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Page 1 of 92 EDCS-1203965 Rev. 1
Application Note
Avaya S8500 Rel. 5.2.1 using SIP via Cisco Unified
Communications Manager–Session Manager Edition 9.0 to Cisco
Unified Communications Manager 9.0 and Cisco Unified Border
Element (Enterprise Edition) Release 9.0 on ISR G2 to Service
Provider
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October 26, 2012 – Rev. 1
Table of Contents
Introduction .............................................................................................................................................................................................................. 4 Network Topology .................................................................................................................................................................................................... 5 Limitations ................................................................................................................................................................................................................ 5 System Components ................................................................................................................................................................................................. 6
Hardware Requirements ...................................................................................................................................................................................... 6 Software Requirements ....................................................................................................................................................................................... 6
Features .................................................................................................................................................................................................................... 7 Features Supported .............................................................................................................................................................................................. 7 Features Not Supported ....................................................................................................................................................................................... 7
Configuration ............................................................................................................................................................................................................ 8 Configuring the Avaya S8500 PBX .................................................................................................................................................................... 8
Software/Firmware Versions .......................................................................................................................................................................... 8 System Parameters IP Options ....................................................................................................................................................................... 9 IP Nodes ....................................................................................................................................................................................................... 11 IP Network Region ....................................................................................................................................................................................... 12 IP Codec Set ................................................................................................................................................................................................. 14 Signaling Group ........................................................................................................................................................................................... 15 Trunk Group ................................................................................................................................................................................................. 15 Route Pattern ................................................................................................................................................................................................ 18 AAR/ARS Analysis ..................................................................................................................................................................................... 19 Uniform Dialing Plan ................................................................................................................................................................................... 20 ISDN Public/Unknown Numbering Plan ..................................................................................................................................................... 21 Incoming-call-handling-trmt ........................................................................................................................................................................ 22 Station Configuration (IP Phone) ................................................................................................................................................................. 23 Station Configuration (Analog Line)............................................................................................................................................................ 25
Configuring the Cisco Unified Communications Manager – Session Manager Edition .................................................................................... 27 Cisco Unified Communications Manager – Session Manager Edition software version ............................................................................. 27 Configuration of Device Pool to Region mapping ....................................................................................................................................... 28 Configuration of Partitions ........................................................................................................................................................................... 28 Configuration of Calling Search Spaces ....................................................................................................................................................... 29 Configuration of Translation Pattern used to strip leading digits on inbound calls from SP ........................................................................ 31 Configuration of SIP Profile used by SIP trunks .......................................................................................................................................... 32 Configuration of SIP Normalization Script (used by SIP trunk to Avaya PBX) .......................................................................................... 38 Configuration of SIP trunks to PSTN ........................................................................................................................................................... 41 Configuration of SIP trunk to Avaya PBX ................................................................................................................................................... 44 Configuration of SIP trunk to Cisco UCM ................................................................................................................................................... 47 Configuration of Route Patterns – To Avaya PBX ...................................................................................................................................... 50 Configuration of Route Patterns – To Cisco UCM ...................................................................................................................................... 52 Configuration of Route Patterns – To PSTN ................................................................................................................................................ 54
Configuring the Cisco Unified Communications Manager ............................................................................................................................... 57 Cisco Unified Communications Manager Software Version ........................................................................................................................ 57 Configuration of Service Parameters – Cisco CallManager ......................................................................................................................... 58 Configuration of Audio Codec Preference List ............................................................................................................................................ 58 Configuration of Device Pool to Region mapping ....................................................................................................................................... 59 Configuration of Conference Bridge ............................................................................................................................................................ 60 Configuration of Media Resource Group ..................................................................................................................................................... 61 Configuration of Media Resource Group List .............................................................................................................................................. 62 Configuration of SIP Profile ........................................................................................................................................................................ 63 Configuration of SIP Trunk to SME ............................................................................................................................................................ 65 Configuration of Route Pattern to PSTN through SME ............................................................................................................................... 68 Configuration of Route Pattern to Avaya PBX through SME ...................................................................................................................... 69 Configuration of Cisco 7965 SIP Phone ...................................................................................................................................................... 70 Configuration of Cisco 7965 SCCP Phone................................................................................................................................................... 75
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Configuration of MGCP FAX Gateway (VG224) ........................................................................................................................................ 80 Configuration of MGCP FAX Gateway Analog Endpoint ........................................................................................................................... 81
Configuring the Cisco UBE - Enterprise ........................................................................................................................................................... 82 Acronyms .......................................................................................................................................................................................................... 90 Important Information ....................................................................................................................................................................................... 91
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Introduction
This application note describes the necessary steps and configurations for connectivity between Avaya S8500 release 5.2.1, and a Cisco
Unified Communications Manager (Cisco UCM) version 9.0 with Cisco Unified Communications Manager-Session Management
Edition (Cisco UCM-SME) Version 9.0.
The network topology diagram (Figures 1) shows the test setup for end-to-end interoperability between Cisco Unified Communications
Manager (Cisco UCM) Release 9.0 connected to the Avaya S8500 PBX via a Cisco UCM-SME using SIP trunks (between Cisco UCM-
SME and Avaya PBX) and SIP trunks (between the Cisco UCM-SME and Cisco UCM). Features tested are basic call, 3-way (ad-hoc)
conference, call transfer (attended and unattended), call forward (all, busy and no answer), hold/resume, fax transmission, and DTMF
interworking. This test setup also includes a connection to a Service Provider, using SIP trunks. Cisco Unified Border Element (Cisco
UBE) on ISR is used as a session border controller (SBC), providing demarcation, security, and interworking services between the
customer’s private network and the service provider’s SIP network.
