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Document Overview

This document provides a configuration guide for Sonus Session Border Controller (SBC) SBC SWe Series when connecting to Cisco Unified Communications Manager 11 (CUCM 11) and CenturyLink SIP Trunk.

This configuration guide supports features given in Cisco UCM configuration guide.

Introduction

The interoperability compliance testing focuses on verifying various inbound and outbound call flows between Sonus SBC SWe Series and CUCM 11 .

Audience

This technical document is intended for telecommunication engineers with the purpose of configuring the Sonus SBC SWe series aspects of the CenturyLink  SIP trunk group together with the CUCM 11 . Navigating a third-party server and Sonus SBC Web browser user interface, Embedded Management Application (EMA) is required. Understanding the basic concepts for IP/Routing, SIP/RTP and TLS are also required for completing the configuration and any necessary troubleshooting.

This configuration guide is offered as a convenience to Sonus customers. The specifications and information regarding the product in this guide are subject to change without notice. All statements, information, and recommendations in this guide are believed to be accurate but are presented without warranty of any kind, express or implied, and are provided “AS IS”. Users must take full responsibility for the application of the specifications and information in this guide.

Requirements

The following equipment and software were used for the sample configuration provided:

Table : Requirements

 EquipmentVersion
Sonus NetworksSBC SWeV05.01.00-R000
 ConnexIP OSV03.01.00-R000
 SonusDBV05.01.00-R000
 EMAV05.01.00-R000
Third-Party EquipmentCisco Unified Communications Manager 11.0.1.21900-11

Cisco IP Phone 7942

9.3.1.57

 

Reference Configuration

The following reference configuration shows connectivity between third-party server and Sonus SBC SWe.

Figure : Reference Configuration

 

Support

For any questions regarding this document or the content herein, please contact your maintenance and support provider.

 

Third-party Product Features

The testing was executed with the CenturyLink test plan, and the following features were tested:

  • Basic originated and terminated calls
  • Basic inbound/outbound call
  • Hold and resume
  • Call Forwarding Unconditional
  • FAX
  • DTMF
  • Dual Trunk
  • Features codes

Verify License

No special licensing required


CUCM 11 Configuration

The following new configurations are included in this section:

  1. Security Profile
  2. SIP Profile
  3. SIP Trunk
  4. Route Group
  5. Route List
  6. Route Pattern

1.Security Profile

Select System > Security > SIP Trunk Security Profile

Figure : Security Profile First Trunk

Figure : Security Profile Second Trunk

 

2.SIP Profile

Select Device > Device Settings > SIP Profile

Figure : SIP Profile

 


3.SIP Trunk

Select Device > Trunk > Add New

Figure : First SIP Trunk

 


 

Figure : Second SIP Trunk

 

4.Route Group

Select Call Routing > Route/Hunt > Route Group > Add New

Figure : Route Group

 

5.Route List

Select Call Routing > Route/Hunt > Route List > Add New

Figure : Route List

 

6.Route Pattern

Select Call Routing > Route/Hunt > Route Pattern > Add New

Figure : Route Pattern


Sonus SBC SWe Series Configuration

 This section provides example CLI configuration details for the SWe Series.

 

Complete Configuration
configure

#UDP Port Range for RTP (media)
set system media mediaPortRange baseUdpPort 1024 maxUdpPort 65148
commit
 
#DSP Resources
set system mediaProfile compression 90 tone 10
commit

#CUCM11 codecs
set profiles media codecEntry G711u_CUCM codec g711 packetSize 20 law ULaw
set profiles media codecEntry G711u_CUCM dtmf relay rfc2833 removeDigits enable
set profiles media codecEntry G711u_CUCM fax failureHandling continue toneTreatment none
set profiles media codecEntry G711u_CUCM modem failureHandling continue toneTreatment none
commit

#CenturyLink codecs
set profiles media codecEntry G729_CenturyLink codec g729a packetSize 20 preferredRtpPayloadType 128
set profiles media codecEntry G711u_CenturyLink codec g711 packetSize 20 law ULaw
set profiles media codecEntry G711u_CenturyLink dtmf relay rfc2833 removeDigits enable
set profiles media codecEntry G729_CenturyLink dtmf relay rfc2833 removeDigits enable
set profiles media codecEntry G711u_CenturyLink fax failureHandling continue toneTreatment ignoreDetectionAllowPeerToNegotiateFaxRelay
set profiles media codecEntry G729_CenturyLink fax failureHandling continue toneTreatment ignoreDetectionAllowPeerToNegotiateFaxRelay
set profiles media codecEntry G729_CenturyLink modem failureHandling continue toneTreatment applyFaxTreatment
set profiles media codecEntry G711u_CenturyLink modem failureHandling continue toneTreatment applyFaxTreatment
commit

