3. Configure Avaya Communication Manager
Contents
1. 3 Configure Avaya Communication Manager This section describes a procedure for setting up a SIP trunk between Avaya Communication Manager and Avaya SES which includes steps for setting up a list of IP code set an IP network region a signaling group and its interface Before a trunk can be configured it is necessary to verify if there is enough capacity to setup an additional trunk Also a procedure is described here to configure SIP telephones in Avaya Communication Manager Configuration in the following sections is only for the fields where a value needs to be entered or modified Default values are used for all other fields For further information related to configure Avaya Communication Manager refer to 1 and 2 These steps are performed from the Avaya Communication Manager System Access Terminal SAT interface Grandstream and other SIP telephones are configured as off PBX telephones in Avaya Communication Manager Avaya Communication Manager does not directly control an off PBX telephone but its features and calling privileges can be applied to it by associating a local on PBX telephone with the off PBX telephone Similarly a SIP telephone in Avaya SES is associated with an on PBX telephone on Avaya Communication Manager SIP Telephones register with the Avaya SES and use Avaya Communication Manager for call origination and termination services Throughout the rest of this document on PBX telephones associated with SIP2. CS can be same or different from SIP UserID p not displayed for security protection CA optional e g John Doe C No Yes C No Yes fo in minutes default 1 hour max 45 days 5060 default 5060 feo in seconds Between 1 3600 default is 20 UDP TCP I in audio M viaRTP RFC2833 M via SIP INFO C No amp Yes No C Yes No No C Yes C Yes choice 1 PCMU choice 2 PCMA a choice 3 G 723 1 choice 4 G 729A B choice 5 G 726 32 choice 6 iLBC F choice 7 G 722 wide band x choice 8 Gsm z Update Cancel Reboot AT Reviewed RRR m d y Solution amp Interoperability Test Lab Application Notes 2007 Avaya Inc All Rights Reserved 23 of 30 GrandstreamSIP doc 6 Interoperability Compliance Testing The focus of the interoperability compliance testing was primarily on verifying call establishment on the Grandstream telephones and operations such as dialing methods manual re dial and phone book hold mute transfer and conference Grandstream telephones interactions with SES Avaya Communication Manager and Avaya SIP H 323 and digital telephones were also verified 6 1 General Test Approach The general test approach was to place calls to and from the Grandstream GXP2020 and GXP1200 telephones and exercise basic telephone operations The main objectives were to verify that Grandstream telephones successfully register with Avaya SES
3. Avaya 6400 and 8400 Series Digital Telephones Avaya G650 Media Gateway C LAN 192 45 100 147 MEDPRO 192 45 103 148 VLAN 103 192 45 103 024 VLAH 100 192 45 100 0 24 LAN 3 VLAN 53 192 45 53 0 24 Avaya 4600 Series IP Telephones Grandstream GXP1200 IP Phone 192 45 53 90 Grandstream GXP2020 Enterprise IP Phone 192 45 53 80 Figure 1 Sample configuration AT Reviewed RRR m d y Solution amp Interoperability Test Lab Application Notes 2007 Avaya Inc All Rights Reserved VLAN 52 192 45 52 0 24 Avaya 8710 Media Servers Avaya SIP Enablement Services Server 192 45 52 160 3 of 30 GrandstreamSIP doc 2 Equipment and Software Validated The following equipment and software firmware were used for the sample configuration provided Equipment Software Firmware Avaya S8710 Media Server Avaya Communication Manager 4 0 1 R014x 00 1 731 2 Avaya G650 Media Gateway TN2312BP IP Server Interface HW12 FW040 TN799DP C LAN Interface HW01 FW024 TN2302AP IP Media Processor HW20 FW117 Avaya SIP Enablement Services Server SES 4 0 SES 4 0 0 0 033 6 Avaya 4600 Series IP Telephones 2 2 3 4610SW SIP 2 3 4602SW H 323 2 6 4610SW H 323 2 5 4625SW H 323 Avaya 6400 and 8400 Series Digital Telephones Avaya Analog Telephone 5 Grandstream Networks GXP2020 Telephone 1 1 5 15 Grandstream Networks GXP1200 Telephone 1 1 5 15
4. Grandstream telephones successfully establish calls with Avaya SIP H 323 and digital telephones attached to Avaya SES or Avaya Communication Manager Grandstream telephones successfully establish calls with PSTN telephones through Avaya Communication Manager Grandstream telephones successfully handle concurrent calls Grandstream telephones successfully negotiate the right codec Grandstream telephones successfully shuffle for VoIP calls Grandstream telephones successfully transmit DTMF during a call Grandstream telephones successfully hold and transfer a call Grandstream telephones establish a three party conference call and display calling party number Grandstream telephones successfully tags layer 2 802 1p and layer 3 DiffServ QoS packets For serviceability testing failures such as cable pulls and hardware resets were applied For performance testing a conference call involving two Grandstream telephones and two Avaya telephones was formed as follows A call was established between an Avaya telephone and a Grandstream telephone The Grandstream telephone then used its second call appearance to establish a call with another Grandstream telephone and bridged the two lines together forming a 3 party conference The second Grandstream telephone then used its second call appearance to establish a call with another Avaya telephone and bridged its two lines together effectively forming a 4 party conference AT Reviewed
