTE for Bell Canada SIP Trunking via Ingate Date: April 13, 2016 App Note Number: TC - 16032 For use with: Product: System: Bell Canada SIP Trunking via Ingate ShoreTel Connect ONSITE

More information

Configuring Interactive Intelligence ININ IP PBX For tw telecom SIP Trunking service USER GUIDE

Configuring Interactive Intelligence ININ IP PBX For tw telecom SIP Trunking service USER GUIDE Configuring Interactive Intelligence ININ IP PBX For tw telecom SIP Trunking service USER GUIDE Version 1.0 August 23, 2012 Copyright 2012 by tw telecom inc. All Rights Reserved. This document is the property

More information

SIP Trunking Quick Reference Document

SIP Trunking Quick Reference Document SIP Trunking Quick Reference Document Publication Information SAMSUNG TELECOMMUNICATIONS AMERICA reserves the right without prior notice to revise information in this publication for any reason. SAMSUNG

More information

OfficeMaster Gate (Virtual) Enterprise Session Border Controller for Microsoft Lync Server. Quick Start Guide

OfficeMaster Gate (Virtual) Enterprise Session Border Controller for Microsoft Lync Server. Quick Start Guide OfficeMaster Gate (Virtual) Enterprise Session Border Controller for Microsoft Lync Server Quick Start Guide October 2013 Copyright and Legal Notice. All rights reserved. No part of this document may be

More information

Technical Configuration Notes

Technical Configuration Notes MITEL SIP CoE Technical Configuration Notes Configure MCD for use with OpenIP SIP Trunking service SIP CoE 11-4940-00186 NOTICE The information contained in this document is believed to be accurate in

More information

Technical Configuration Notes

Technical Configuration Notes MITEL SIPCoE Technical Configuration Notes Configure Inn-Phone SIP Phone for use with MCD SIP CoE NOTICE The information contained in this document is believed to be accurate in all respects but is not

More information

EarthLink Business SIP Trunking. ININ IC3 IP PBX Customer Configuration Guide

EarthLink Business SIP Trunking. ININ IC3 IP PBX Customer Configuration Guide EarthLink Business SIP Trunking ININ IC3 IP PBX Customer Configuration Guide Publication History First Release: Version 1.0 August 30, 2011 CHANGE HISTORY Version Date Change Details Changed By 1.0 8/30/2011

More information

Setup Reference Guide for KX-NS1000 to SBC SIP Trunking

Setup Reference Guide for KX-NS1000 to SBC SIP Trunking Setup Reference Guide for KX-NS1000 to SBC SIP Trunking Method of connection by "WAN Global IP address directly" (i.e. SBC is the Perimeter Router device.) Panasonic IP-PBX (KX-NS1000 Version2 series),

More information

Cisco Unified Communications 500 Series

Cisco Unified Communications 500 Series Cisco Unified Communications 500 Series IP PBX Provisioning Guide Version 1.0 Last Update: 02/14/2011 Page 1 DISCLAIMER The attached document is provided as a basic guideline for setup and configuration

More information

Mediatrix 4404 Step by Step Configuration Guide June 22, 2011

Mediatrix 4404 Step by Step Configuration Guide June 22, 2011 Mediatrix 4404 Step by Step Configuration Guide June 22, 2011 Proprietary 2011 Media5 Corporation Table of Contents First Steps... 3 Identifying your MAC Address... 3 Identifying your Dynamic IP Address...

More information

Voice over IP Basics for IT Technicians

Voice over IP Basics for IT Technicians Voice over IP Basics for IT Technicians White Paper Executive summary The IP phone is coming or has arrived on desk near you. The IP phone is not a PC, but does have a number of hardware and software elements

More information

How To Guide. SIP Trunking Configuration Using the SIP Trunk Page

How To Guide. SIP Trunking Configuration Using the SIP Trunk Page How To Guide SIP Trunking Configuration Using the SIP Trunk Page For the Ingate SIParators and Firewalls using software release 4.9.2 or later. Updated to show features available from release 4.10.x May

