Application Note. CenturyLink (Qwest) SIP Trunk Interoperability Configuration Guide for ShoreTel & Adtran SBC. November 15, 2013

Transcription

1 Product: ShoreTel ADTRAN CenturyLink Corporation Innovation Network App N ote IN Date: November 15, 2013 System version: ShoreTel 14.1 Application Note CenturyLink (Qwest) SIP Trunk Interoperability Configuration Guide for ShoreTel & Adtran SBC November 15, 2013 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.

2 Table of Contents APPLICATION NOTE OVERVIEW CENTURYLINK SIP TRUNKING ARCHITECTURE OVERVIEW CENTURYLINK SALES & SUPPORT INTEROPERABILITY REQUIREMENTS, VALIDATION AND LIMITATIONS VERSION SUPPORT VALIDATION RESULTS SUMMARY SHORETEL SHORETEL UNSUPPORTED FEATURES SHORETEL CONFIGURATION Call Control Settings Sites Settings Switch Settings - Allocating Ports System Settings Trunk Groups System Settings Individual Trunks ADTRAN ADTRAN PRODUCT INFORMATION ADTRAN SBC PRODUCT CONFIGURATION Network Topology Login Configuring the Adtran SBC SBC basic voice configuration rules SBC IP-PBX trunk configuration SBC ITSP CTL trunk configuration Adtran SBC Router configs ADTRAN TROUBLESHOOTING Call Flow Examples ADTRAN Troubleshooting Tools ADTRAN SALES & TECHNICAL SUPPORT Sales Technical Support DOCUMENT AND SOFTWARE COPYRIGHTS TRADEMARKS DISCLAIMER COMPANY INFORMATION pg. 2

3 1 Overview This document provides details for connecting the ShoreTel system though the Adtran SBC to CenturyLink SIP Trunk 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 Adtran SBC which is connected to both the LAN and WAN and acts as a secure gateway to CenturyLink for SIP Trunking. 1.1 CenturyLink SIP Trunking Architecture Overview 1.2 CenturyLink Sales & Support CenturyLink: pg. 3

4 2 Interoperability Requirements, Validation and Limitations 2.1 Version Support Products are certified via the Technology Partner Validation Process for the ShoreTel system. The table below contains the matrix of Adtran versions firmware releases certified on the identified ShoreTel software releases. Adtran SBC R V ShoreTel 13.3 ShoreTel 14.1 ü ü 2.2 Validation Results Summary Following are the Test Plan Results when testing with CenturyLink Corporation. Table 1: Initialization and Basic Calls ID Name Description Notes 1.1 Setup and initialization 1.2 Outbound Call (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 Verify successful setup and initialization of the system under test (SUT) Verify calls outbound placed through the 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. pg. 4

5 Table 2: Media and Dual-Tone Multi-Frequency (DTMF) Support ID Name Description Notes 2.1 Media Support ShoreTel Phone to Verify call connection and audio path from a ShoreTel phone to an external destination SUT through the service provider using all supported codecs with both sides set to a common codec. 2.1 Media Support SIP Reference to SUT Verify call connection and audio path from SIP Reference phones to an external destination through the service provider using all supported codes with both sides set to a common codec. 2.2 Codec Negotiation Verify codec negotiation between the SUT and the calling device with each side configured for a different codec. 2.3 DTMF Transmission Out of Band / In Band 2.4 Auto Attendant Menu 2.5 Auto Attendant Menu Dial by Name 2.6 Auto Attendant Menu Checking Voice Mail Mailbox Verify transmission of in-band and out-ofband digits per RFC 2833 for various devices connected to the SUT. Verify that inbound calls are properly terminated on the ShoreTel Auto Attendant menu and that you can transfer to the desired extension. Verify that inbound calls are properly terminated on the ShoreTel Auto Attendant menu and that you can transfer to the desired extension using the Dial by Name feature. Verify that inbound calls are properly terminated on the ShoreTel Auto Attendant menu and that you can transfer to the Voice Mail Login Extension. Table 3: Performance & Quality of Service ID Name Description Notes 3.1 Voice Quality Service Levels Verify the SUT can provide a voice quality service-level agreement (SLA) across the WAN from the customer premises to the SUT SIP gateway. 3.2 Capacity Test Verify the service provider interface can sustain services through period of heavy outbound and inbound load. 3.3 Post Dial Delay Verify that post dial delay is within acceptable limits. 3.4 Billing Accuracy Verify that all test calls made are accurately reflected in the SUT s Call Detail Record (CDR) and billing reports. pg. 5