During testing, a Cisco ISR 2921 voice gateway was used to run the Cisco Unified Border Element features set. However other Cisco
voice gateways can be used. The decision to choose the Cisco gateway model is left to the customer. The customer should choose a
Cisco IOS gateway model based on the capabilities and the capacity that will be required based on the planned network deployment.
Here is a list of Cisco IOS products capable of running Cisco UBE.
Cisco 3900 Series Integrated Services Routers
Cisco 2900 Series Integrated Services Routers
Cisco AS5350XM Universal Gateway
Cisco AS5400XM Universal Gateway
Cisco ASR 1000 Series Aggregation Services Routers
Consult your Cisco representative for the correct IOS image and for the specific application and Device Unit License and Feature
License requirements for all your Cisco Unified Communications Manager with Cisco Unified Border Element components.
If additional guidance on the Cisco UBE is needed, please refer to the Cisco UBE section on the Cisco Interoperability Portal
(www.cisco.com/go/interoperability).
This configuration was tested using a generic SIP Trunk Service Provider. Results may vary based on Service Provider being used.
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Network Topology
Figure 1. Basic Call Setup
Service Provider
or Simulated Service
Provider CUBECUBE
Avaya S8500
Phone PBX1
Phone PBX2
Cisco UCM-SME
Rel. 9.0(1)
Phone C2
SIP
SIP
SIP
SIP
Cisco UBE (on ISR 2921)
8.8 = IOS 15.2(4)M1
Phone C1
CPE to CPE
CPE to PSTN (SP)
FAX PBX3
FAX C3
Phone SP1 Phone SP1 FAX SP3
Cisco ISR – for IOS
MTP requirement
Cisco UCM
Rel. 9.0(1)
Capabilities
Voice/fax calls including supplementary services can be successfully established between endpoints controlled by the Avaya PBX and
endpoints controlled by the Cisco Unified Communications Manager.
Voice/fax calls including supplementary services can be successfully established between endpoints controlled by the Avaya PBX and
the PSTN, using Cisco UBE as a session border controller.
Limitations
Avaya PBX
Centralized Avaya voicemail using QSIG integration to the Avaya PBX is not supported. SIP-to-QSIG interworking on the Avaya does
not provide diversion information over the QSIG call leg. Centralized voicemail using Cisco Unity/Unity Connection integrated to Cisco
UCM is supported, so long as Diversion header is passed to Cisco Unified Communications Manager. This can be achieved by enabling
support of Diversion Header on the Avaya Communication Manager 5.0 SIP trunk group configuration form, or by using a SIP
Normalization Script converting History-info headers into Diversion headers.
T.38 fax-relay Error Correction Mode (ECM). The Avaya PBX does not currently support Error Correction Mode (ECM) during T.38
fax transmissions.
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Fax transmissions using Super Group 3 (SG3) fax protocol. The Avaya PBX currently supports fax transmission speeds up to 9600 bps
when T.38 fax relay is used. Also noticed during testing is the PBX’s inability to always detect SG3 fax tones. Because of this,
customers should avoid configuring fax machines connected to the Avaya PBX for SG3 protocol.
Cisco UBE (Enterprise Edition)
None found
Cisco UCM
Cisco UCM does not natively support History-info headers. A SIP Normalization script must be applied to SIP trunk(s) to Avaya PBX in
order to convert History-info (supported the Avaya PBX) to Diversion header (Cisco UCM-supported). This is required whenever a
Cisco UCM-hosted centralized voicemail platforms (such as Unity Connection) is used.
System Components
Hardware Requirements
Cisco MCS 7800 Unified Communications Manager Appliances
2 Cisco Unified IP phone 7965 configured as SCCP phones
2 Cisco Unified IP phone 7975 configured as SIP phones
Avaya S8500 PBX
Avaya digital and IP stations
TN799DP C-LAN Circuit Pack
TN2302AP Med-Pro Circuit Pack
Cisco ISR 2921 (Cisco UBE-Enterprise)
VG224 Analog Gateway (Fax application). Alternatively, other Cisco IOS gateway(s) can be used. This component may be an H.323,
SIP, or MGCP gateway. The protocol is optional, and the choice is left up to the customer’s network design.
Software Requirements
The following software is required:
Cisco Unified Communications Manager Release 9.0 - Session Manager Edition
Cisco Unified Communications Manager Release 9.0 – Cisco UCM
Cisco Unified Border Element (CISCO UBE) Release 9.0 with IOS version 15.2(4)M1. This configuration was tested with C2900-
universalk9_npe-mz.SPA.152-4.M1; however, this document is also applicable to all IOS 15.2(4)M/CUBE 9.0 and 15.3(2)T/CUBE 9.0
software releases. The documented CISCO UBE configuration can be supported with the following IOS feature sets: UNIVERSAL
Consult your Cisco representative for the correct IOS image and for the specific application and Device Unit License and Feature
License requirements for all your Cisco Unified Communications components. For reference, please follow this link:
http://www.cisco.com/en/US/prod/collateral/voicesw/ps6790/gatecont/ps5640/order_guide_c07_462222.html
Cisco Unity Connection (CUC) Release 8.6. This solution was tested with 8.6.1.20002-109
Cisco VG224 Gateway (IP-TDM) IOS version: 15.1(4)M4
Avaya Aura Communication Manager Release 5.2.1
Avaya 9600 Series H.323 IP Phone firmware version 3.1.1
Avaya TN799DP Firmware Vintage 39
Avaya TN2302AP Firmware Vintage 121
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Features
This section lists supported and unsupported features.