More information

Setup Reference Guide for KX-TDE/NCP to SBC SIP Trunking

Setup Reference Guide for KX-TDE/NCP to SBC SIP Trunking Setup Reference Guide for KX-TDE/NCP to SBC SIP Trunking Method of connection by "WAN Global IP address directly" (i.e. SBC is the Perimeter Router device.) Panasonic IP-PBX (KX-TDE/NCP Version6 series),

More information

ShoreTel Installation Guide

ShoreTel Installation Guide Version 5.2.2 ShoreTel Installation Guide December 22, 2010 Copyright 2010 FaxCore, Inc 1 P a g e Abstract By integrating FaxCore with the ShoreTel System, the customer can have a complete telephony solution

More information

Application Notes for Configuring Intelepeer SIP Trunking with Avaya IP Office 7.0 - Issue 1.0

Application Notes for Configuring Intelepeer SIP Trunking with Avaya IP Office 7.0 - Issue 1.0 Avaya Solution & Interoperability Test Lab Application Notes for Configuring Intelepeer SIP Trunking with Avaya IP Office 7.0 - Issue 1.0 Abstract These Application Notes describe the procedures for configuring

More information

Configuration Notes 283

Configuration Notes 283 Mediatrix 4400 Digital Gateway VoIP Trunking with a Legacy PBX June 21, 2011 Proprietary 2011 Media5 Corporation Table of Contents Table of Contents... 2 Introduction... 3 Mediatrix 4400 Digital Gateway

More information

Optimum Business SIP Trunk Set-up Guide

Optimum Business SIP Trunk Set-up Guide Optimum Business SIP Trunk Set-up Guide For use with IP PBX only. SIPSetup 07.13 FOR USE WITH IP PBX ONLY Important: If your PBX is configured to use a PRI connection, do not use this guide. If you need

More information

Application Notes for Configuring Avaya IP Office 9.0 with HIPCOM SIP Trunk Issue 1.0

Application Notes for Configuring Avaya IP Office 9.0 with HIPCOM SIP Trunk Issue 1.0 Avaya Solution & Interoperability Test Lab Application Notes for Configuring Avaya IP Office 9.0 with HIPCOM SIP Trunk Issue 1.0 Abstract These Application Notes describe the procedures for configuring

More information

SIP Trunking using Optimum Business SIP Trunk Adaptor and ShoreTel IP PBX Phone System

SIP Trunking using Optimum Business SIP Trunk Adaptor and ShoreTel IP PBX Phone System SHORETEL SIP Trunking using Optimum Business SIP Trunk Adaptor and ShoreTel IP PBX Phone System Goal The purpose of this configuration guide is to describe the steps needed to configure the ShoreTel IP

More information

MITEL SIP CoE Technical. Configuration Note. Configure MCD for use with Thinktel SIP Trunking Service. SIP CoE 12-4940-00197

MITEL SIP CoE Technical. Configuration Note. Configure MCD for use with Thinktel SIP Trunking Service. SIP CoE 12-4940-00197 MITEL SIP CoE Technical Configuration Note Configure MCD for use with SIP Trunking Service SIP CoE NOTICE The information contained in this document is believed to be accurate in all respects but is not

More information

Mediatrix 3000 with Asterisk June 22, 2011

Mediatrix 3000 with Asterisk June 22, 2011 Mediatrix 3000 with Asterisk June 22, 2011 Proprietary 2011 Media5 Corporation Table of Contents Introduction... 3 Network Topology... 3 Equipment Detail... 3 Configuration of the Fax Extension... 4 Configuration

More information

TPP-10024 Date: March 2010 Product: ShoreTel NICE System version: ShoreTel 9.2

TPP-10024 Date: March 2010 Product: ShoreTel NICE System version: ShoreTel 9.2 T P P A P P N O T E TPP-10024 Date: March 2010 Product: ShoreTel NICE System version: ShoreTel 9.2 Abstract The NICE recording suite is a comprehensive solution that enables ShoreTel customers to perform

More information

nexvortex SIP Trunking Implementation & Planning Guide V1.5

nexvortex SIP Trunking Implementation & Planning Guide V1.5 nexvortex SIP Trunking Implementation & Planning Guide V1.5 510 S PRING S TREET H ERNDON VA 20170 +1 855.639.8888 Introduction Welcome to nexvortex! This document is intended for nexvortex Customers and