6 Table 4: Enhanced Services and Features ID Name Description Notes 4.1 Caller ID Name and Number - Inbound 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 Verify that Caller ID name and number is received from SIP endpoint device 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. 4.7 Conference Ad Hoc 4.8 Inbound Direct Inward Dialing/Dialed Number Identification Service (DID/DNIS) 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. Notes: If you dial 9911 call goes out immediately. If you dial 911 there is a 5 second delay. This is documented in the ShoreTel SysAdminGuide. If the user forgets to dial an access code before dialing 911, the system waits five seconds before routing the call to a 911-capable trunk. This pause has been introduced to eliminate accidental calls to Operator Assisted Verify that 0+ calls are routed to an operator for calling assistance Inbound / Verify that calls with Blocked Caller ID Outbound call route properly and the answering phone with Blocked does not display any Caller ID Caller ID information. Conditional Pass Note: This was tested in a controlled environment without actual circuits, just to verify call placement. pg. 6

7 ID Name Description Notes 4.12 Inbound call to a Hunt Group 4.13 Inbound Call to a Workgroup 4.14 Inbound Call to DNIS/DID and Leave a Voice Mail Message 4.15 Call Forward FindMe 4.16 Call Forward Always 4.17 Inbound / Outbound Fax Calls Verify that calls route to the proper hunt group and are answered by an available hunt group member with audio in both directions using G.729 and G.711 codecs. Verify that calls route to the proper workgroup and are answered successfully by an available workgroup agent with audio in both directions using G.729 and G.711 codecs. Verify that inbound calls to a user, via DID/DNIS, routes to the proper user mailbox and a message can be left with proper audio. Verify that inbound calls are forwarded to a user s FindMe destination. Verify that inbound calls are immediately automatically forwarded to a user s external destination. Verify that inbound/outbound fax calls complete successfully ShoreTel UCB Verify that inbound calls are properly forwarded to the ShoreTel UCB, that it properly accepts the access code, and you re able to participate in the conference bridge Inbound Call to Bridged Call Appearance (BCA) Extension 4.20 Inbound Call to a Group Pickup Verify that inbound calls properly presented to all of the phones that have BCA configured and that the call can be answered, placed on-hold and then transferred. Verify that inbound calls are properly presented to all of the phones that have Group Pickup configured and that the call can be answered, placed on-hold and then transferred. Table 5: Security ID Name Description Notes N/A 5.1 Digest Authentication Verify the SUT supports the use of digest authentication for service access for inbound and outbound calls. pg. 7

8 3 ShoreTel The configuration information below shows examples for configuring the ShoreTel, Adtran and ITSP. Even though configuration requirements can vary from setup to setup, the information provided in these steps, along with the Planning and Installation Guide and documentation provided by Adtran and the ITSP should prove to be sufficient. However every design can vary and some may require more planning then 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 ShoreTel supports Music On Hold (MOH) over SIP trunks. The maximum number of music on hold (MOH) streams that a SIP-enabled switch can 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 is configured with SIP profile supporting media hairpining and the trunk is on a half-width switch. Silence detection on trunk-to-trunk transfers is not supported, it requires a physical trunk. The ShoreTel system does not initiate calls with a 30ms payload, all calls are initiated with a 20ms payload. Fax (and modem) redirection is supported with SIP trunks only if the carrier or ITSP supports T.38. There may be other feature limitations when using SIP Trunks. Please consult Chapter 18 of ShoreTel Administration Guide as well as ShoreTel Partner guide at following location wnloads/shoretel_13_partner_guide.pdf pg. 8