Features Supported
Basic calls
CLIP-Calling line (Number) identification presentation
CLIR-Calling line (Number) identification restriction
COLP-Connected line (Number) identification presentation
COLR- Connected line (Number) identification restriction
CNIP-Calling name identification presentation
CNIR-Calling name identification restriction
CONP-Connected name identification presentation
CONR- Connected name identification restriction
Consultation transfer – Local and Network/External
Early Attended transfer – Local and Network/External
Call forward Local – Unconditional, Busy and No reply (See Limitations section for details.)
Call forward Network/External – Unconditional, Busy and No reply (See Limitations section for details.)
DTMF interworking (using RFC 2833 DTMF relay)
MWI—Message Waiting Indicator (lamp ON, lamp OFF) across SIP trunk to Avaya PBX
Fax transmissions (T.38 and G.711 pass-through)
Features Not Supported
Centralized QSIG Voicemail hosted by Avaya PBX.
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Configuration
This section contains configuration menus and commands and describes configuration sequences and tasks.
Configuring the Avaya S8500 PBX
1. Configure node-name IP table to include Cisco UCM-SME as a valid IP node.
2. Configure the ip-network-region to assign to the SIP trunk.
3. Configure the ip-codec-set to assign to ip-network-region used by the SIP trunk.
4. Add the new signaling group.
5. Add the new trunk group.
6. Add the new route pattern.
7. Configure AAR/ARS Table entries.
8. Configure Uniform Dialing Plan.
9. Configure ISDN Public/Unknown Numbering Table entry.
10. Configure Incoming Call Handling Treatment for trunk group.
Configuration Menus and Commands for Avaya S8500 PBX
Software/Firmware Versions
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System Parameters IP Options
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IP Nodes
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IP Network Region
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IP Codec Set
Note: The ip-codec-set configuration above is assigned to SIP trunk(s) using codecs G.729a, G.711Mulaw, and T.38 fax relay. If the
Service Provider does not support T.38 fax relay, a trunk using G.711 codec is required, with FAX Mode set to “off”.
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Signaling Group
Trunk Group
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Route Pattern
Note: The Route Pattern configuration above is assigned to AAR code 217, used to route calls to Cisco UCM’s 4-digit extensions. As you
will notice, route pattern 445 “strips” the first 3 digits of the routing number (AAR code 217) before the INVITE message is transmitted
over the SIP trunk.
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Note: The Route Pattern configuration above is assigned to ARS analysis table entry 1408, used to route calls to the SP using 1+10-digit
dialing. As you will notice, route pattern 446 does not perform any “digit stripping” before the INVITE message is transmitted over the
SIP trunk, thus using all 11 digits.
AAR/ARS Analysis
Note: AAR code 217 is used to route calls to Cisco UCM extensions over Route Pattern 445. This code will be used in the Uniform
Dialplan table, and will be prefixed onto the 4-digit numbers assigned to Cisco UCM stations. As stated previously, Route Pattern 445
will “strip” the AAR code, leaving only the dialed 4-digit number in the outbound INVITE message.
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Note: Dial String 1408 is used to route calls to the Service Provider. After dialing 9 (ARS access code) plus 1-408-nxx-xxxx, all 11 digits
are included in the outbound INVITE message (parameter “Call Type” must be set to “natl”) and the call is routed over Route Pattern
446. This Route Pattern is configured not to “strip” any leading digits.
Uniform Dialing Plan
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ISDN Public/Unknown Numbering Plan
Note: The table above is used to define numbering plans to be used on ISDN/SIP calls. In the example above, 4-digit extensions in the
5XXX range are used on Cisco UCM, while 4-digit extensions in the 4XXX range are used by the Avaya PBX. Also note the digit
transformation performed for extensions 4114 and 4124: when calls are sent over trunk group 445 (trunk group used to connect the
PBX to Cisco UCM-SME), ext. 4114 is “transformed” into phone number 732-320-4353, while ext. 4124 is “transformed” into phone
number 732-320-4353 (SIP trunk Service Provider DID numbers).
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Incoming-call-handling-trmt
Note: The table above is used to apply changes to incoming called numbers. In this case, DID numbers provided by the Service Provider
do not match extension numbers, and have to be “translated”. SP-provided DID phone numbers 732-320-4352 and 732-320-4353, after
being modified by Cisco UCM-SME (configured to strip all leading digits, and pass the last 4 digits over the next call leg), need to also
be changed to their assigned extension numbers. As you will notice, this table is configured to change number 4352 to ext. 4124, and
4353 to ext. 4124
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Station Configuration (IP Phone)
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Station Configuration (Analog Line)
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Configuring the Cisco Unified Communications Manager – Session Manager Edition
1. Cisco UCM - Session Manager Edition software version
2. Device Pool and Region mapping configuration
3. Partitions configuration
4. Calling Search Space configuration
5. Translation Pattern configuration
6. SIP profile (used by SIP trunks) configuration
7. SIP Normalization Script (used by SIP trunk to Avaya PBX) configuration
8. SIP trunk configuration to SP
9. SIP trunk configuration to Avaya PBX
10. SIP Trunk configuration to Cisco UCM
11. Route Pattern configuration to Avaya PBX
12. Route Pattern configuration to Cisco UCM
13. Route Pattern configuration to SP
Cisco Unified Communications Manager – Session Manager Edition software version
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Configuration of Device Pool to Region mapping
Navigation Path: System Region
Configuration of Partitions
Navigation Path: Call Routing Class of Control Partition
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Configuration of Calling Search Spaces
Navigation Path: Call Routing Class of Control Calling Search Space
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Configuration of Translation Pattern used to strip leading digits on inbound calls from SP
Navigation Path: Call Routing Translation Pattern
Note: The Translation Pattern above is used by Cisco UCM - SME in order to strip leading digits from the called party number on
inbound calls. All leading digits are stripped, and only the last 4 digits are passed to the Avaya PBX and Cisco UCM. As you’ll notice
above, the SIP Trunk Service Provider prefixes 10-digit DID numbers with 5 zero’s.