5. 54007 8710 nodeid78 onfiguration vianagement 10 Click Continue at the bottom of the right panel Wrote Wrote Wrote Wrote Wrote Wrote Wrote 2 2 1 1 I 2 2 m Continue Wrote 1 domain access record Wrote 1 proxy configuration record Wrote 1 proxy configuration record Wrote 1 proxy configuration record wrote 8 system parameters records Deleted 3 subscriber records Update 54007 on home node 192 45 52 160 contact set records public address records contact record identity record presence list record access control list records extension records AT Reviewed RRR m d y Solution amp Interoperability Test Lab Application Notes 2007 Avaya Inc All Rights Reserved 19 of 30 GrandstreamSIP doc 5 Configure Grandstream Telephones This section describes the steps for configuring the Grandstream telephones Grandstream GXP2020 and GXP1200 have similar configuration steps except Grandstream GXP2020 supports up to six separate SIP accounts whereas GXP1200 supports up to two separate SIP Accounts This section assumes that the Grandstream telephone s IP address is already configured Configuration steps described in this section apply only to the fields where a value needs to be modified or entered Default values are used for all other fields Screens shots shown here are for GXP2020 but GXP1200 has similar screens For further information on Grandstream telephones refe
6. SIP Trunk SIP Trunk Link Type CTCP TLS SIP Trunk IP Address fi92 45 100 147 Media Server Media Server Admin Address 65 see Help Media Server Admin Login Media Server Admin Password z s ee ee Media Server Admin Password Export Import to Provision Confirm Update Adjunct Systems Fields marked are required AT Reviewed Solution amp Interoperability Test Lab Application Notes 15 of 30 RRR m d y 2007 Avaya Inc All Rights Reserved GrandstreamSIP doc Step Description 5 AT Reviewed RRR m d y In the left pane of the SIP Server Management page expand Users and click Add ianagement IM Logs Trace Logger Export Import to Provision ma Top Manage Users Manage Conferencing Manage Media Server Extensions Manage Emergency Contacts Manage Hosts Manage Media Servers Manage Adjunct Systems Manage Services Server Configuration Certificate Management IM Logs Trace Logger Export Import to Pro ision Solution amp Interoperability Test Lab Application Notes 2007 Avaya Inc All Rights Reserved Add and delete Users Add and delete Conference Extensions Add and delete Media Server Extensions Add and delete Emergency Contacts Add and delete Hosts Add and delete Media Servers Add and delete Adjunct Systems Start and stop server processes on this host Edit Properties of the system Manage Web C
7. Solution amp Interoperability Test Lab Application Notes 24 of 30 RRR m d y 2007 Avaya Inc All Rights Reserved GrandstreamSIP doc 6 2 Test Results The test objectives of Section 6 1 were verified For serviceability testing the Grandstream telephones operated properly after recovering from failures such as cable disconnects and resets of the Grandstream telephones the Avaya SES server and Avaya Communication Manager For performance testing the conference call was successfully maintained for approximately two hours Grandstream telephones successfully shuffled to communicate directly with the other telephones Grandstream telephones successfully negotiated the codec to be used and properly tagged layer 2 and layer 3 QoS packets The following observations were made during testing e Grandstream telephones cannot mute all parties if it initiates the conference Only the last party added is muted e Grandstream telephones only support UDP as SIP transport Grandstream Networks will address and attempt to resolve the above observations in future firmware releases Contact Grandstream Networks www grandstream com for further updates AT Reviewed Solution amp Interoperability Test Lab Application Notes 25 of 30 RRR m d y 2007 Avaya Inc All Rights Reserved GrandstreamSIP doc 7 Verification Steps The following steps may be used to verify the configuration Verify that the Grandstream telephones successfully register with