9 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. Note: ShoreTel basically just points its Individual SIP Trunks to the Adtran SBC. 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. pg. 9

10 Call Control Options The first step is to make sure that the Enable SIP Session Timer box is checked. 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. pg. 10

11 Note: Always Use Port 5004 for RTP will be grayed out by default if SIP servers, SIP trunks or SIP extensions are configured Unchecking the box for 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) is 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 pg. 11

12 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 ITSP SIP Trunking. See the ShoreTel Planning and Installation Guide for more information. The next settings to verify are the Intra-Site Calls and the Inter-Site Calls settings. For the Intra-Site Calls, verify that the desired audio bandwidth is selected from the CODEC LIST for calls within the system. The settings should then be confirmed for the desired audio bandwidth CODEC LIST for Inter-Site calls (calls between sites). Note: CODEC LIST selection here is just an example. Please see the Planning and Installation Guide for additional information on CODEC LIST selection. 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) Switch Settings - Allocating Ports The final general settings to input are the ShoreGear switch settings. These changes are modified by selecting Administration, then Switches 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 will be displayed. Within the Edit ShoreGear Switch screen, select the desired number of SIP Trunks from the ports available. pg. 12

13 ShoreGear Switch Settings Each port designated as a SIP Trunk enables the support for 5 individual trunks. Music On Hold Source need to be checked if MOH expected on SIP trunk System Settings Trunk Groups ShoreTel Trunk Groups support s only Static SIP endpoint Individual Trunks. 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. pg. 13

14 Administration Trunk Groups This selection brings up the Trunk Groups screen. 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 and click on the Go link from Add new trunk group at site:. The Edit SIP Trunk Group screen will appear. Select CenturyLink profile from the dropdown list. SIP Trunk Group Settings pg. 14

15 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 Qwest has been created. The Enable Digest Authentication field is not required when connecting to an ADTRAN SBC. The Enable SIP Info for G.711 DTMF Signaling box should not be enabled (checked). pg. 15

16 SIP profile CenturyLink System parameters in CenturyLink profile: Options Ping = 1 a periodic OPTIONS ping is implemented to detect if connectivity exists to the peer. If there is no response, or an unexpected error response, the trunk will be placed OOS until it responds appropriately again. Options Period = period between OPTIONS pings is configurable via OptionsPeriod profile setting. Hairpin = 1 - will hairpin media through the switch only when needed for particular features such as simultaneous ringing and external assignment, for example. See System Administration Guide for more details. Note: Video calls will NOT hairpin through the switch. Invoking a feature that requires hairpinning will drop video. SendMacIn911CallSetup = 1 - The ShoreTel SIP trunk transports the MAC address of a ShoreTel IP phone when emergency call was placed. AddG729AnnexB_NO= 1 - Absence of support for G.729 Annex B is flagged. EnableP-Asserted Identity = 1 - Support for the P-Asserted-Identity and Privacy headers to meet the user privacy requirements. RFC 3325 pg. 16

17 HistoryInfo= diversion - SIP Diversion is the ability to pass History Information on SIP trunks from a ShoreTel system to a SIP provider. This new enhancement provides a way to consistently and simply pass valid data to the SIP trunk provider to successfully route the call. Register = 0 - ShoreTel SIP trunks will not send a periodic REGISTER requests to CenturyLink. Please refer to the ShoreTel Administration Guide for further details on SIP profiles. The next item to change in the Edit SIP Trunks Group screen is to make the appropriate settings for the Inbound: fields. Inbound Within the Inbound: settings ensure the Number of Digits from CO is set to 10 and enable the DNIS or DID parameters as needed. It is no longer necessary to enable the Extension parameter for SIP Trunks, it defaults to disabled, but can be enabled (please refer to the ShoreTel Planning and Installation Guide for further information on configuration). pg. 17