#Tenor fax codecs
set profiles media codecEntry G729_Tenor_T38 codec g729a packetSize 20 preferredRtpPayloadType 128
set profiles media codecEntry G711u_Tenor_T38 codec g711 packetSize 20 law ULaw
set profiles media codecEntry G711u_Tenor_T38 dtmf relay rfc2833 removeDigits enable
set profiles media codecEntry G729_Tenor_T38 dtmf relay rfc2833 removeDigits enable
set profiles media codecEntry G711u_Tenor_T38 fax failureHandling continue toneTreatment ignoreDetectionAllowPeerToNegotiateFaxRelay
set profiles media codecEntry G729_Tenor_T38 fax failureHandling continue toneTreatment ignoreDetectionAllowPeerToNegotiateFaxRelay
set profiles media codecEntry G729_Tenor_T38 modem failureHandling continue toneTreatment applyFaxTreatment
set profiles media codecEntry G711u_Tenor_T38 modem failureHandling continue toneTreatment applyFaxTreatment
commit

#Internal Side Configuration
#IP Interface Group
set addressContext default ipInterfaceGroup Private ipInterface Private-pkt0 ceName SWECENTURYLINK01 portName pkt0 ipAddress 10.35.179.249 prefix 26 mode outOfService state disabled
set addressContext default ipInterfaceGroup Private ipInterface Private-pkt0 mode inService state enabled
commit

#IP Static Route
set addressContext default staticRoute 0.0.0.0 0 10.35.179.193 Private Private-pkt0 preference 100
commit

#SBC Configuration for CUCM11 Trunk
#Packet Service Profile (PSP)
set profiles media packetServiceProfile CUCM11_PSP packetToPacketControl codecsAllowedForTranscoding thisLeg g711u,g729 otherLeg g711u,g729
set profiles media packetServiceProfile CUCM11_PSP packetToPacketControl transcode conditional
set profiles media packetServiceProfile CUCM11_PSP rtcpOptions rtcp enable terminationForPassthrough enable
set profiles media packetServiceProfile CUCM11_PSP silenceInsertionDescriptor g711SidRtpPayloadType 13 heartbeat enable
commit

#IP Signaling profiles(IPSP)
set profiles signaling ipSignalingProfile CUCM11_IPSP ipProtocolType sipOnly
set profiles signaling ipSignalingProfile CUCM11_IPSP commonIpAttributes flags disableMediaLockDown enable includeReasonHeader enable
set profiles signaling ipSignalingProfile CUCM11_IPSP commonIpAttributes flags minimizeRelayingOfMediaChangesFromOtherCallLegAll enable publishIPInHoldSDP enable sendPtimeInSdp enable
set profiles signaling ipSignalingProfile CUCM11_IPSP commonIpAttributes flags sendRtcpPortInSdp enable
set profiles signaling ipSignalingProfile CUCM11_IPSP commonIpAttributes relayFlags notify enable
set profiles signaling ipSignalingProfile CUCM11_IPSP commonIpAttributes transparencyFlags  mwiBody enable  unknownBody enable unknownHeader enable
set profiles signaling ipSignalingProfile CUCM11_IPSP commonIpAttributes flags includeTransportTypeInContactHeader enable
set profiles signaling ipSignalingProfile CUCM11_IPSP ingressIpAttributes flags sendSdpIn200OkIf18xReliable enable sendSdpInSubsequent18x enable
set profiles signaling ipSignalingProfile CUCM11_IPSP egressIpAttributes flags disable2806Compliance enable
set profiles signaling ipSignalingProfile CUCM11_IPSP egressIpAttributes flags disableOptionalRegisterParameters enable
set profiles signaling ipSignalingProfile CUCM11_IPSP egressIpAttributes transport type1 tcp type2 udp type3 none type4 none
commit
 
#Signaling profiles(SP)
set profiles signaling signalingProfile CUCM11_SP egress egressFlags addPrefix011ForInternationalCalls enable convertNumbersToE164Format enable
commit

#Transparency profile(TP)
set profiles services transparencyProfile CUCM11_TP sipMessageBody all excludedMethods register invite subscribe info publish message refer options update
set profiles services transparencyProfile CUCM11_TP state enabled
commit

#E.164 Profile
set profiles signaling E164Profile CenturyLink_E164  sonusE164ProfCharStar allow sonusE164ProfCharHash allow sonusE164ProfCharHyphen allow
commit

#zone
set addressContext default zone ZONE-INT-VoIP id 3
commit

set addressContext default zone ZONE-INT-CUCM id 4
commit

#SIP signaling port
set addressContext default zone ZONE-INT-VoIP sipSigPort 25 ipInterfaceGroupName Private ipAddressV4 10.35.179.250 portNumber 5060 mode outOfService state disabled siprec disabled transportProtocolsAllowed sip-tcp
set addressContext default zone ZONE-INT-VoIP sipSigPort 25 mode inService state enabled
commit
set addressContext default zone ZONE-INT-CUCM sipSigPort 1 ipInterfaceGroupName Private ipAddressV4 10.35.179.253 portNumber 5060 mode outOfService state disabled siprec disabled 
transportProtocolsAllowed sip-tcp
set addressContext default zone ZONE-INT-CUCM sipSigPort 1 mode inService state enabled
commit