8. provisioned in Section 3 2 e Inter region IP IP Direct Audio Set to yes to allow direct IP to IP audio connectivity between endpoints registered to Avaya Communication Manager or Avaya SES in different IP network regions change ip network region 2 Page 1 of 19 IP NETWORK REGION Region 2 Location Authoritative Domain devconnect com Name MEDIA PARAMETERS Intra region IP IP Direct Audio yes Codec Set 2 Inter region IP IP Direct Audio yes UDP Port Min 2048 IP Audio Hairpinning y UDP Port Max 65535 DIFFSERV TOS PARAMETERS RTCP Reporting Enabled y Call Control PHB Value 46 RTCP MONITOR SERVER PARAMETERS Audio PHB Value 46 Use Default Server Parameters y Video PHB Value 26 802 1P Q PARAMETERS Cell Cemal B02 bo Priority G Auco S02 iio Priority 6 Video 802 1p Priority 5 AUDIO RESOURCE RESERVATION PARAMETERS H 323 IP ENDPOINTS RSVP Enabled n H 323 Link Bounce Recovery y Idle Traffic Interval sec 20 Keep Alive Interval sec 5 Keep Alive Count 5 AT Reviewed Solution amp Interoperability Test Lab Application Notes 7 of 30 RRR m d y 2007 Avaya Inc All Rights Reserved GrandstreamSIP doc Step Description 2 Proceed to Page 3 of IP network region configuration and enable inter region connectivity between regions as per below For this compliance testing codec set was set to the IP codec set configured in Section 3 2 Page 3 of 19 Inter Network Region Connection Management src dst code
9. telephones in such a manner will be referred to as Outboard Proxy SIP OPS stations AT Reviewed Solution amp Interoperability Test Lab Application Notes 4 of 30 RRR m d y 2007 Avaya Inc All Rights Reserved GrandstreamSIP doc 3 1 Capacity Verification Step Description 1 Enter the display system parameters customer options command Verify that there are sufficient Maximum Off PBX Telephones OPS licenses If not contact an authorized Avaya account representative to obtain additional licenses display system parameters customer options Page i oie 10 OPTIONAL FEATURES G3 Version V13 HOC ate oman RFA System ID SID 1 Platform 8 RFA Module ID MID 1 USED Platform Maximum Ports 44000 908 Maximum Stations 36000 410 Maximum XMOBILE Stations 0 0 Maximum Off PBX Telephones WESOOs 5 0 Maximum Off PBX Telephones OPS 200 50 Maximum Off PBX Telephones SCEAN EEO 0 2 Proceed to Page 2 of OPTIONAL FEATURES form Verify that the number of SIP trunks supported by the system is sufficient for the number of SIP trunks needed If not contact an authorized Avaya account representative to obtain additional licenses Note Each SIP call between two SIP endpoints whether internal or external requires two SIP trunks for the duration of the call The license file installed on the system controls the maximum permitted display system param
10. the Avaya SES server by following the Users gt Registered Users links on the SES Administration Web Interface Place calls to and from the Grandstream telephone and verify that the calls are successfully established with two way talk path From the Avaya Communication Manager System Access Terminal SAT interface perform the following steps to verify Audio codec used between two telephones Shuffling between two telephones Step Description 1 Enter status trunk lt t gt command where tis the SIP trunk configured in Section 3 6 Note down the Member with Service State set to in service active In this example 0010 002 and 0010 006 are active and either member can be used to verify whether calls shuffled and which codec was used Status trunk 10 TRUNK GROUP STATUS Member Port Service State Meee Commecicecl POTES Busy 0010 001 T00046 in service idl no 0010 002 T00047 in service active no T0051 0010 003 TO00048 in service idl no 0010 004 T00049 in service idl no 0010 005 TO0050 in service idl no 0010 006 T00051 in service active no T0047 0010 007 T0O0052 in service idl no 0010 008 TO0O053 in service idl no 0010 009 T00054 in service idl no 0010 010 T0O0055 in service idl no AT Reviewed Solution amp Interoperability Test Lab Application Notes 26 of 30 RRR m d y 2007 Avaya Inc All Rights Reserved GrandstreamSIP doc Step Description 2 Ent
11. 0 2007 Avaya Inc All Rights Reserved GrandstreamSIP doc Step Description 7 At the Add Media Server Extension page configure the following Top Extension Set to Phone Number field value configured in Section 3 7 Step 3 Media Server Set to the media server where this OPS station is configured Click Add and then click Continue on the next page not shown Note Media Server was previously configured on SES Help Exit all Add Media Server Extension Users List Add Media Server extension for user 54007 Add Extension 54007 Search Edit Media Server 38710 gt Delete Fields marked are required Password Default Profile Registered Users Conferences Media Server Extensions Emergency Contacts Hosts Media Servers Adjunct Systems o oOo w oO a Configuration Certificate Management IM Logs Trace Logger Export Import to Provision Update 8 Repeat Steps 5 7 as necessary to configure additional Grandstream telephones AT Reviewed Solution amp Interoperability Test Lab Application Notes RRR m d y 2007 Avaya Inc All Rights Reserved 18 of 30 GrandstreamSIP doc Step Description 9 Click Update at the bottom of the left panel to save the configuration completed in the above steps List Media Server Extensions Aas Media Server extensions for user 54007 Move Ext Free EditUser Delete 54007