More information

Fonality. Optimum Business Trunking and the Fonality Trixbox Pro IP PBX Standard Edition V4.1.2- p13 Configuration Guide

Fonality. Optimum Business Trunking and the Fonality Trixbox Pro IP PBX Standard Edition V4.1.2- p13 Configuration Guide Fonality Optimum Business Trunking and the Fonality Trixbox Pro IP PBX Standard Edition V4.1.2- p13 Configuration Guide Fonality Table of Contents 1. Overview 2. SIP Trunk Adaptor Set-up Instructions 3.

More information

MITEL SIP CoE. Technical. Configuration Notes. Configure MCD 6.X for use with babytel SIP trunks. SIP CoE 13-4940-00266

MITEL SIP CoE. Technical. Configuration Notes. Configure MCD 6.X for use with babytel SIP trunks. SIP CoE 13-4940-00266 MITEL SIP CoE Technical Configuration Notes Configure MCD 6.X for use with babytel SIP trunks SIP CoE 13-4940-00266 NOTICE The information contained in this document is believed to be accurate in all respects

More information

Configuring SIP Trunking and Networking for the NetVanta 7000 Series

Configuring SIP Trunking and Networking for the NetVanta 7000 Series 61200796L1-29.4E July 2011 Configuration Guide Configuring for the NetVanta 7000 Series This configuration guide describes the configuration and implementation of Session Initiation Protocol (SIP) trunking

More information

TELEPHONE MAN OF AMERICA. Earning Your Business Every Step of the Way!

TELEPHONE MAN OF AMERICA. Earning Your Business Every Step of the Way! TELEPHONE MAN OF AMERICA Earning Your Business Every Step of the Way! Specializing in Telecom Equipment of all Brands, Carrier Services, Technician Services, Maintenance Agreements & Purchasing Excess

More information

ADTRAN SBC and Avaya IP Office PBX SIP Trunk Interoperability

ADTRAN SBC and Avaya IP Office PBX SIP Trunk Interoperability 6AOSSG001-42B March 2014 Interoperability Guide ADTRAN SBC and Avaya IP Office PBX SIP Trunk Interoperability This guide describes an example configuration used in testing the interoperability of an ADTRAN

More information

Configuration Notes 290

Configuration Notes 290 Configuring Mediatrix 41xx FXS Gateway with the Asterisk IP PBX System June 22, 2011 Proprietary 2011 Media5 Corporation Table of Contents Introduction... 3 About Mediatrix 41xx Series FXS Gateways...

More information

Version Date Status Owner. 1.0 2009-02-04 Released for HiPath OpenOffice ME V1 F. Kneissl / K.-W. Weigt

Version Date Status Owner. 1.0 2009-02-04 Released for HiPath OpenOffice ME V1 F. Kneissl / K.-W. Weigt History of Change Version Date Status Owner 1.0 2009-02-04 Released for HiPath OpenOffice ME V1 F. Kneissl / K.-W. Weigt 1.1 2010-09-01 Update for OpenScape Office MX V2 and hints for Fax F. Kneissl 1.2

More information

APPLICATION NOTE: AN10440

APPLICATION NOTE: AN10440 APPLICATION NOTE: AN10440 Integrating ShoreTel with Microsoft Lync via AudioCodes IP to IP (SIP) Microsoft Lync 2010 and AudioCodes Mediant 1000 Media Gateway Contents Overview... 2 Required Components...