#IP Peer
set addressContext default zone ZONE-INT-VoIP ipPeer CUCM11_SIP_SERVER policy sip fqdnPort 0
set addressContext default zone ZONE-INT-VoIP ipPeer CUCM11_SIP_SERVER ipAddress 10.35.180.110 ipPort 5060
set addressContext default zone ZONE-INT-VoIP ipPeer CUCM11_SIP_SERVER authentication incInternalCredentials enabled
set addressContext default zone ZONE-INT-VoIP ipPeer CUCM11_SIP_SERVER authentication intChallengeResponse enabled
set addressContext default zone ZONE-INT-VoIP ipPeer CUCM11_SIP_SERVER surrogateRegistration userPart 4696806537 authUserName xxxxxxx-4696806537 regAuthPassword xxxx state enabled sendCredentials 
challengeForAnyMessageAndInDialogRequests hostPart voip.centurylink.com
commit

set addressContext default zone ZONE-INT-CUCM ipPeer CUCM11_SIP_SERVER2 policy sip fqdnPort 0
set addressContext default zone ZONE-INT-CUCM ipPeer CUCM11_SIP_SERVER2 ipAddress 10.35.180.110 ipPort 5060 defaultForIp false
set addressContext default zone ZONE-INT-CUCM ipPeer CUCM11_SIP_SERVER2 authentication incInternalCredentials enabled
set addressContext default zone ZONE-INT-CUCM ipPeer CUCM11_SIP_SERVER2 authentication intChallengeResponse enabled
set addressContext default zone ZONE-INT-CUCM ipPeer CUCM11_SIP_SERVER2 surrogateRegistration userPart 4696806538 authUserName xxxxxx-4696806538 regAuthPassword xxxx state enabled sendCredentials 
challengeForAnyMessageAndInDialogRequests hostPart voip.centurylink.com
commit

#SIP trunk group
set addressContext default zone ZONE-INT-VoIP sipTrunkGroup CUCM11_TG media mediaIpInterfaceGroupName Private sourceAddressFiltering disabled
set addressContext default zone ZONE-INT-VoIP sipTrunkGroup CUCM11_TG signaling messageManipulation inputAdapterProfile CenturyLink_HashIn includeAppHdrs disabled
set addressContext default zone ZONE-INT-VoIP sipTrunkGroup CUCM11_TG signaling rel100Support enabled acceptHistoryInfo enabled
set addressContext default zone ZONE-INT-VoIP sipTrunkGroup CUCM11_TG ingressIpPrefix 10.35.180.110 32
set addressContext default zone ZONE-INT-VoIP sipTrunkGroup CUCM11_TG policy callRouting elementRoutingPriority ERP_CenturyLink
set addressContext default zone ZONE-INT-VoIP sipTrunkGroup CUCM11_TG policy digitParameterHandling numberingPlan NANP_ACCESS
set addressContext default zone ZONE-INT-VoIP sipTrunkGroup CUCM11_TG policy media packetServiceProfile CUCM11_PSP
set addressContext default zone ZONE-INT-VoIP sipTrunkGroup CUCM11_TG policy services classOfService DEFAULT_IP
set addressContext default zone ZONE-INT-VoIP sipTrunkGroup CUCM11_TG policy signaling ipSignalingProfile CUCM11_IPSP signalingProfile CUCM11_SP
set addressContext default zone ZONE-INT-VoIP sipTrunkGroup CUCM11_TG services transparencyProfile CUCM11_TP
set addressContext default zone ZONE-INT-VoIP sipTrunkGroup CUCM11_TG signaling E164Profiles e164LocalProfile CenturyLink_E164 e164GlobalProfile CenturyLink_E164
set addressContext default zone ZONE-INT-VoIP sipTrunkGroup CUCM11_TG signaling relayNonInviteRequest enabled
set addressContext default zone ZONE-INT-VoIP sipTrunkGroup CUCM11_TG signaling authentication intChallengeResponse enabled
set addressContext default zone ZONE-INT-VoIP sipTrunkGroup CUCM11_TG state enabled mode inService
commit