12. 2020 and GXP1200 are SIP based VoIP telephones Grandstream GXP2020 telephone is typically used in an enterprise or small business environment and Grandstream GXP1200 telephone is used by residential or Small Office and Home Office SoHo users During compliance testing Grandstream telephones successfully registered with Avaya SES placed and received calls to and from SIP and non SIP telephones and executed other telephony features like three way conference transfers holds etc Grandstream telephones can bridge calls on a single line to establish a three party conference Grandstream GXP2020 supports up to six and GXP1200 is a single line telephone Grandstream telephones support IM and Presence but no testing was done because of incompatibility with Avaya s implementation Figure 1 illustrates a sample configuration consisting of a pair of Avaya S8710 Media Servers an Avaya G650 Media Gateway an Avaya SIP Enablement Services SES server and the Grandstream telephones Avaya Communication Manager is installed on the 8710 Media Servers The solution described herein is also extensible to other Avaya Media Servers and Media Gateways For completeness Avaya 4600 Series SIP IP Telephones Avaya 4600 Series H 323 IP Telephones and Avaya 6400 and 8400 Series Digital Telephones are included in Figure 1 to demonstrate calls between the SIP based Grandstream telephones and Avaya SIP H 323 and digital telephones The analog PSTN telephone is also in
13. 5 6 System Up Time 0 day s 21 hour s 51 minute s Registered Account 1 Yes Account 2 Yes Account 3 No Account 4 No Account 5 No Account 6 No disabled AT Reviewed RRR m d y Solution amp Interoperability Test Lab Application Notes 2007 Avaya Inc All Rights Reserved 28 of 30 GrandstreamSIP doc 8 Support For technical support on Grandstream Networks telephones consult the support pages at http www grandstream com contact_us html or contact Grandstream Networks technical support at e Telephone 1 617 566 9300 e E mail support grandstream com 9 Conclusion These Application Notes describe a solution comprised of Avaya Communication Manager 4 0 1 Avaya SES 4 0 and Grandstream Networks SIP telephones Grandstream GXP2020 and GXP1200 are SIP based VoIP telephones Grandstream GXP2020 telephone is typically used in an enterprise or small business environment and Grandstream GXP1200 telephone is used by residential or SoHo users During compliance testing Grandstream telephones successfully registered with Avaya SES placed and received calls to and from SIP and non SIP telephones and executed other telephony features like three way conference transfers hold etc The objective of Section 6 1 were met with some exceptions noted in Section 6 2 10 Additional References Product documentation for Avaya products may be found at http support avaya com 1 Administrator Guide for Avaya Comm
14. ATION OPTIONS LOSSA GaOUlp EZ Personalized Ringing Pattern 1 Data Module n Message Lamp Ext 54007 Speakerphone 2 way Mute Button Enabled y Display Language english Media Complex Ext IP SoftPhone n 2 Proceed to Page 3 of the STATION form and add the required number of call appr entries in BUTTON ASSIGNMENT field The number of call appearances should match the Call Limit field value in Step 4 add station 54007 Page 3 Ole 3 STATION SITE DATA Room Headset n Jack Speaker n Cable Mounting d HOOR Cord Length 0 Building Set Color ABBREVIATED DIALING IEG EAL List2 imal ie SB BUTTON ASSIGNMENTS 1 call appr 5g 2 call appr G8 33 Caill aljsjare he 4 38 AT Reviewed Solution amp Interoperability Test Lab Application Notes 11 of 30 RRR m d y 2007 Avaya Inc All Rights Reserved GrandstreamSIP doc Step Description 2 Enter the add off pbx telephone station mapping command and configure the following e Station Extension Set the extension of the OPS station as configured above e Application Set to OPS e Phone Number Enter the number that the Grandstream telephone will use for registration and call termination In the example below the Phone Number is the same as the Station Extension but is not required to be the same e Trunk Selection Set to the trunk group number configured in Section 3 6 add off pbx telepho
15. AVAYA Avaya Solution amp Interoperability Test Lab Application Notes for the Grandstream Networks SIP Telephones with Avaya Communication Manager 4 0 1 and Avaya SIP Enablement Services 4 0 Issue 0 1 Abstract These Application Notes describe a solution comprised of Avaya Communication Manager 4 0 1 Avaya SIP Enablement Services 4 0 and Grandstream Networks SIP telephones Grandstream GXP2020 and GXP1200 are SIP based VoIP telephones Grandstream GXP2020 telephone is typically used in an enterprise or small business environment and Grandstream GXP1200 telephone is used by residential or Small Office and Home Office users During compliance testing Grandstream telephones successfully registered with Avaya SES placed and received calls to and from SIP and non SIP telephones and executed other telephony features like three way conference transfers holds etc Information in these Application Notes has been obtained through compliance testing and additional technical discussions Testing was conducted via the DevConnect Program at the Avaya Solution and Interoperability Test Lab AT Reviewed Solution amp Interoperability Test Lab Application Notes 1 of 30 RRR m d y 2007 Avaya Inc All Rights Reserved GrandstreamSIP doc 1 Introduction These Application Notes describe a solution comprised of Avaya Communication Manager 4 0 1 Avaya SIP Enablement Services SES 4 0 and Grandstream Networks SIP telephones Grandstream GXP