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Configuration of SIP Profile used by SIP trunks
Navigation Path: Device Device Settings SIP Profile
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Configuration of SIP Normalization Script (used by SIP trunk to Avaya PBX)
Navigation Path: Device Device Settings SIP Normalization Script
Note: Historically, Avaya supported SIP History-Info header when providing call forward (diversion) information over SIP trunks. As
of Avaya Communication Manager software version 5.X, however, SIP trunk groups can be configured to provide Diversion header.
Cisco Unified Communications Manager uses Diversion header. A SIP Normalization script can be used in order to convert History-
Info headers into Diversion headers when connecting to Avaya PBX’s older than software version CM 5.X. This is useful whenever
Cisco Unity/Unity Connection centralized voicemail (integrated with Cisco Unified Communications Manager) is used to support both
Avaya and Cisco end users. The full content of the SIP Normalization Script is captured below:
M = {} M.allowHeaders = {"History-Info"} trace.enable() function M.outbound_INVITE(msg) local callid = msg:getHeader("Call-ID") trace.format("M.outbound_INVITE: callid is '%s'", callid) local di = msg:getHeader("Diversion") if not di then return end msg:convertDiversionToHI() msg:removeHeader("Diversion")
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local historyInfos = msg:getHeaderValues("History-Info") msg:removeHeader("History-Info") local newHi = "" for i, hi in ipairs(historyInfos) do local main_header = string.match(hi, '(.*)?') or string.match(hi, "(.*)>;index=(.*)") local embed_header = string.match (hi, '?Reason=sip(.*)>') local index = string.match(hi, '>;index=(.*)') local hiNext = historyInfos[i + 1] local indexNext = string.match(hiNext or "", '>;index=(.*)') trace.format("main_header is '%s'", main_header or "nil") if i == 1 then local firstHi = string.format("%s>;index=%s", main_header, index) firstHi = string.gsub(firstHi, "@(.*):%d+", "@%1") msg:addHeader("History-Info", firstHi) end if embed_header then trace.format("embed_header is '%s'", embed_header) embed_header = string.gsub(embed_header, "unconditional", "Moved Temporarily") embed_header = string.gsub(embed_header, ";", "%%3B") embed_header = string.gsub(embed_header , "=", "%%3D") embed_header = string.gsub(embed_header, "\"", "%%22") embed_header = string.gsub(embed_header, " ", "%%20") embed_header = string.format("?Reason=SIP%s%s", embed_header, "&Reason=Redirection%3Bcause%3DCFI") end -- Get rid of the port number main_header = string.gsub(main_header, "@(.*):%d+", "@%1") if not indexNext then local left, right = string.match(index, "(%d+)%.(%d+)") indexNext = string.format("%s.%s", left + 1, right) end hi = string.format("%s%s>;index=%s", main_header, embed_header or "", indexNext) msg:addHeader("History-Info", hi) end end local HiCauseToDiversion = { } HiCauseToDiversion["302"] = "unconditional" HiCauseToDiversion["486"] = "user-busy" HiCauseToDiversion["408"] = "no-answer" HiCauseToDiversion["480"] = "deflection" HiCauseToDiversion["487"] = "deflection" HiCauseToDiversion["503"] = "unavailable" HiCauseToDiversion["404"] = "unknown" function convertHIToDiversion(msg)
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local historyInfos = msg:getHeaderValues("History-Info") for i, hi in ipairs(historyInfos) do hi = string.gsub(hi, "%%3B", ";") hi = string.gsub(hi, "%%3D", "=") hi = string.gsub(hi, "%%22", "\"") hi = string.gsub(hi, "%%20", " ") -- Reason=SIP;cause=302;text="Moved Temporarily" local uri, reason, cause, text = string.match(hi, "<(sip:.*@.*)?Reason=(SIP);cause=(%d+);text=(\".*\")") trace.format("hi: uri '%s', reason '%s', cause '%s', text '%s'", uri or "nil", reason or "nil", cause or "nil", text or "nil") if reason == "SIP" then local dReason = HiCauseToDiversion[cause] or "unknown" local diversion = string.format("<%s>;reason=\"%s\"", uri, dReason) msg:addHeader("Diversion", diversion) end end end function M.inbound_INVITE(msg) local callid = msg:getHeader("Call-ID") trace.format("M.inbound_INVITE: callid is '%s'", callid) local hist = msg:getHeader("History-Info") local di = msg:getHeader("Diversion") if hist then local context = msg:getContext() if context then context["History-Info"] = hist end if not di then convertHIToDiversion(msg) end end local di = msg:getHeader("Diversion") if di then trace.format(" -- found Diversion header") msg:removeHeader("History-Info") -- replace unknown to unconditional di = string.gsub(di, "unknown", "unconditional") msg:modifyHeader("Diversion", di) end end
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--[[ function M.outbound_ANY_INVITE(msg) local context = msg:getContext() if context then msg:addHeader("History-Info", context["History-Info"]) end end --]] return M
Configuration of SIP trunks to PSTN
Navigation Path: Device Trunk
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Note: Generally, SIP Trunk Service Providers require all outbound calls over SIP trunk to provide a valid 10-digit DID number as
caller ID. Configuring a valid 10-digit Caller ID DN in parameter Caller Information Caller ID DN ensures that outbound calls will
be processed even when originating from telephones not configured with valid 10-digit DID numbers.