More information

Internet Telephony PBX System. IPX-300 Series. Quick Installation Guide

Internet Telephony PBX System. IPX-300 Series. Quick Installation Guide Internet Telephony PBX System IPX-300 Series Quick Installation Guide Overview PLANET IPX-300/IPX-300W IP PBX telephony systems ( IP PBX in the following term) are designed and optimized for the small

More information

EarthLink Business SIP Trunking. NEC SV8100 IP PBX Customer Configuration Guide

EarthLink Business SIP Trunking. NEC SV8100 IP PBX Customer Configuration Guide EarthLink Business SIP Trunking NEC SV8100 IP PBX Customer Configuration Guide Publication History First Release: Version 1.0 August 30, 2011 CHANGE HISTORY Version Date Change Details Changed By 1.0 8/30/2011

More information

ESI SIP Trunking Installation Guide

ESI SIP Trunking Installation Guide ESI SIP Trunking Installation Guide 0450-1227 Rev. B Copyright 2009 ESI (Estech Systems, Inc.). Information contained herein is subject to change without notice. ESI products are protected by various U.S.

More information

MITEL SIP CoE. Technical. Configuration Notes. Configure the Mitel 3300 MCD 4.1 for use with Paetec Broadworks Softswitch. SIP CoE 08-4940-00035

MITEL SIP CoE. Technical. Configuration Notes. Configure the Mitel 3300 MCD 4.1 for use with Paetec Broadworks Softswitch. SIP CoE 08-4940-00035 MITEL SIP CoE Technical Configuration Notes Configure the Mitel 3300 MCD 4.1 for use with Broadworks Softswitch SIP CoE 08-4940-00035 NOTICE The information contained in this document is believed to be

More information

SIP Proxy Server. Administrator Installation and Configuration Guide. V2.31b. 09SIPXM.SY2.31b.EN3

SIP Proxy Server. Administrator Installation and Configuration Guide. V2.31b. 09SIPXM.SY2.31b.EN3 SIP Proxy Server Administrator Installation and Configuration Guide V2.31b 09SIPXM.SY2.31b.EN3 DSG, DSG logo, InterPBX, InterServer, Blaze Series, VG5000, VG7000, IP590, IP580, IP500, IP510, InterConsole,

More information

Voice over IP (VoIP) Basics for IT Technicians

Voice over IP (VoIP) Basics for IT Technicians Voice over IP (VoIP) Basics for IT Technicians VoIP brings a new environment to the network technician that requires expanded knowledge and tools to deploy and troubleshoot IP phones. This paper provides

More information

Application Notes for Configuring SIP Trunking between Metaswitch MetaSphere CFS and Avaya IP Office Issue 1.0

Application Notes for Configuring SIP Trunking between Metaswitch MetaSphere CFS and Avaya IP Office Issue 1.0 Avaya Solution & Interoperability Test Lab Application Notes for Configuring SIP Trunking between Metaswitch MetaSphere CFS and Avaya IP Office Issue 1.0 Abstract These Application Notes describe the steps

More information

VoIP Network Configuration Guide

VoIP Network Configuration Guide The owner friendly phone system for small business VoIP Network Configuration Guide Release 7.10 Copyright 2011 Fortinet, Inc. All rights reserved. Fortinet, FortiGate, FortiGuard, FortiCare, FortiManager,

More information

EarthLink Business SIP Trunking. Shoretel IP PBX Customer Configuration Guide

EarthLink Business SIP Trunking. Shoretel IP PBX Customer Configuration Guide EarthLink Business SIP Trunking Shoretel IP PBX Customer Configuration Guide Publication History First Release: Version 2.0 April 20, 2012 CHANGE HISTORY Version Date Change Details Changed By 1.0 4/16/2011

More information

EarthLink Business SIP Trunking. NEC SV8300 IP PBX Customer Configuration Guide

EarthLink Business SIP Trunking. NEC SV8300 IP PBX Customer Configuration Guide EarthLink Business SIP Trunking NEC SV8300 IP PBX Customer Configuration Guide Publication History First Release: Version 1.0 May 18, 2012 CHANGE HISTORY Version Date Change Details Changed By 1.0 5/18/2012

More information

nexvortex Setup Guide

nexvortex Setup Guide nexvortex Setup Guide CISCO UC500 March 2012 Introduction This document is intended only for nexvortex customers and resellers as an aid to setting up the Cisco PBX software to connect to the nexvortex

More information

FortiVoice. Version 7.00 VoIP Configuration Guide

FortiVoice. Version 7.00 VoIP Configuration Guide FortiVoice Version 7.00 VoIP Configuration Guide FortiVoice Version 7.00 VoIP Configuration Guide Revision 2 14 October 2011 Copyright 2011 Fortinet, Inc. All rights reserved. Contents and terms are subject