set addressContext default zone ZONE-INT-CUCM sipTrunkGroup CUCM11_TG2 media mediaIpInterfaceGroupName Private sourceAddressFiltering disabled
set addressContext default zone ZONE-INT-CUCM sipTrunkGroup CUCM11_TG2 signaling messageManipulation inputAdapterProfile CenturyLink_HashIn includeAppHdrs disabled
set addressContext default zone ZONE-INT-CUCM sipTrunkGroup CUCM11_TG2 signaling rel100Support enabled acceptHistoryInfo enabled
set addressContext default zone ZONE-INT-CUCM sipTrunkGroup CUCM11_TG2 ingressIpPrefix 10.35.180.110 32
set addressContext default zone ZONE-INT-CUCM sipTrunkGroup CUCM11_TG2 policy callRouting elementRoutingPriority ERP_CenturyLink
set addressContext default zone ZONE-INT-CUCM sipTrunkGroup CUCM11_TG2 policy digitParameterHandling numberingPlan NANP_ACCESS
set addressContext default zone ZONE-INT-CUCM sipTrunkGroup CUCM11_TG2 policy media packetServiceProfile CUCM11_PSP
set addressContext default zone ZONE-INT-CUCM sipTrunkGroup CUCM11_TG2 policy services classOfService DEFAULT_IP
set addressContext default zone ZONE-INT-CUCM sipTrunkGroup CUCM11_TG2 policy signaling ipSignalingProfile CUCM11_IPSP signalingProfile CUCM11_SP
set addressContext default zone ZONE-INT-CUCM sipTrunkGroup CUCM11_TG2 services transparencyProfile CUCM11_TP
set addressContext default zone ZONE-INT-CUCM sipTrunkGroup CUCM11_TG2 signaling E164Profiles e164LocalProfile CenturyLink_E164 e164GlobalProfile CenturyLink_E164
set addressContext default zone ZONE-INT-CUCM sipTrunkGroup CUCM11_TG2 signaling relayNonInviteRequest enabled
set addressContext default zone ZONE-INT-VoIP sipTrunkGroup CUCM11_TG2 signaling authentication intChallengeResponse enabled
set addressContext default zone ZONE-INT-CUCM sipTrunkGroup CUCM11_TG2 state enabled mode inService
commit


#SBC Configuration for Fax Trunk
#Packet Service Profile
set profiles media packetServiceProfile TENOR_FAX_PSP codec codecEntry1 G711u_Tenor_T38 codecEntry2 G729_Tenor_T38
set profiles media packetServiceProfile TENOR_FAX_PSP packetToPacketControl codecsAllowedForTranscoding thisLeg g711u,g729,t38 otherLeg g711u,g729,t38
set profiles media packetServiceProfile TENOR_FAX_PSP packetToPacketControl transcode only
set profiles media packetServiceProfile TENOR_FAX_PSP rtcpOptions rtcp enable
commit

#IP signaling profile
set profiles signaling ipSignalingProfile TENOR_FAX_IPSP ipProtocolType sipOnly
set profiles signaling ipSignalingProfile TENOR_FAX_IPSP commonIpAttributes flags disableMediaLockDown enable includeReasonHeader enable
set profiles signaling ipSignalingProfile TENOR_FAX_IPSP commonIpAttributes flags minimizeRelayingOfMediaChangesFromOtherCallLegAll enable publishIPInHoldSDP enable sendPtimeInSdp enable
set profiles signaling ipSignalingProfile TENOR_FAX_IPSP commonIpAttributes flags sendRtcpPortInSdp enable
set profiles signaling ipSignalingProfile TENOR_FAX_IPSP commonIpAttributes relayFlags notify enable
set profiles signaling ipSignalingProfile TENOR_FAX_IPSP commonIpAttributes transparencyFlags  mwiBody enable  unknownBody enable unknownHeader enable
set profiles signaling ipSignalingProfile TENOR_FAX_IPSP ingressIpAttributes flags sendSdpIn200OkIf18xReliable enable sendSdpInSubsequent18x enable
set profiles signaling ipSignalingProfile TENOR_FAX_IPSP egressIpAttributes flags disable2806Compliance enable
set profiles signaling ipSignalingProfile TENOR_FAX_IPSP egressIpAttributes transport type1 udp type2 none type3 none type4 none
commit

#Signaling profiles
set profiles signaling signalingProfile TENOR_FAX_SP egress egressFlags addPrefix011ForInternationalCalls enable convertNumbersToE164Format enable
commit

#IP peer
set addressContext default zone ZONE-INT-VoIP ipPeer TENOR_SIP_SERVER policy sip fqdnPort 0
set addressContext default zone ZONE-INT-VoIP ipPeer TENOR_SIP_SERVER ipAddress 10.35.137.43 ipPort 5084 defaultForIp false
set addressContext default zone ZONE-INT-VoIP ipPeer TENOR_SIP_SERVER authentication incInternalCredentials enabled
set addressContext default zone ZONE-INT-VoIP ipPeer TENOR_SIP_SERVER authentication intChallengeResponse enabled
set addressContext default zone ZONE-INT-VoIP ipPeer TENOR_SIP_SERVER surrogateRegistration userPart 4696806537 authUserName xxxxxxx-4696806537 regAuthPassword xxxx state disabled sendCredentials 
challengeForAnyMessageAndInDialogRequests hostPart voip.centurylink.com
commit