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Configuration of SIP trunk to Avaya PBX
Navigation Path: Device Trunk
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Configuration of SIP trunk to Cisco UCM
Navigation Path: Device Trunk
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Configuration of Route Patterns – To Avaya PBX
Navigation Path: Call Routing Route/Hunt Route Pattern
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Configuration of Route Patterns – To Cisco UCM
Navigation Path: Call Routing Route/Hunt Route Pattern
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Configuration of Route Patterns – To PSTN
Navigation Path: Call Routing Route/Hunt Route Pattern
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Configuring the Cisco Unified Communications Manager
1. Cisco Unified Communications Manager Version
2. Service Parameters configuration
3. Audio Codec Preference List configuration
4. Device pool and Region mapping configuration
5. Conference Bridge configuration
6. Media Resource Group configuration
7. Media Resource Group List configuration
8. SIP Profile configuration
9. SIP Trunk to SME configuration
10. Route Pattern configuration to SP
11. Route Pattern configuration to Avaya
12. Cisco IP Phone 7960 SCCP Configuration
13. Cisco IP Phone 7960 SIP Configuration
14. MGCP Fax gateway configuration
Cisco Unified Communications Manager Software Version
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Configuration of Service Parameters – Cisco CallManager
Navigation Path: System Service Parameters
Note: Service Parameter “Duplex Streaming Enabled” must be set to “True” in order to successfully provide MoH/Ringback to Avaya
IP phones (H.323) and outside (PSTN) callers when calls are placed on hold and/or transferred from Cisco UCM stations.
Configuration of Audio Codec Preference List
Navigation Path: System Region Information Audio Codec Preference List
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Note: CISCO UCM 9.0 introduced a new feature: Audio Codec Preference List. This feature allows for configuration of the order of
audio codec preference, both for inter- and intra-Region calls. Add a new Audio Codec Preference List, with G.729 codecs configured
above G.711 (higher priority). This new Audio Codec Preference list is assigned to the Region used by the Device Pool for Conference
Bridges, Transcoders, and phones requiring G.711 codec. With this configuration in place, inbound calls, as well as call conferences
initiated by Cisco IP phones, will use G.729 codec as their first choice codec.
Configuration of Device Pool to Region mapping
Navigation Path: System Region Information Region
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Configuration of Conference Bridge
Navigation Path: Media Resources Conference Bridge
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Conference Bridge IOS configuration:
sccp local GigabitEthernet0/0
sccp ccm 172.20.236.50 identifier 1 version 7.0
sccp
!
sccp ccm group 1
bind interface GigabitEthernet0/0
associate ccm 1
priority 1
associate profile 98 register cfb112233445566
!
dspfarm profile 98 conference
codec g729r8
codec g711ulaw
maximum sessions 8
associate application SCCP
Configuration of Media Resource Group
Navigation Path: Media Resources Media Resource Group
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Configuration of Media Resource Group List
Navigation Path: Media Resources Media Resource Group List
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Configuration of SIP Profile
Navigation Path: Device Device Settings SIP Profile
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Configuration of SIP Trunk to SME
Navigation Path: Device Trunk
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Configuration of Route Pattern to PSTN through SME
Navigation Path: Call Routing Route/Hunt Route Pattern
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Configuration of Route Pattern to Avaya PBX through SME
Navigation Path: Call Routing Route/Hunt Route Pattern
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Configuration of Cisco 7965 SIP Phone
Navigation Path: Device Phone
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Configuration of Cisco 7965 SCCP Phone
Navigation Path: Device Phone
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Configuration of MGCP FAX Gateway (VG224)
Navigation Path: Device Gateway
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Configuration of MGCP FAX Gateway Analog Endpoint
Navigation Path: Device Gateway
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Configuring the Cisco UBE - Enterprise
CUBE_ISRG2_ATT#show version
Cisco IOS Software, C2900 Software (C2900-UNIVERSALK9_NPE-M), Version 15.2(4)M1,
RELEASE SOFTWARE (fc1)
Technical Support: http://www.cisco.com/techsupport
Copyright (c) 1986-2012 by Cisco Systems, Inc.
Compiled Thu 26-Jul-12 20:54 by prod_rel_team
ROM: System Bootstrap, Version 15.0(1r)M1, RELEASE SOFTWARE (fc1)
CUBE_ISRG2_ATT uptime is 1 week, 5 days, 6 hours, 0 minutes
System returned to ROM by power-on
System image file is "flash:c2900-universalk9_npe-mz.SPA.152-4.M1.bin"
Last reload type: Normal Reload
Last reload reason: power-on
Cisco CISCO2921/K9 (revision 1.0) with 487424K/36864K bytes of memory.
Processor board ID FTX1348AHMN
3 Gigabit Ethernet interfaces
1 terminal line
DRAM configuration is 64 bits wide with parity enabled.
255K bytes of non-volatile configuration memory.
255488K bytes of ATA System CompactFlash 0 (Read/Write)
License Info:
License UDI:
-------------------------------------------------
Device# PID SN
-------------------------------------------------
*0 CISCO2921/K9 FTX1348AHMN
Technology Package License Information for Module:'c2900'
------------------------------------------------------------------------------------
Technology Technology-package Technology-package
Current Type Next reboot
------------------------------------------------------------------------------------
ipbase ipbasek9 Permanent ipbasek9
security None None None
uc uck9 Permanent uck9
data None None None
Configuration register is 0x2102
CUBE_ISRG2_ATT#show running
Building configuration...
Current configuration : 10206 bytes
!
! Last configuration change at 10:45:48 PST Mon Aug 27 2012
version 15.2
service timestamps debug datetime msec
service timestamps log datetime msec
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no service password-encryption
service sequence-numbers
!
hostname CUBE_ISRG2_ATT
!
boot-start-marker
boot system flash:c2900-universalk9_npe-mz.SPA.152-4.M1.bin
boot-end-marker
!