More information

TOSHIBA. Optimum Business Trunking and the Toshiba IPedge IP PBX V Configuration Guide

TOSHIBA. Optimum Business Trunking and the Toshiba IPedge IP PBX V Configuration Guide TOSHIBA Optimum Business Trunking and the Toshiba IPedge IP PBX V1.1.3-06 Configuration Guide TOSHIBA Table of Contents 1. Overview 2. SIP Trunk Adaptor Set-up Instructions 3. Additional Set-up Information

More information

IP Office Technical Tip

IP Office Technical Tip IP Office Technical Tip Tip no: 188 Release Date: September 27, 2007 Region: GLOBAL Verifying IP Office SIP Trunk Operation IP Office back-to-back SIP Line testing IP Office Release 4.0 supports SIP trunking.

More information

SIP Trunking Application Notes V1.3

SIP Trunking Application Notes V1.3 SIP Trunking Application Notes V1.3 Publication Information SAMSUNG TELECOMMUNICATIONS AMERICA reserves the right without prior notice to revise information in this publication for any reason. SAMSUNG

More information

SIP Trunking Configuration with

SIP Trunking Configuration with SIP Trunking Configuration with Microsoft Office Communication Server 2007 R2 A Dell Technical White Paper End-to-End Solutions Team Dell Product Group - Enterprise THIS WHITE PAPER IS FOR INFORMATIONAL

More information

VOIP NETWORK CONFIGURATION GUIDE RELEASE 6.10

VOIP NETWORK CONFIGURATION GUIDE RELEASE 6.10 TALKSWITCH DOCUMENTATION VOIP NETWORK CONFIGURATION GUIDE RELEASE 6.10 CT.TS005.002606 ANSWERS WITH INTELLIGENCE INTRODUCTION About this guide This guide will help you plan and configure a TalkSwitch system

More information

SIP Trunking using Optimum Business SIP Trunk Adaptor and the Cisco Call Manager Express Version 8.5

SIP Trunking using Optimum Business SIP Trunk Adaptor and the Cisco Call Manager Express Version 8.5 CISCO SIP Trunking using Optimum Business SIP Trunk Adaptor and the Cisco Call Manager Express Version 8.5 Goal The purpose of this configuration guide is to describe the steps needed to configure the

More information

#!) * & /! $* - 01 $& -$ 2 1 $& -# 32# $- - + $- -*!45 $-

#!) * & /! $* - 01 $& -$ 2 1 $& -# 32# $- - + $- -*!45 $- ! Version Date Status Owner 1.0 2009-11-09 Released 1.1 2010-01-05 Hints for Fax added Provider hints and new layout 1.2 2010-12-17 Additional hints and corrections 1.3 2011-05-19 Updates for new STUN

More information

MITEL SIP CoE. Technical. Configuration Note. Configure MCD for use with Intelepeer Service provider SIP Trunking. SIP CoE 14-4940-00313

MITEL SIP CoE. Technical. Configuration Note. Configure MCD for use with Intelepeer Service provider SIP Trunking. SIP CoE 14-4940-00313 MITEL SIP CoE Technical Configuration Note Configure MCD for use with Intelepeer Service provider SIP Trunking SIP CoE 14-4940-00313 NOTICE The information contained in this document is believed to be

More information

Vega 100G and Vega 200G Gamma Config Guide

Vega 100G and Vega 200G Gamma Config Guide Vega 100G and Vega 200G Gamma Config Guide This document aims to go through the steps necessary to configure the Vega 100G and 200G gateways to be used with a Gamma SIP Trunk. When a SIP trunk is provisioned

More information

TPP: 10273 Date: June, 2010 Product: ShoreTel ADTRAN System version: ShoreTel 10.x

TPP: 10273 Date: June, 2010 Product: ShoreTel ADTRAN System version: ShoreTel 10.x T P P A P P N O T E TPP: 10273 Date: June, 2010 Product: ShoreTel ADTRAN System version: ShoreTel 10.x Abstract The ShoreTel system has some known limitations with SIP trunking, please refer to ShoreTel