#SIP trunk group
set addressContext default zone ZONE-INT-VoIP sipTrunkGroup TENOR_TG_FAX media mediaIpInterfaceGroupName Private
set addressContext default zone ZONE-INT-VoIP sipTrunkGroup TENOR_TG_FAX signaling messageManipulation inputAdapterProfile CenturyLink_HashIn includeAppHdrs disabled
set addressContext default zone ZONE-INT-VoIP sipTrunkGroup TENOR_TG_FAX signaling rel100Support enabled acceptHistoryInfo enabled
set addressContext default zone ZONE-INT-VoIP sipTrunkGroup TENOR_TG_FAX ingressIpPrefix 10.35.137.43 32
set addressContext default zone ZONE-INT-VoIP sipTrunkGroup TENOR_TG_FAX policy callRouting elementRoutingPriority ERP_CenturyLink
set addressContext default zone ZONE-INT-VoIP sipTrunkGroup TENOR_TG_FAX policy digitParameterHandling numberingPlan CenturyLink_NANP
set addressContext default zone ZONE-INT-VoIP sipTrunkGroup TENOR_TG_FAX policy media packetServiceProfile TENOR_FAX_PSP
set addressContext default zone ZONE-INT-VoIP sipTrunkGroup TENOR_TG_FAX policy services classOfService DEFAULT_IP
set addressContext default zone ZONE-INT-VoIP sipTrunkGroup TENOR_TG_FAX policy signaling ipSignalingProfile TENOR_FAX_IPSP signalingProfile TENOR_FAX_SP
set addressContext default zone ZONE-INT-VoIP sipTrunkGroup TENOR_TG_FAX services transparencyProfile CenturyLink
set addressContext default zone ZONE-INT-VoIP sipTrunkGroup TENOR_TG_FAX state enabled mode inService
commit


#External Side SBC Configuration

#IP Interface Group
set addressContext default zone ZONE-EXT-ACCESS sipSigPort 10 ipInterfaceGroupName Public ipAddressV4 216.110.2.227 portNumber 5060 mode outOfService state disabled transportProtocolsAllowed sip-udp sip-tcp 
set addressContext default zone ZONE-EXT-ACCESS sipSigPort 10 mode inService state enabled
commit
 
set addressContext default ipInterfaceGroup Public ipInterface Public-pkt2 ceName SWECENTURYLINK01 portName pkt2 ipAddress 216.110.2.226 prefix 28 mode outOfService state disabled
set addressContext default ipInterfaceGroup Public ipInterface Public-pkt2 mode inService state enabled
commit

#IP static route
set addressContext default staticRoute 216.206.64.0 24 216.110.2.225 Public Public-pkt2 preference 100
commit

#SBC Configuration for CenturyLink SIP Trunk

#Packet Service Profile (PSP)
set profiles media packetServiceProfile CenturyLink_PSP packetToPacketControl codecsAllowedForTranscoding thisLeg g711u,g729,t38 otherLeg g711u,g729,t38
set profiles media packetServiceProfile CenturyLink_PSP packetToPacketControl transcode conditional
set profiles media packetServiceProfile CenturyLink_PSP rtcpOptions rtcp enable terminationForPassthrough enable
set profiles media packetServiceProfile CenturyLink_PSP silenceInsertionDescriptor g711SidRtpPayloadType 13 heartbeat enable
set profiles media packetServiceProfile CenturyLink_PSP typeOfService 160
commit

#IP Signaling profiles(IPSP)
set profiles signaling ipSignalingProfile CenturyLink_IPSP ipProtocolType sipOnly
set profiles signaling ipSignalingProfile CenturyLink_IPSP commonIpAttributes flags disableMediaLockDown enable includeReasonHeader enable
set profiles signaling ipSignalingProfile CenturyLink_IPSP commonIpAttributes flags minimizeRelayingOfMediaChangesFromOtherCallLegAll enable publishIPInHoldSDP enable sendPtimeInSdp enable
set profiles signaling ipSignalingProfile CenturyLink_IPSP commonIpAttributes flags sendRtcpPortInSdp enable
set profiles signaling ipSignalingProfile CenturyLink_IPSP commonIpAttributes relayFlags notify enable
set profiles signaling ipSignalingProfile CenturyLink_IPSP commonIpAttributes transparencyFlags  mwiBody enable  unknownBody enable unknownHeader enable
set profiles signaling ipSignalingProfile CenturyLink_IPSP ingressIpAttributes flags sendSdpIn200OkIf18xReliable enable sendSdpInSubsequent18x enable
set profiles signaling ipSignalingProfile CenturyLink_IPSP egressIpAttributes flags disable2806Compliance enable
set profiles signaling ipSignalingProfile CenturyLink_IPSP egressIpAttributes transport type1 udp type2 none type3 none type4 none
set profiles signaling ipSignalingProfile CenturyLink_IPSP egressIpAttributes privacy privacyInformation pAssertedId
commit