!
logging queue-limit 10000
logging buffered 20000000
logging persistent filesize 20000000
logging rate-limit 10000
no logging console
enable secret 5 $1$4jgu$npJCRdswNO47pZhBy3fbi/
enable password cisco
!
no aaa new-model
clock timezone PST -8 0
!
!
!
ip domain name yourdomain.com
no ip cef
no ipv6 cef
multilink bundle-name authenticated
!
!
!
crypto pki trustpoint TP-self-signed-1100168695
enrollment selfsigned
subject-name cn=IOS-Self-Signed-Certificate-1100168695
revocation-check none
rsakeypair TP-self-signed-1100168695
!
!
crypto pki certificate chain TP-self-signed-1100168695
certificate self-signed 01
30820254 308201BD A0030201 02020101 300D0609 2A864886 F70D0101 04050030
31312F30 2D060355 04031326 494F532D 53656C66 2D536967 6E65642D 43657274
69666963 6174652D 31313030 31363836 3935301E 170D3132 30383038 31363134
33365A17 0D323030 31303130 30303030 305A3031 312F302D 06035504 03132649
4F532D53 656C662D 5369676E 65642D43 65727469 66696361 74652D31 31303031
36383639 3530819F 300D0609 2A864886 F70D0101 01050003 818D0030 81890281
8100DC2A 57584BA4 9E848CCB AE8FC0F9 4CC9D68A 56FEE2D8 B3711442 0A45DCE7
875F5622 E504025D 3BF6DDE7 894E8DDC 6F7A6E21 7EB2012F 551B8100 DBF61436
89CB7CA4 2C40EDCD 395AC7D0 F759F0C8 E942220B FDB9F6E9 34067A81 DE1BE5A9
DA7FB98E 533FB54E E3747FA3 758F19D9 BA886C9A 16FCD1A7 3B3DD80C 195110B2
A62B0203 010001A3 7C307A30 0F060355 1D130101 FF040530 030101FF 30270603
551D1104 20301E82 1C55434D 2D495352 47322D41 54542E79 6F757264 6F6D6169
6E2E636F 6D301F06 03551D23 04183016 80148994 E1E21AE2 848D4387 C647F727
5DB0345F 0ACC301D 0603551D 0E041604 148994E1 E21AE284 8D4387C6 47F7275D
B0345F0A CC300D06 092A8648 86F70D01 01040500 03818100 050B77EA 063D97C6
75C7E8E6 C256D569 86872149 ABD193ED DAE3220D F41FA5F0 867494BD D53F54A0
0BCD8990 2108F2E0 74D872AA 7038638F A1316A6A ADC28EDA 440B4483 DB11E722
F695EA40 98762C08 34FF30F9 0ADF2F99 516B51BF F285B76D 4A047C85 04DA0632
B55DCDF7 000F9BAC 22E7899F 50C2E66C 7A2CCC7E 7A82FEA6
quit
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voice-card 0
dspfarm
dsp services dspfarm
!
!
!
voice service voip
address-hiding
mode border-element
allow-connections sip to sip
no supplementary-service sip moved-temporarily
redirect ip2ip
h323
sip
error-passthru
asserted-id pai1 early-offer forced
midcall-signaling passthru
privacy-policy passthru
g729 annexb-all
!
voice class codec 12 codec preference 1 g729r8 bytes 30
codec preference 2 g711ulaw
!
voice class sip-profiles 13 response ANY sip-header Allow-Header modify "UPDATE," ""
request INVITE sdp-header Audio-Attribute modify "a=ptime:20" "a=ptime:30"
response ANY sdp-header Audio-Attribute modify "a=ptime:20" "a=ptime:30"
request REINVITE sdp-header Attribute modify "a=T38FaxFillBitRemoval:0" ""
request INVITE sdp-header Audio-Attribute add "a=ptime:30"
!
!
license udi pid CISCO2921/K9 sn FTX1348AHMN
!
!
archive
log config
hidekeys
username Cisco password 0 cisco
!
redundancy
!
!
no ip ftp passive
!
translation-rule 1
!
!
!
1 This command enables router to send P-Asserted ID within the SIP Message Header. Alternatively, this command can also be applied to
individual dial-peers (voice-class sip asserted-id pai) 2 This command configures the codec preference to be assigned to dial-peers. Alternatively, single codec’s can be configured into individual
dial-peers 3 This SIP Profile removes UPDATE from SIP Message Header to/from Cisco UCM, as it can cause problems during unattended call transfers.
By default, Cisco 6900-series IP phones use ptime value of 20 ms. Some Service Provider networks prefer ptime value of 30 ms. This SIP
profile modifies SDP ptime value from 20 to 30. It is also used to remove SDP media attribute "a=T38FaxFillBitRemoval:0", as it can cause
T.38 fax relay transmissions to fail
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!
!
interface Embedded-Service-Engine0/0
no ip address
shutdown
!
interface GigabitEthernet0/0
description Connection to UC Interop lab network
ip address 172.20.110.158 255.255.255.0
duplex auto
speed auto
!
interface GigabitEthernet0/1
description connection to ATT Network
ip address 99.136.XXX.XXX 255.255.255.0
duplex auto
speed auto
!
interface GigabitEthernet0/2
no ip address
shutdown
duplex auto
speed auto
!
ip default-gateway 172.20.110.1
ip forward-protocol nd
!
ip http server
ip http access-class 23
ip http authentication local
ip http secure-server
ip http timeout-policy idle 60 life 86400 requests 10000
!
ip dns server view-group DNS
no ip pim dm-fallback
ip route 172.20.0.0 255.255.0.0 172.20.110.1
ip route 172.30.0.0 255.255.0.0 172.20.43.1
ip route 207.242.XXX.XXX 255.255.255.255 99.136.XXX.XXX
ip route 207.242.XXX.XXX 255.255.255.255 99.136.XXX.XXX
!
access-list 23 permit 10.10.10.0 0.0.0.7
!