More information

1.1.3 Versions Verified SIP Carrier status as of 18 Sep 2014 : validated on CIC 4.0 SU6.

1.1.3 Versions Verified SIP Carrier status as of 18 Sep 2014 : validated on CIC 4.0 SU6. 1 SIP Carriers 1.1 Telstra 1.1.1 Warnings Check the SIP 3 rd Party SIP Carrier Matrix for certification status, and supported features. More info about the SIP 3 rd Party SIP Carrier Matrix can be found

More information

Setup Reference guide for PBX to SBC interconnection

Setup Reference guide for PBX to SBC interconnection Setup Reference guide for PBX to SBC interconnection Method of connection by "LAN interface only" i.e. SBC is placed behind the Perimeter Router / Fire-wall. Panasonic PBX (KX-TDE, NCP series), Media5

More information

Configuration of the Intertex IX78 E-SBC with IP-PBXs and Telia SIP Trunking Services

Configuration of the Intertex IX78 E-SBC with IP-PBXs and Telia SIP Trunking Services 2010-03-07 Configuration of the Intertex IX78 E-SBC with IP-PBXs and Telia SIP Trunking Services For the IX78 ADSL T8 GW2 model with Telia SIP-anslutning using Telia Bredband Företag or Telia ProLane Table

More information

EarthLink Business SIP Trunking. Switchvox SMB 5.5 & Adtran SIP Proxy Implementation Guide

EarthLink Business SIP Trunking. Switchvox SMB 5.5 & Adtran SIP Proxy Implementation Guide EarthLink Business SIP Trunking Switchvox SMB 5.5 & Adtran SIP Proxy Implementation Guide Publication History First Release: Version 1.0 April 20, 2012 CHANGE HISTORY Version Date Change Details Changed

More information

Siemens OpenScape Voice V7 SIP Connectivity with OpenScape SBC V7. to Integra SIP Service

Siemens OpenScape Voice V7 SIP Connectivity with OpenScape SBC V7. to Integra SIP Service Siemens OpenScape Voice V7 SIP Connectivity with OpenScape SBC V7 to Integra SIP Service February 2013 Page 1 of 47 Document History Date Version / Issue Modified by Section(s) Affected High-Level Description

More information

Technical Configuration Notes

Technical Configuration Notes MITEL SIPCoE Technical Configuration Notes Configure the MCD 6.0 for use with the Commend SIP Doorphone SIP CoE 13-4940-00251 NOTICE The information contained in this document is believed to be accurate

More information

Application Notes for Configuring SIP Trunking between McLeodUSA SIP Trunking Solution and an Avaya IP Office Telephony Solution 1.

Application Notes for Configuring SIP Trunking between McLeodUSA SIP Trunking Solution and an Avaya IP Office Telephony Solution 1. Avaya Solution & Interoperability Test Lab Application Notes for Configuring SIP Trunking between McLeodUSA SIP Trunking Solution and an Avaya IP Office Telephony Solution 1.0 Abstract These Application

More information

VoIP CONFIGURATION GUIDE FOR MULTI-LOCATION NETWORKS

VoIP CONFIGURATION GUIDE FOR MULTI-LOCATION NETWORKS VoIP CONFIGURATION GUIDE FOR MULTI-LOCATION NETWORKS INTRODUCTION About this guide This guide is designed to help you plan and configure a TalkSwitch multi-location network for Voice over IP (VoIP). NOTE:

More information

Configuring for Integra Telecom SIP Solutions

Configuring for Integra Telecom SIP Solutions February 2013 Configuring for Integra Telecom SIP Solutions Section Title Page Background 1 Table 1 Software & Firmware Versions Tested 1 Figure 1 Generic IP PBX Test Configuration (with Transparent SIP

More information

SIP Trunking with Allworx. Configuration Guide for Matrix SETU VoIP Gateways

SIP Trunking with Allworx. Configuration Guide for Matrix SETU VoIP Gateways SIP Trunking with Allworx Configuration Guide for Matrix SETU VoIP Gateways Disclaimer Matrix Comsec Pvt. Ltd. reserves the right to change, at any time, without prior notice, the product design, specifications,