18 Enable Tandem Trunking if you plan on transferring calls to external parties via the SIP trunk. Note: This section is configured no different than any normal Trunk Group Trunk Services On the Trunk Services: screen, make sure the appropriate services are checked or unchecked based on what CenturyLink supports and what features are needed from this Trunk Group. The parameter "Enable Original Caller Information" needs to be enabled. pg. 18

19 Make sure that "Remove leading 1 from 1+10D" parameter is enabled (checked) in the Trunk Digit Manipulation section. After these settings are made to the Edit SIP Trunk Group screen, press the Save button to complete the changes. This completes the settings needed to set up the trunk groups on the ShoreTel system. Logout of ShoreTel Director, you will then be presented with the ShoreTel Director login page. On your keyboard, hold down the <CTRL> and <Shift> keys and with the mouse pointer click on the U of the Username: field. This will enable the Support Entry mode of the ShoreTel Director, as referenced below: Log into ShoreTel Director with your normal administration user credentials. pg. 19

20 Navigate to the Edit SIP Trunk Group page, by selecting Administration followed by Trunks, then Trunk Groups, then in the Trunk Groups page, select the Trunk Group you created for CenturyLink (in above steps). This action brings up the Edit SIP Trunk Group page. Scroll down to the bottom of the page, in the Trunk Group Dialing Rules: parameter section, to the right of the Custom: parameter click on the Edit button. As noted below: This action brings up the Trunk Groups Dialing Rules Webpage Dialog as noted below: In the blank area of the Webpage Dialog enter ;10E and click on the Save button. Be sure to enter the exact syntax, this includes the semicolon, one, zero followed by a capital E. This syntax is case sensitive, verify that it matches the screen shot above. This completes the settings needed to set up the trunk groups on the ShoreTel system. pg. 20

21 3.2.5 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 Headquarters and the trunk group is SIP. Click on the Go button to bring up the Edit Trunk screen. Edit Trunks Screen for Individual Trunks pg. 21

22 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 Adtran SBC 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, press the Save button to input the changes. 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 Adtran SBC 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 4 Adtran ADTRAN, Inc. is a Huntsville, AL based telecommunications company. ADTRAN has a broad portfolio of products that enable Carrier and Enterprise networks. ADTRAN is the market leader for VoIP access and SIP Trunking Media Gateways. ADTRAN Routers and VoIP Gateways are designed to enable SIP and Unified communications over the Internet and to provide survivability in case of a WAN access failure. 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. Adtran 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. pg. 22

23 4.1 Adtran Product Information ADTRAN gateways with the SBC feature pack provide SIP interoperability demanded at the premises creating a service provider migration path to various services, from business trunking and hosted VoIP onto native SIP trunking. Designed to ease the need for extensive interoperability testing, ADTRAN gateways with the SBC Feature Pack provide the tools necessary to normalize, secure and troubleshoot the SIP to SIP communication between a carrier network and the customers SIP compliant equipment. The ADTRAN products that support the SBC feature pack are the NetVanta 3430 ( G4SBC), the NetVanta 3448 ( G4SBC), the NetVanta 4430 ( G3SBC), the Total Access 908e ( L1SBC and F2SBC) and the NetVanta 6250 ( F2SBC). For the most up to date information about ADTRAN SBC products, visit ADTRAN gateways with the SBC feature pack support SIP and RTP protocol normalization and media anchoring. The SBC feature pack allows for operation in a B2BUA mode and other advanced customization tools to allow integration with other vendors. One of those advanced customization tools included in ADTRAN gateways with the SBC feature pack is SIP Header Manipulation Rules. This feature allows the manipulation of both SIP headers and message bodies in SIP transmissions, based on configurable rules. These rules can be applied to both outbound and inbound messages, and can be used to match SIP headers, modify existing SIP headers or the body of SIP messages, add SIP headers, remove SIP headers, and store variable information. SIP header and message manipulation from ADTRAN solves interoperability issues present in disparate networks 4.2 ADTRAN SBC Product Configuration The following section will briefly describe the configuration of the ADTRAN products. Further configuration of the ADTRAN products can be found on the searchable ADTRAN support forums, Below are links directly to a few documents very useful for configuring and troubleshooting the ADTRAN SBC. ADTRAN SBC Sample Configuration Session Border Controllers in ADTRAN OS Configuring Media Anchoring in ADTRAN OS Manipulating SIP Headers and Messages in ADTRAN OS pg. 23