!
control-plane
!
!
no ccm-manager fax protocol cisco
ccm-manager music-on-hold
!
mgcp
no mgcp package-capability res-package
no mgcp package-capability fxr-package
mgcp fax t38 ecm
!
mgcp profile default
!
!
dial-peer voice 1999 voip
description Outgoing calls to SP - Facing SP network
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destination-pattern 1T
session protocol sipv2
session target ipv4:207.242.xxx.xxx
voice-class codec 1
voice-class sip asymmetric payload full
voice-class sip asserted-id pai
voice-class sip privacy-policy passthru
voice-class sip early-offer forced
voice-class sip profiles 1
dtmf-relay rtp-nte4 fax-relay sg3-to-g3
fax rate 14400 bytes 48
fax protocol t38 version 0 ls-redundancy 5 hs-redundancy 1 fallback pass-through g711ulaw5 no vad
!
dial-peer voice 2000 voip
description Outgoing calls to SP - Facing Cisco UCM-SME
session protocol sipv2
incoming called-number 1T
voice-class codec 1
voice-class sip asymmetric payload full
voice-class sip asserted-id pai
voice-class sip profiles 1
dtmf-relay rtp-nte
fax-relay sg3-to-g3
fax protocol t38 version 0 ls-redundancy 5 hs-redundancy 1 fallback pass-through g711ulaw
no vad
!
dial-peer voice 732 voip
description Incoming calls from SP - Facing Cisco UCM-SME
destination-pattern 00000[37][13][24].......
session protocol sipv2
session target ipv4:172.20.236.252
voice-class codec 1
voice-class sip asymmetric payload full
voice-class sip asserted-id pai
voice-class sip privacy-policy passthru
voice-class sip profiles 1
dtmf-relay rtp-nte
fax-relay sg3-to-g3
fax rate 14400 bytes 48
fax protocol t38 version 0 ls-redundancy 5 hs-redundancy 1 fallback pass-through g711ulaw
no vad
!
dial-peer voice 733 voip
description Incoming calls from SP - Facing SP network
session protocol sipv2
incoming called-number 00000[37][13][24].......
voice-class codec 1
voice-class sip asymmetric payload full
voice-class sip asserted-id pai
voice-class sip profiles 1
dtmf-relay rtp-nte
fax-relay sg3-to-g3
4 This command enables DTMF digit passing using RTP NTE (RFC2833) to calls matching this dial-peer
5 This command enables Cisco UBE to perform T.38 fax relay. To change fax protocol to pass-through using G.711mulaw, the command has to
be changed to “fax protocol pass-through g711ulaw”
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fax rate 14400
fax protocol t38 version 0 ls-redundancy 5 hs-redundancy 1 fallback pass-through g711ulaw
no vad
!
dial-peer voice 11 voip
description Outgoing Intl Calls to SP - Facing SP network
destination-pattern 011T
session protocol sipv2
session target ipv4:207.242.xxx.xxx
voice-class codec 1
voice-class sip asymmetric payload full
voice-class sip asserted-id pai
voice-class sip privacy-policy passthru
voice-class sip early-offer forced
voice-class sip profiles 1
dtmf-relay rtp-nte
fax-relay sg3-to-g3
fax rate 14400 bytes 48
fax protocol t38 version 0 ls-redundancy 5 hs-redundancy 1 fallback pass-through g711ulaw
no vad
!
dial-peer voice 12 voip
description Outgoing Intl calls to SP - Facing Cisco UCM-SME
session protocol sipv2
incoming called-number 011T
voice-class codec 1
voice-class sip asymmetric payload full
voice-class sip profiles 1
dtmf-relay rtp-nte
fax-relay sg3-to-g3
fax protocol t38 version 0 ls-redundancy 5 hs-redundancy 1 fallback pass-through g711ulaw
no vad
!
dial-peer voice 511 voip
description N11 Calls to SP - Facing SP network
destination-pattern [459]11
session protocol sipv2
session target ipv4:207.242.xxx.xxx
voice-class codec 1
voice-class sip asymmetric payload full
voice-class sip asserted-id pai
voice-class sip privacy-policy passthru
voice-class sip early-offer forced
voice-class sip profiles 1
dtmf-relay rtp-nte
no vad
!
dial-peer voice 512 voip
description N11 Calls to SP - Facing Cisco UCM-SME
session protocol sipv2
incoming called-number .11
voice-class codec 1
voice-class sip asymmetric payload full
voice-class sip asserted-id pai
voice-class sip profiles 1
dtmf-relay rtp-nte
no vad
!
dial-peer voice 2001 voip
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description Incoming local 7-digit calls from SP - Facing Cisco UCM-SME
destination-pattern 3204...
session protocol sipv2
session target ipv4:172.20.236.50
voice-class codec 1
voice-class sip asymmetric payload full
voice-class sip asserted-id pai
voice-class sip privacy-policy passthru
voice-class sip early-offer forced
voice-class sip profiles 1
dtmf-relay rtp-nte
fax-relay sg3-to-g3
fax rate 14400 bytes 48
fax protocol t38 version 0 ls-redundancy 5 hs-redundancy 1 fallback pass-through g711ulaw
no vad
!
dial-peer voice 2002 voip
description Incoming local 7-digit calls from SP - Facing SP network
session protocol sipv2
incoming called-number 3204...