#Signaling profiles(SP)
set profiles signaling signalingProfile CenturyLink_SP egress egressFlags addPrefix011ForInternationalCalls enable convertNumbersToE164Format enable
commit

#Transparency profile(TP)
set profiles services transparencyProfile CenturyLink_TP sipMessageBody all excludedMethods register invite subscribe info publish message refer options update
set profiles services transparencyProfile CenturyLink_TP state enabled
commit

#DM/PM rule
set profiles digitParameterHandling dmPmCriteria CenturyLink_HashPrefix criteriaType digit digitType calledNumber parameterPresenceCheck exists
set profiles digitParameterHandling dmPmCriteria CenturyLink_HashPrefix digitCriteria digitMatch value startDigitPosition 0 numberOfDigits 3 matchValue 999
set profiles digitParameterHandling dmPmCriteria CenturyLink_HashPrefix digitCriteria digitMatch operation equals
set profiles digitParameterHandling dmPmCriteria CenturyLink_HashPrefix digitCriteria egressFlag value send operation ignore
set profiles digitParameterHandling dmPmCriteria CenturyLink_HashPrefix digitCriteria natureOfAddress value 950 operation ignore
set profiles digitParameterHandling dmPmCriteria CenturyLink_HashPrefix digitCriteria numberLength value 5 operation equals
commit

set profiles digitParameterHandling dmPmRule CenturyLink_Hash_Add subRule 0 criteria CenturyLink_HashPrefix ruleType digit
set profiles digitParameterHandling dmPmRule CenturyLink_Hash_Add subRule 0 digitManipulation digitStringManipulation replacement type constant digitString calledNumber startDigitPosition 0 numberOfDigits 1 value #
set profiles digitParameterHandling dmPmRule CenturyLink_Hash_Add subRule 0 digitManipulation digitStringManipulation startDigitPosition 0 numberOfDigits 3 action none
set profiles digitParameterHandling dmPmRule CenturyLink_Hash_Add subRule 0 digitManipulation numberParameterManipulation natureOfAddress none numberingPlanIndicator none numberLength 3 presentation none screening none includeInEgress none
set profiles digitParameterHandling dmPmRule CenturyLink_Hash_Add subRule 0 digitManipulation numberType calledNumber
commit

#Another entry for a default Prefix Profile 
set profiles digitParameterHandling prefixProfile NA_DIAL_PLAN entry * 0 1 15 callType nationalType numberLeadingPrefixDigits 0 numberLeadingPrefixDigitsToStrip 0 applyDmRule disable determineArea disabl

#Zone
set addressContext default zone ZONE-EXT-ACCESS id 20
commit

#SIP signaling port
set addressContext default zone ZONE-EXT-ACCESS sipSigPort 10 ipInterfaceGroupName Public ipAddressV4 216.110.2.227 portNumber 5060 mode outOfService state disabled transportProtocolsAllowed sip-udp
set addressContext default zone ZONE-EXT-ACCESS sipSigPort 10 dscpValue 24
set addressContext default zone ZONE-EXT-ACCESS sipSigPort 10 mode inService state enabled
commit

#IP peer
set addressContext default zone ZONE-EXT-ACCESS ipPeer CenturyLink_IPP policy sip fqdnPort 0
set addressContext default zone ZONE-EXT-ACCESS ipPeer CenturyLink_IPP ipAddress 216.206.64.91 ipPort 5100 defaultForIp false
set addressContext default zone ZONE-EXT-ACCESS ipPeer CenturyLink_IPP2 policy sip fqdnPort 0
set addressContext default zone ZONE-EXT-ACCESS ipPeer CenturyLink_IPP2 ipAddress 216.206.66.91 ipPort 5100 defaultForIp false
commit

#SIP trunk group
set addressContext default zone ZONE-EXT-ACCESS sipTrunkGroup CenturyLink_TG media mediaIpInterfaceGroupName Public 
set addressContext default zone ZONE-EXT-ACCESS sipTrunkGroup CenturyLink_TG signaling messageManipulation outputAdapterProfile CenturyLink_SMM includeAppHdrs disabled
set addressContext default zone ZONE-EXT-ACCESS sipTrunkGroup CenturyLink_TG signaling rel100Support enabled acceptHistoryInfo enabled
set addressContext default zone ZONE-EXT-ACCESS sipTrunkGroup CenturyLink_TG ingressIpPrefix 216.206.64.91 32
set addressContext default zone ZONE-EXT-ACCESS sipTrunkGroup CenturyLink_TG policy callRouting elementRoutingPriority ERP_CenturyLink
set addressContext default zone ZONE-EXT-ACCESS sipTrunkGroup CenturyLink_TG policy digitParameterHandling numberingPlan NANP_ACCESS egressDmPmRule CenturyLink_Hash_Add
set addressContext default zone ZONE-EXT-ACCESS sipTrunkGroup CenturyLink_TG policy media packetServiceProfile CenturyLink_PSP
set addressContext default zone ZONE-EXT-ACCESS sipTrunkGroup CenturyLink_TG policy services classOfService DEFAULT_IP
set addressContext default zone ZONE-EXT-ACCESS sipTrunkGroup CenturyLink_TG policy signaling ipSignalingProfile CenturyLink_IPSP signalingProfile CenturyLink_SP
set addressContext default zone ZONE-EXT-ACCESS sipTrunkGroup CenturyLink_TG services transparencyProfile CenturyLink_TP
set addressContext default zone ZONE-EXT-ACCESS sipTrunkGroup CenturyLink_TG signaling relayNonInviteRequest enabled
set addressContext default zone ZONE-EXT-ACCESS sipTrunkGroup CenturyLink_TG signaling authentication authUserPart xxxxxxx-4696806537 authPassword xxxx intChallengeResponse enabled
set addressContext default zone ZONE-EXT-ACCESS sipTrunkGroup CenturyLink_TG state enabled mode inService
commit