24 4.2.1 Network Topology This would be the typical installation of the Adtran SBC into a ShoreTel deployment using CTL sip Trunks. There are many different types of CTL networks and your network details may vary Login Adtran SBC can be managed by console port, http, https, telnet and SSH. Most initial configuration will via console port. Default user/pass/enable : admin/password/password Console port speed 9600, 8, n, Configuring the Adtran SBC With the CLI enter the Ethernet configuration needed for the private interface and the public ITSP interface. Below is an example: Console or Telnet to SBC: User Access Login Username: admin Password: password ADTRAN > en type this to enter enable mode Password: password ADTRAN# # shows that you are in enable mode ADTRAN# configure terminal to enter configuration mode ADTRAN(config)# below is the information on the Ethernet interfaces interface eth 0/1 description inside/private network ip address IP based on the Customer LAN no ip proxy-arp media-gateway ip primary no shutdown no lldp send-and-receive interface eth 0/2 description outside/public network ip address IP based on the CTL order no ip proxy-arp ip access-policy Public media-gateway ip primary no shutdown pg. 24

25 no lldp send-and-receive SBC basic voice configuration rules voice forward-mode local voice codec-list G729_first codec g729 codec g711ulaw voice codec-list G729_only codec g729 voice codec-list G711_only codec g711ulaw voice codec-list G711_first codec g711ulaw codec g729 voice trunk-list PBX_Trunks trunk T10 voice trunk-list Provider_Trunks trunk T01 trunk T02 trunk T03 trunk T04 ip rtp media-anchoring Add this policy when using ShoreTel 13.3 and g729 first codec list on the CTL and ShoreTel trunk hmr policy Shoretel_Inbound rule-set Remove hmr policy Shoretel_Trunk_Outbound rule-set Shoretel_Early_Media_Workaround 10 Add this hmr rule-set when using ShoreTel 13.3 and g729 first codec list on the CTL and ShoreTel trunk hmr rule-set Remove729 message-rule Modify_Body message-type request 10 modify body match-value "/(m=audio \d+ RTP\/AVP 0 )([\d ]*)18/" new-value /\1\2/ 10 modify body match-value /(\r\n)a=fmtp:18.*\r\n/ new-value /\1/ 20 hmr rule-set Shoretel_Early_Media_Workaround message-rule Convert_183_to_180 message-type response 10 pg. 25

26 modify header sip-status-line position first match-value "/183 Session Progress/" new-value "/180 Ringing/" SBC IP-PBX trunk configuration voice trunk T10 type sip description "SIP_to_Customer_PBX SG220" sip-server primary max-number-calls set this to match the ShoreTel Sip trunk on the ShoreGear hmr Shoretel_Trunk_Outbound out add this rule if using g729 first and using ShoreTel version 13.3 hmr Shoretel_Inbound in trust-domain codec-list G711_first both match codec list on sip trunk order grammar from host local transfer-mode network voice grouped-trunk CUSTOMER_SIP_PBX trunk T10 accept $ cost 0 permit list Provider_Trunks deny all other trunks deny all other ani SBC ITSP CTL trunk configuration Centurylink has two sip trunk end points Chicago and Houston that can each deliver session trunk and or usage trunks. Session trunks deliver a PRI/TIE like service that supports both inbound and outbound calls. Usage trunks deliver 8xx and RDID Remote DID that supports inbound only calls from the ITSP. Add voice trunk T01-T04 as required by sip trunk orders from CTL voice trunk T01 type sip description "Chicago_NBS session" sip-server primary IP provided by CTL on sip trunk order max-number-calls provided by the CTL sip trunk order trust-domain codec-list G711_first both provided by the CTL sip trunk order voice trunk T02 type sip description "Houston_NBS session" sip-server primary max-number-calls trust-domain codec-list G729_first both voice trunk T03 type sip description "Chicago_RDID _NBS" pg. 26