voice-class codec 1
voice-class sip asymmetric payload full
voice-class sip asserted-id pai
voice-class sip profiles 1
dtmf-relay rtp-nte
fax-relay sg3-to-g3
fax rate 14400 bytes 48
fax protocol t38 version 0 ls-redundancy 5 hs-redundancy 1 fallback pass-through g711ulaw
no vad
!
dial-peer voice 2005 voip
description Outgoing call to Operator – Facing SP network
destination-pattern 0T
session protocol sipv2
session target ipv4:207.242.xxx.xxx
voice-class codec 1
voice-class sip asymmetric payload full
voice-class sip asserted-id pai
voice-class sip privacy-policy passthru
voice-class sip early-offer forced
voice-class sip profiles 1
dtmf-relay rtp-nte
no vad
!
dial-peer voice 2006 voip
description Outgoing call to Operator – Facing Cisco UCM-SME
session protocol sipv2
incoming called-number 0T
voice-class codec 1
voice-class sip asymmetric payload full
voice-class sip asserted-id pai
voice-class sip profiles 1
dtmf-relay rtp-nte
no vad
!
!
sip-ua
no remote-party-id
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disable-early-media 1806 retry invite 2
!
!
!
gatekeeper
shutdown
!
!
!
line con 0
exec-timeout 0 0
line aux 0
line 2
no activation-character
no exec
transport preferred none
transport input all
transport output pad telnet rlogin lapb-ta mop udptn v120 ssh
stopbits 1
line vty 0 4
exec-timeout 15 0
password cisco
login
transport input all
!
scheduler allocate 20000 1000
!
end
6 This command allows Cisco UBE to disable early-media upon receiving 180 Ringing message. It is required in order to provide ringback tone
during unattended call transfers
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Acronyms
Acronym Definitions
ANF-PR Additional Network Feature Path Replacement
AOC Advice-of-charge. Information element is sent with the connection setup information for incoming Euro-ISDN connections. The AOC IE is used for call charge calculation.
Cisco UCM Cisco Unified Communications Manager
CCBS Call Completion to Busy Subscriber
CCNR Call Completion on No Reply
CFB Call Forwarding on Busy
CFNR Call Forwarding No Reply
CFU Call Forwarding Unconditional
CLIP Calling Line (Number) Identification Presentation
CLIR Calling Line (Number) Identification Restriction
CNIP Calling Name Identification Presentation
CNIR Calling Name Identification Restriction
COLP Connected Line (Number) Identification Presentation
COLR Connected Line (Number) Identification Restriction
CONP Connected Name Identification Presentation
CONR Connected Name Identification Restriction
CT Call Transfer
MWI Message Waiting Indicator
PSTN Public Switched Telephone Network
SP Service Provider
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Important Information
THE SPECIFICATIONS AND INFORMATION REGARDING THE PRODUCTS IN THIS MANUAL ARE SUBJECT TO CHANGE
WITHOUT NOTICE. ALL STATEMENTS, INFORMATION, AND RECOMMENDATIONS IN THIS MANUAL ARE BELIEVED TO
BE ACCURATE BUT ARE PRESENTED WITHOUT WARRANTY OF ANY KIND, EXPRESS OR IMPLIED. USERS MUST TAKE
FULL RESPONSIBILITY FOR THEIR APPLICATION OF ANY PRODUCTS.
IN NO EVENT SHALL CISCO OR ITS SUPPLIERS BE LIABLE FOR ANY INDIRECT, SPECIAL, CONSEQUENTIAL, OR
INCIDENTAL DAMAGES, INCLUDING, WITHOUT LIMITATION, LOST PROFITS OR LOSS OR DAMAGE TO DATA
ARISING OUT OF THE USE OR INABILITY TO USE THIS MANUAL, EVEN IF CISCO OR ITS SUPPLIERS HAVE BEEN
ADVISED OF THE POSSIBILITY OF SUCH DAMAGES.
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Corporate
Headquarters
Cisco Systems, Inc.
170 West Tasman Drive
San Jose, CA 95134-1706
USA
www.cisco.com
Tel: 408 526-4000
800 553-NETS (6387)
Fax: 408 526-4100
European
Headquarters
Cisco Systems International
BV
Haarlerbergpark
Haarlerbergweg 13-19
1101 CH Amsterdam
The Netherlands
www-europe.cisco.com
Tel: 31 0 20 357 1000
Fax: 31 0 20 357 1100
Americas
Headquarters
Cisco Systems, Inc.
170 West Tasman Drive
San Jose, CA 95134-1706
USA
www.cisco.com
Tel: 408 526-7660
Fax: 408 527-0883
Asia Pacific
Headquarters
Cisco Systems, Inc.
Capital Tower
168 Robinson Road
#22-01 to #29-01
Singapore 068912
www.cisco.com
Tel: +65 317 7777
Fax: +65 317 7799
Cisco Systems has more than 200 offices in the following countries and regions. Addresses, phone numbers, and fax numbers are listed on
the Cisco Web site at www.cisco.com/go/offices.
Argentina • Australia • Austria • Belgium • Brazil • Bulgaria • Canada • Chile • China PRC • Colombia • Costa Rica • Croatia • Czech
Republic • Denmark • Dubai, UAE • Finland • France • Germany • Greece • Hong Kong SAR • Hungary • India • Indonesia • Ireland •
Israel • Italy • Japan • Korea • Luxembourg • Malaysia • Mexico • The Netherlands • New Zealand • Norway • Peru • Philippines •
Poland • Portugal • Puerto Rico • Romania • Russia • Saudi Arabia • Scotland • Singapore • Slovakia • Slovenia • South Africa • Spain •
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