16. atch the SIP Domain value in Section 4 Step 2 add signaling group 10 Page OS SIGNALING GROUP Group Number 10 Group Type sip Transport Method tls Near end Node Name CLAN 1A06 Far end Node Name SES Near end Listen Port 5061 iaia Sinvel Lasem Dora 5061 Far end Network Region 2 Far end Domain devconnect com Bypass If IP Threshold Exceeded n DTMF over IP rtp payload Direct IP IP Audio Connections y IP Audio Hairpinning n Session Establishment Timer min 120 AT Reviewed RRR m d y Solution amp Interoperability Test Lab Application Notes 9 of 30 2007 Avaya Inc All Rights Reserved GrandstreamSIP doc 3 6 SIP Trunking This section describes the steps for administering a trunk group in Avaya Communication Manager for communication between Avaya Communication Manager and Avaya SES Step Description 1 Issue the command add trunk group lt t gt where tis an unallocated trunk group and configure the following 3 5 the maximum permitted add trunk group 10 Group Number 10 Group Name Direction two way Dial Access n Queue Length 0 Service Type ti SIP SES DevConl TRUNK GROUP Group Type Cor Outgoing Display n Auth Code n sip e Group Type Set to the Group Type field value configured in Section 3 5 e TAC Trunk Access Code Set to any available trunk access code e Signaling Group Set to the Group Number field value con
17. c direct Total Video Dyn rgn rgn set WAN WAN BW limits WAN BW limits Intervening regions CAC IGAR 2 1 2 y NoLimit n 2 OANA UBWNDN ial 12 15 14 LS NNNNNNNNNNNNDND NN 3 4 IP Node Names This section describes the steps for setting IP node name for Avaya SES in Avaya Communication Manager Step Description 1 Enter the change node names ip command and add a node name for Avaya SES along with its IP address change node names ip page L of i IP NODE NAMES Name IP Address CLAN 1A06 R92 459 200 TAT MEDPRO 1A13 192 45 4 103 148 SES 192 45 52 160 AT Reviewed Solution amp Interoperability Test Lab Application Notes 8 of 30 RRR m d y 2007 Avaya Inc All Rights Reserved GrandstreamSIP doc 3 5 SIP Signaling This section describes the steps for administering a signaling group in Avaya Communication Manager for communication between Avaya Communication Manager and Avaya SIP Enablement Services Step Description 1 Issue the command add signaling group lt s gt where s is an available signaling group and configure the following Group Type Set to sip Transport Method Set to tls Near end Node Name Set to CLAN name as displayed in Section 3 4 Far end Node Name Set to Avaya SES name configured in Section 3 4 Far end Network Region Set to the region configured in Section 3 3 Far end Domain Set to the devconnect com This should m
18. cluded to demonstrate calls routed by Avaya Communication Manager between the Grandstream telephones and the PSTN The Grandstream telephone originates a call by sending a call request SIP INVITE message to the Avaya SES server The Avaya SES server routes the call over a SIP trunk to Avaya Communication Manager for origination services If the call is destined for another local SIP telephone then Avaya Communication Manager routes the call back over the SIP trunk to Avaya SES server for delivery to destination SIP telephone Otherwise Avaya Communication Manager routes the call to the PSTN a local Avaya H 323 digital or analog telephone an adjunct a vector a hunt group etc depending on the destination number For a call arriving at Avaya Communication Manager that is destined for the Grandstream telephone Avaya Communication Manager routes the call over the SIP trunk to the Avaya SES server for delivery to Grandstream telephone These application notes assume that Avaya Communication Manager and Avaya SES are already installed and basic configuration steps have been performed Only steps relevant to this compliance test will be described in this document For further details on configuration steps not covered in this document consult 3 and 4 AT Reviewed Solution amp Interoperability Test Lab Application Notes 2 of 30 RRR m d y 2007 Avaya Inc All Rights Reserved GrandstreamSIP doc Simulated PSTN D PSTN Telephone
19. codec sets may be used within and between network regions For the to none as encryption is currently not supported for SIP telephony compliance testing G 711MU and G 729AB were used and Media Encryption was set IP Codec Set CodeeeSct cm Audio Silence Frames Packet Codec Suppression Per Pkt Size ms G 711MU n 2 20 G 729AB n 2 20 sa copy Gn SN CO ON Media Encryption none IS change ip codec set 2 Page Lor 2 AT Reviewed RRR m d y Solution amp Interoperability Test Lab Application Notes 2007 Avaya Inc All Rights Reserved 6 of 30 GrandstreamSIP doc 3 3 IP Network Region This section describes the steps for administering an IP network region in Avaya Communication Manager for communication between Avaya Communication Manager and Avaya SES Step Description 1 Enter the change ip network region lt n gt command where n is a number between 1 and 250 inclusive and configure the following e Authoritative Domain Set to the devconnect com This should match the SIP Domain value in Section 4 Step 2 e Intra region IP IP Direct Audio Set to yes to allow direct IP to IP audio connectivity between endpoints registered to Avaya Communication Manager or Avaya SES in the same IP network region e Codec Set Set the codec set number as