27 sip-server primary max-number-calls 10 trust-domain codec-list G729_first both voice trunk T04 type sip description "Houston_RDID_ NBS" sip-server primary max-number-calls 10 trust-domain codec-list G729_first both voice grouped-trunk CTL_NBS trunk T01 Add session trunk T01 or T02 based on CTL sip trunk order; do not add T03 or T04 accept NXX-NXX-XXXX cost 0 accept 1-NXX-NXX-XXXX cost 0 accept 011-$ cost 0 accept 411 cost 0 accept 611 cost 0 accept 911 cost 0 permit list PBX_Trunks deny all other trunks deny all other ani Adtran SBC Router configs ip firewall ip firewall stealth ip access-list standard SIP_ CTL remark IP subnets used for CTL Sip trunk add permit list provided by the CTL sip trunk order permit ip policy-class Public allow list SIP_CTL self ip route route assigned per CTL order pg. 27

28 4.3 Adtran Troubleshooting Call Flow Examples Incoming Call Incoming calls will always originate from the Service Provider and be addressed directly to the Adtran IP Address. The Adtran in turn will route the call to the ShoreGear switch Outgoing Call Outgoing calls will always originate from the ShoreTel Phones, and then the ShoreGear switch makes a call directly to the Adtran IP address. The Adtran in turn will route the call to the ITSP. pg. 28

29 4.3.2 ADTRAN Troubleshooting Tools Show and Debug Commands The ADTRAN has several show debugging commands that will print information on the console session. The show commands will print a snapshot of the activity on the SBC. The debug commands will print real time activity on the SBC. Some useful debugging commands are listed below. Use the show ip rtp media sessions command to display all of the anchored RTP flow associations and the number of relayed packets per association currently active in an anchored RTP flow. In addition, the TTL and the session type (digital signal processing (DSP), media-anchored, or transcoded) for the association is displayed. #show ip rtp media sessions Call ID Anchored Address Remote Address TTL Packets OvrrdTypeSession : : NoAudioDSP : : NoAudioDSP : : NoAudioDSP : : NoAudioDSP Use the debug voice switchboard command to display the decision making process of the SBC as it routes incoming and outgoing calls. This will show the trunk that the call was received from and the trunk that was selected to route the call to. #debug voice switchboard Use the debug sip stack messages command to display the SIP messages that are received by or sent from the ADTRAN SBC. This command will display the full SIP headers as well as the SDP body and will tell whether the ADTRAN SBC received or transmitted the message.. #debug sip stack messages Packet Capture The Packet Capture capability of the ADTRAN SBC allows for the capture and export of all traffic on any one or ALL interfaces simultaneously. Then export to pg. 29

30 your PC where it can be viewed in Wireshark or Ethereal. For more information about configuring packet capture, see the ADTRAN Support Forums Check Network Standard PING and Trace Route commands are available for simple network checks. 4.4 ADTRAN Sales & Technical Support Sales If you would like an ADTRAN sales representative to contact you, call You can also visit the ADTRAN website at or send an to Technical Support For post-sales technical support, call and have your serial number ready for the agent. For non-urgent issues, you may send an to and include the serial number in the . You may also open a support ticket at pg. 30

31 5 Document and Software Copyrights Copyright 2013 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. 6 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. 7 Disclaimer 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. 8 Company Information ShoreTel, Inc. 960 Stewart Drive Sunnyvale, California USA fax pg. 31