More information

Technical Configuration Notes

Technical Configuration Notes MITEL SIPCoE Technical Configuration Notes Configure Mitel UC360 SIP Phone and Mitel MCD for use with VidyoWay SIP CoE 13-4940-00228 NOTICE The information contained in this document is believed to be

More information

AudioCodes. MP-20x Telephone Adapter. Frequently Asked Questions (FAQs)

AudioCodes. MP-20x Telephone Adapter. Frequently Asked Questions (FAQs) AudioCodes MP-20x Telephone Adapter Frequently Asked Questions (FAQs) Page 2 AudioCodes Customer Support Table of Contents Introduction... 6 Frequently Asked Questions... 7 Web Access... 7 Q1: How must

More information

Wave SIP Trunk Configuration Guide FOR BROADVOX

Wave SIP Trunk Configuration Guide FOR BROADVOX Wave SIP Trunk Configuration Guide FOR BROADVOX Last updated 1/7/2014 Contents Overview... 1 Special Notes... 1 Before you begin... 1 Required SIP trunk provisioning and configuration information... 1

More information

OpenScape Business V2

OpenScape Business V2 OpenScape Business V2 Tutorial System Device@Home Configuration Version 1.1 Table of Contents 1. Configuration Overview 4 1.1. Network Scenario Description: 4 1.2. Configuration Steps 5 1.2.1. Overview

More information

How to Configure the Cisco UC500 for use with Integra Telecom SIP Solutions

How to Configure the Cisco UC500 for use with Integra Telecom SIP Solutions How to Configure the Cisco UC500 for use with Integra Telecom SIP Solutions Overview: This document provides a reference for configuration of the Cisco UC500 IP PBX to connect to Integra Telecom SIP Trunks.

More information

Network Connection Considerations for Microsoft Response Point 1.0 Service Pack 2

Network Connection Considerations for Microsoft Response Point 1.0 Service Pack 2 Network Connection Considerations for Microsoft Response Point 1.0 Service Pack 2 Updated: February 2009 Microsoft Response Point is a small-business phone solution that is designed to be easy to use and

More information

Application Notes for Configuring Broadvox SIP Trunking with Avaya IP Office Release 8.0 Issue 1.0

Application Notes for Configuring Broadvox SIP Trunking with Avaya IP Office Release 8.0 Issue 1.0 Avaya Solution & Interoperability Test Lab Application Notes for Configuring Broadvox SIP Trunking with Avaya IP Office Release 8.0 Issue 1.0 Abstract These Application Notes describe the steps necessary

More information

MITEL SIP CoE. Technical. Configuration Notes. Configure MCD 4.1 for use with SKYPE SIP Trunking. SIP CoE 10-4940-00120

MITEL SIP CoE. Technical. Configuration Notes. Configure MCD 4.1 for use with SKYPE SIP Trunking. SIP CoE 10-4940-00120 MITEL SIP CoE Technical Configuration Notes Configure MCD 4.1 for use with SKYPE SIP Trunking SIP CoE 10-4940-00120 NOTICE The information contained in this document is believed to be accurate in all respects

More information

SIP Trunking Service Configuration Guide for Time Warner Cable Business Class

SIP Trunking Service Configuration Guide for Time Warner Cable Business Class SIP Trunking Service Configuration Guide for Time Warner Cable Business Class NDA-31669 Issue 1.0 NEC Corporation of America reserves the right to change the specifications, functions, or features at

More information

MITEL SIP CoE. Technical. Configuration Notes. Configure MCD 6.X for use with VoiceHost SIP trunks. SIP CoE

MITEL SIP CoE. Technical. Configuration Notes. Configure MCD 6.X for use with VoiceHost SIP trunks. SIP CoE MITEL SIP CoE Technical Configuration Notes Configure MCD 6.X for use with VoiceHost SIP trunks SIP CoE 13-4940-00284 NOTICE The information contained in this document is believed to be accurate in all

More information