20. ep 2 Outbound Proxy Set to the Avaya SES server IP address SIP User ID Set to the User Id field value configured in Section 4 Step 6 Authenticate ID Set to the User Id field value configured in Section 4 Step 6 Authenticate Password Set to the Password field value configured in Section 4 Step 6 Name Enter any descriptive name SIP Transport Set to UDP Send DTMF set to via RTP Turn off speaker on remote disconnect Set the value to Yes Special Vocoder This should have at least one of the codecs configured in Section 3 2 Click Update Repeat this step to configure additional accounts For GXP2020 up to six accounts can be configured and for GXP1200 up to two accounts can be configured CNo Yes 54007 e g MyCompany devconnectcom e g sip mycompany com or IP address Account Active Account Name SIP Server Outbound Proxy SIP User ID Authenticate ID Authenticate Password Name SIP Registration Unregister On Reboot Register Expiration local SIP port SIP Registration Failure Retry Wait Time SIP Transport Send DTME Turn aff speaker on remote disconnect Check SIP User ID for incoming INVITE Refer To Use Target Contact Disable Multiple Media Attribute in SDP Preferred Vocoder in listed order hazass2teo e g proxy myprowider com or IP address 54007 the user part of an SIP address payo ti i i Cs
21. er status trunk lt m gt where m is the member in active state as noted in the previous step for verification of codec used and shuffling status e Codec The codec used for Audio is G 711MU in this example e Shuffling If the Near end IP Addr and Far end IP Addr for Audio are using the same port and the Audio Connection Type is ip direct it signifies that shuffling was successful In this example shuffling was successful status trunk 10 2 Page 1 of 2 TRUNK STATUS Trunk Group Member 0010 002 Service State in service active Borca TOOL Maintenance Busy No Signalling Group ID Conmmeetec Poresi T00S1 IPOE NGar anc IP Agee g Bore parendi IUP Acc 0 Pore Steinene OLAOGIY A92 A50 100 147 2 5061 192 45 52 160 3 SGI G 711MU Audio 192 45 53 101 34008 192 45 53 102 34008 Video vice COC oer Authentication Type None Audio Connection Type ip direct AT Reviewed Solution amp Interoperability Test Lab Application Notes 27 of 30 RRR m d y 2007 Avaya Inc All Rights Reserved GrandstreamSIP doc Step Description 3 Select the STATUS tab at the Grandstream Device Configuration screen and verify the following e Verify the IP Address is correct e Verify the Software Version is correct e Verify the Accounts configured in Section 5 Step 3 are registered with Avaya SES MAC Address 00 0B 82 12 12 42 IP Address 192 45 53 80 Product Model GXP2020 Software Version Program 1 1 5 15 Bootloader 1 1
22. ertificate Download IM Logs Manage SIP Trace Logs Export and import data using Provision on this host 16 of 30 GrandstreamSIP doc Step Description 6 At the Add User page configure the following AT Reviewed RRR m d y Pa Tal ge Default Profile Registered Users Conferences Media Server Extensions Emergency Contacts Hosts Media Servers Adjunct Systems G Server Configuration Certificate Management IM Logs Trace Logger Export Import to Provision e Primary Handle Enter the phone number of the Grandstream telephone This number was configured in Section 3 7 Step 3 User ID Set to any descriptive name Password and Confirm Password Specify a password that the Grandstream telephone will use to register with Avaya SES e Host Select the IP address or Fully Qualified Domain Name FQDN of the Avaya SES server First Name and Last Name Enter descriptive names Check the Add Media Server Extension checkbox e Click Add when finished and then click Continue on the next page not shown m Add User Primary Handle s4007 User ID s4007 oo Password Peres Confirm Password Peres Host 192 45 52 160 First Name Grandstream Last Name exez000 Address 1 MT Address 2 PM Office Po City PI State PI Country PI Zip C ea Fields marked are required Solution amp Interoperability Test Lab Application Notes 17 of 3
23. eters customer options Page 2 ie 10 OPTIONAL FEATURES Te PORT CAPAC ELITES USED Maximum Administered H 323 Trunks 200 148 Maximum Concurrently Registered IP Stations 1000 2 Maximum Administered Remote Office Trunks 0 0 Maximum Concurrently Registered Remote Office Stations 0 0 Maximum Concurrently Registered IP eCons 0 0 Max Concur Registered Unauthenticated H 323 Stations 0 0 Maximum Video Capable H 323 Stations 0 0 Maximum Video Capable IP Softphones 0 0 Maximum Administered SIP Trunks 200 153 Maximum Number of DS1 Boards with Echo Cancellation 0 0 Maximum TN2501 VAL Boards 1 ib Maximum G250 G350 G700 VAL Sources 0 0 Maximum TN2602 Boards with 80 VoIP Channels 2 0 Maximum TN2602 Boards with 320 VoIP Channels 2 il Maximum Number of Expanded Meet me Conference Ports 0 0 NOTE You must logoff amp login to effect the permission changes AT Reviewed Solution amp Interoperability Test Lab Application Notes 5 of 30 RRR m d y 2007 Avaya Inc All Rights Reserved GrandstreamSIP doc 3 2 IP Codec Set This section describes the steps for administering codec set in Avaya Communication Manager This codec set is used in the IP network region for communications between Avaya Communication Manager and Avaya SES Step Description 1 Enter the change ip codec set lt c gt command where c is a number between 1 and 7 inclusive IP codec sets are used in Section 3 3 for configuring IP network region to specify which