set addressContext default zone ZONE-EXT-ACCESS sipTrunkGroup CenturyLink_TG2 media mediaIpInterfaceGroupName Public 
set addressContext default zone ZONE-EXT-ACCESS sipTrunkGroup CenturyLink_TG2 signaling messageManipulation outputAdapterProfile CenturyLink_SMM2 includeAppHdrs disabled
set addressContext default zone ZONE-EXT-ACCESS sipTrunkGroup CenturyLink_TG2 signaling rel100Support enabled acceptHistoryInfo enabled
set addressContext default zone ZONE-EXT-ACCESS sipTrunkGroup CenturyLink_TG2 ingressIpPrefix 216.206.66.91 32
set addressContext default zone ZONE-EXT-ACCESS sipTrunkGroup CenturyLink_TG2 policy callRouting elementRoutingPriority ERP_CenturyLink
set addressContext default zone ZONE-EXT-ACCESS sipTrunkGroup CenturyLink_TG2 policy digitParameterHandling numberingPlan NANP_ACCESS egressDmPmRule CenturyLink_Hash_Add
set addressContext default zone ZONE-EXT-ACCESS sipTrunkGroup CenturyLink_TG2 policy media packetServiceProfile CenturyLink_PSP
set addressContext default zone ZONE-EXT-ACCESS sipTrunkGroup CenturyLink_TG2 policy services classOfService DEFAULT_IP
set addressContext default zone ZONE-EXT-ACCESS sipTrunkGroup CenturyLink_TG2 policy signaling ipSignalingProfile CenturyLink_IPSP signalingProfile CenturyLink_SP
set addressContext default zone ZONE-EXT-ACCESS sipTrunkGroup CenturyLink_TG2 services transparencyProfile CenturyLink_TP
set addressContext default zone ZONE-EXT-ACCESS sipTrunkGroup CenturyLink_TG2 signaling relayNonInviteRequest enabled
set addressContext default zone ZONE-EXT-ACCESS sipTrunkGroup CenturyLink_TG2 signaling authentication authUserPart xxxxxxx-4696806538 authPassword xxxx intChallengeResponse enabled
set addressContext default zone ZONE-EXT-ACCESS sipTrunkGroup CenturyLink_TG2 state enabled mode inService
commit
 
#Global Call Routing Configuration
#Element Routing Priority
set profiles callRouting elementRoutingPriority ERP_CenturyLink
set profiles callRouting elementRoutingPriority ERP_CenturyLink entry nationalType 2 entityType none
set profiles callRouting elementRoutingPriority ERP_CenturyLink entry userName 2 entityType none
set profiles callRouting elementRoutingPriority ERP_CenturyLink entry internationalType 2 entityType none
set profiles callRouting elementRoutingPriority ERP_CenturyLink entry nationalType 1 entityType trunkGroup
set profiles callRouting elementRoutingPriority ERP_CenturyLink entry internationalType 1 entityType trunkGroup
set profiles callRouting elementRoutingPriority ERP_CenturyLink entry userName 1 entityType trunkGroup
commit
 
#CUCM11 Routing
set global callRouting routingLabel TO_CUCM11_TG routingLabelRoute 0 trunkGroup CUCM11_TG ipPeer CUCM11_SIP_SERVER proportion 100 cost 100 inService inService testing normal
set global callRouting routingLabel TO_CUCM11_TG overflowNOA none overflowNPI none routePrioritizationType sequence action routes numRoutesPerCall 10
set global callRouting routingLabel TO_CUCM11_TG2 routingLabelRoute 0 trunkGroup CUCM11_TG2 ipPeer CUCM11_SIP_SERVER2 proportion 100 cost 100 inService inService testing normal
set global callRouting routingLabel TO_CUCM11_TG2 overflowNOA none overflowNPI none routePrioritizationType sequence action routes numRoutesPerCall 10
set global callRouting route trunkGroup CUCM11_TG SWECENTURYLINK01 standard Sonus_NULL Sonus_NULL all all ALL none Sonus_NULL routingLabel TO_CenturyLink
set global callRouting route trunkGroup CUCM11_TG2 SWECENTURYLINK01 standard Sonus_NULL Sonus_NULL all all ALL none Sonus_NULL routingLabel TO_CenturyLink2
commit