24. figured in Section e Number of Members Allowed values are between 0 and 255 Set to a value large enough to accommodate the number of SIP telephone extensions being used e Group Name Enter any descriptive name Note Each SIP call between two SIP endpoints whether internal or external requires two SIP trunks for the duration of the call The license file installed on the system controls Page i ge Al CDR Reports y TUM amp AL TAC 110 Night Service Signaling Group 10 Number of Members 150 AT Reviewed RRR m d y Solution amp Interoperability Test Lab Application Notes 2007 Avaya Inc All Rights Reserved 10 of 30 GrandstreamSIP doc 3 7 SIP Stations This section describes the steps for administering OPS stations in Avaya Communication Manager and associating the OPS station extensions with the telephone numbers of the Grandstream telephones Step Description 1 Enter the add station lt s gt command where s is an available extension in the dial plan to administer an OPS station On Page of the station form configure the following fields e Type Set to 6408D e Port Set to X e Name Enter any descriptive name add station 54007 Page i OE 4 STATION Extension 54007 Lock Messages n Recs Type 6408D Security Code TNA dl Port X Coverage Path 1 COR SRI Name GXP2020 Coverage Path 2 COSL keint tO Siecic aos ST
25. l IP 192 45 52 160 Local Name SES DevCon1 devconnect cam Logical IP 192 45 52 160 Logical Name SES DevCon1 devconnect com Gateway IP Address 192 45 52 1 AT Reviewed Solution amp Interoperability Test Lab Application Notes 13 of 30 RRR m d y 2007 Avaya Inc All Rights Reserved GrandstreamSIP doc Step Description 3 To enable secure SIP trunking between Avaya SES and Avaya Communication Manager add a media server corresponding to Avaya Communication Manager from the SIP Server Management page e Click the sign to expand the options under Media Servers e Click Add Extensions Cont Manage Media Server Interfaces List Media Servers Add Media Server List all media server interfaces Add a media server interface AT Reviewed RRR m d y Solution amp Interoperability Test Lab Application Notes 14 of 30 2007 Avaya Inc All Rights Reserved GrandstreamSIP doc Step Description 4 At the Add Media Server Interface page provision SIP Trunk parameters as follows for connectivity to Avaya Communications Manager e SIP Trunk Link Type Set to the Transport Method field value in Section 3 5 e SIP Trunk IP Address Set to the CLAN IP address as displayed in Section 3 4 e Click Add when finished and then click Continue on the confirmation page not shown all Add Media Server Interface Media Server Interface sevi0 SSCS Name Host 192 45 52 160
26. ne station mapping Page il eye 2 STATIONS WITH OFF PBX TELEPHONE INTEGRATION Station Application Dial Phone Number Trunk Configuration Extension Prefix Selection Set 54007 OPS 54007 10 1 4 Proceed to Page 2 of station mapping form and verify that the Call Limit field value matches the number of call appearances configured in Step 2 add off pbx telephone station mapping 54008 Page 2 Gir 2 STATIONS WITH OFF PBX TELEPHONE INTEGRATION Station Call Mapping Calls Allowed Bridged Extension Limit Mode Allowed Calls 54008 2 both all both 1 3 Repeat Steps 1 and 2 as necessary to administer additional OPS stations and associations for Grandstream telephones AT Reviewed Solution amp Interoperability Test Lab Application Notes 12 of 30 RRR m d y 2007 Avaya Inc All Rights Reserved GrandstreamSIP doc 4 Configure Avaya SIP Enablement Services This section describes the steps for creating SIP trunk between Avaya SES and Avaya Communication Manager Also SIP user accounts are configured in Avaya SES and associated with an Avaya Communication Manager OPS station extension The Grandstream telephone will register with Avaya SES using the SIP user accounts For further information related to configure Avaya SES refer to 5 and 6 Configuration in the following steps is only for the fields where a value needs to be entered or modified Default values are used for all other fields Step Description 1 Open a web browser enter ht
27. ons or comments pertaining to these Application Notes along with the full title name and filename located in the lower right corner directly to the Avaya DevConnect Program at devconnect avaya com AT Reviewed Solution amp Interoperability Test Lab Application Notes 30 of 30 RRR m d y 2007 Avaya Inc All Rights Reserved GrandstreamSIP doc
28. r to 5 and 6 Note Due to the page size only the most relevant fields have been included in the screen shots Step Description 1 Open a web browser and enter http a b c d for the URL where a b c d is the IP address of the Grandstream telephone Enter the password and click Login to proceed to the next screen Password Login AT Reviewed Solution amp Interoperability Test Lab Application Notes 20 of 30 RRR m d y 2007 Avaya Inc All Rights Reserved GrandstreamSIP doc Step Description 2 Select the BASIC SETTINGS tab and check the statically configured as option to configure as follows e IP Address Set the IP address e Subnet Mask Set the subnet mask e Default Router Set the default router e Click Update to modify the values End User Password purposely not displayed for security protection IP Address dynamically assigned via DHCP default or PPPoE will attempt PPPoE if DHCP fails and following is non blank PPPoE account ID PPPoE password Host name Option 12 Domain name OO Option 15 Vendor Class ID Grandstream GxP2020 Option 60 Grandstream GxP2020 Preferred DNS server fo Jo Jo Jo statically configured as IP Address fi 92 Jas E Jeo Subnet Mask 255 Jess 255 Default Router h 92 jas E M DNS Server 1 fo Jo J J DNS Server 2 fo fo fo Update Cancel Reboot AT Reviewed Solution amp Interoperabilit
29. tp lt IP address of Avaya SES server gt admin for the URL and log in with the appropriate credentials Click on the Launch Administration Web Interface link upon successful login 2 On the SIP Server Management page e Click the sign to expand the options under Server Configuration e Click System Properties e Verify the SIP Domain matches the Far end Domain field value configured for the signaling group on Avaya Communication Manager in Section 3 5 Help Exit Top a Wears View System Properties Conferences K SES_Yersion SES 4 0 0 0 033 6 Media Server Extensions 2 a System Configuration simplex Emergency Contacts Host Type home edge Hosts Media Servers SIP Domain devconnect com Note that the DNS domain is devconnect com If you are unsure about this field most often the SIP domain should be the root level DNS domain For example for a DNS domain of eastcoast example com the SIP EN ae ee domain would likely be configured to example com This SIRNA figuration allows SIP calls and instant messages to users with handles system Properties of the format handle example com Servic Admin Accounts Licens License Host localhost IM Log Settings onfiguration Management System Access Login Management System eS Access Password te Management Logger HAEE DiffServ TOS Parameters Call Control PHB value jas 802 1 Parameters Priority value Je Network Properties Loca
30. unication Manager Issue 2 1 May 2006 Document Number 03 300509 2 Administration for Network Connectivity for Avaya Communication Manager Issue 11 February 2006 Document Number 555 233 504 3 SIP Support in Release 3 1 of Avaya Communication Manager Issue 6 February 2006 Document Number 555 245 206 4 Installing and Administering SIP Enablement Services R4 0 Issue 2 0 August 2006 Document Number 03 600768 Product documentation for Grandstream Networks products may be found at http www grandstream com 5 Grandstream GXP2020 user manual GXP2020UsersManual pdf 6 Grandstream GXP1200 user manual GXP1200UserManual pdf AT Reviewed Solution amp Interoperability Test Lab Application Notes 29 of 30 RRR m d y 2007 Avaya Inc All Rights Reserved GrandstreamSIP doc 2007 Avaya Inc All Rights Reserved Avaya and the Avaya Logo are trademarks of Avaya Inc All trademarks identified by and are registered trademarks or trademarks respectively of Avaya Inc All other trademarks are the property of their respective owners The information provided in these Application Notes is subject to change without notice The configurations technical data and recommendations provided in these Application Notes are believed to be accurate and dependable but are presented without express or implied warranty Users are responsible for their application of any products specified in these Application Notes Please e mail any questi
31. y Test Lab Application Notes 21 of 30 RRR m d y 2007 Avaya Inc All Rights Reserved GrandstreamSIP doc Step Description 3 Select the ADVANCED SETTINGS tab and configure as follows e Layer 3 QoS Set to the desired value between 0 and 63 For compliance testing a value of 48 was used e 802 1p priority value Set to the desired value between 0 and 7 For compliance testing a value of 6 was used e Click Update to modify the values Admin Password purposely not displayed for security protection Silence Suppression No Yes Voice Frames per TX le up to 10 20 32 64 for G711 G726 G723 other codecs respectively Layer 3 QoS jas DiffServ or Precedence value Layer 2 QoS 802 1QNLAN Tag 53 802 1p priority value fe 0 7 No Key Entry Timeout a in seconds default is 4 seconds Use as Dial Key C No Yes local RTP port 5004 1024 65535 default 5004 Use random port No Yes keep alive interval fo in seconds default 20 seconds DTMF Payload Type hi 01 Update Cancel Reboot AT Reviewed Solution amp Interoperability Test Lab Application Notes 22 of 30 RRR m d y 2007 Avaya Inc All Rights Reserved GrandstreamSIP doc Step Description 4 Select the ACCOUNTI tab and configure as follows Account Name Set to the Primary Handle field value configured in Section 4 Step 6 SIP Server Set to the SIP Domain field value configured in Section 4 St
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