#CenturyLink routing
set global callRouting routingLabel TO_CenturyLink routingLabelRoute 0 trunkGroup CenturyLink_TG ipPeer CenturyLink_IPP proportion 100 cost 100 inService inService testing normal
set global callRouting routingLabel TO_CenturyLink overflowNOA none overflowNPI none routePrioritizationType proportionAllocation action routes numRoutesPerCall 10
set global callRouting routingLabel TO_CenturyLink2 routingLabelRoute 0 trunkGroup CenturyLink_TG2 ipPeer CenturyLink_IPP2 proportion 100 cost 100 inService inService testing normal
set global callRouting routingLabel TO_CenturyLink2 overflowNOA none overflowNPI none routePrioritizationType proportionAllocation action routes numRoutesPerCall 10
set global callRouting route trunkGroup CenturyLink_TG SWECENTURYLINK01 standard Sonus_NULL Sonus_NULL all all ALL none Sonus_NULL routingLabel TO_CUCM11_TG
set global callRouting route trunkGroup CenturyLink_TG2 SWECENTURYLINK01 standard Sonus_NULL Sonus_NULL all all ALL none Sonus_NULL routingLabel TO_CUCM11_TG2 
commit

set global callRouting routingLabel TO_TENOR routingLabelRoute 0 trunkGroup TENOR_TG_FAX ipPeer TENOR_SIP_SERVER proportion 100 cost 100 inService inService testing normal
set global callRouting routingLabel TO_TENOR  overflowNOA none overflowNPI none routePrioritizationType sequence action routes numRoutesPerCall 10
set global callRouting route trunkGroup CenturyLink_TG SWECENTURYLINK01 standard 4696806542 +1 all all ALL none Sonus_NULL routingLabel TO_TENOR

#SMM
set profiles signaling sipAdaptorProfile CenturyLink_HashIn rule 1 criterion 1 type message
set profiles signaling sipAdaptorProfile CenturyLink_HashIn rule 1 criterion 1 message messageTypes all
set profiles signaling sipAdaptorProfile CenturyLink_HashIn rule 1 applyMatchHeader one
set profiles signaling sipAdaptorProfile CenturyLink_HashIn rule 1 criterion 2 type header
set profiles signaling sipAdaptorProfile CenturyLink_HashIn rule 1 criterion 2 header name To condition exist
set profiles signaling sipAdaptorProfile CenturyLink_HashIn rule 1 action 1 type header operation regsub
set profiles signaling sipAdaptorProfile CenturyLink_HashIn rule 1 action 1 to type header value To
set profiles signaling sipAdaptorProfile CenturyLink_HashIn rule 1 action 1 regexp string %23 matchInstance all
set profiles signaling sipAdaptorProfile CenturyLink_HashIn rule 1 action 1 from type value value 999
set profiles signaling sipAdaptorProfile CenturyLink_HashIn rule 2 criterion 2 type header
set profiles signaling sipAdaptorProfile CenturyLink_HashIn rule 2 criterion 2 type header
set profiles signaling sipAdaptorProfile CenturyLink_HashIn rule 2 criterion 2 header name Request-Line condition exist
set profiles signaling sipAdaptorProfile CenturyLink_HashIn rule 2 criterion 1 type message
set profiles signaling sipAdaptorProfile CenturyLink_HashIn rule 2 criterion 1 message messageTypes all
set profiles signaling sipAdaptorProfile CenturyLink_HashIn rule 2 applyMatchHeader one
set profiles signaling sipAdaptorProfile CenturyLink_HashIn rule 2 action 1 type header operation regsub
set profiles signaling sipAdaptorProfile CenturyLink_HashIn rule 2 action 1 to type header value Request-Line
set profiles signaling sipAdaptorProfile CenturyLink_HashIn rule 2 action 1 regexp string %23
set profiles signaling sipAdaptorProfile CenturyLink_HashIn rule 2 action 1 from type value value 999
set profiles signaling sipAdaptorProfile CenturyLink_HashIn state enable
commit

set profiles signaling sipAdaptorProfile CenturyLink_SMM rule 1 criterion 1 type message
set profiles signaling sipAdaptorProfile CenturyLink_SMM rule 1 criterion 1 message messageTypes requestAll
set profiles signaling sipAdaptorProfile CenturyLink_SMM rule 1 criterion 2 type header
set profiles signaling sipAdaptorProfile CenturyLink_SMM rule 1 criterion 2 header name P-Asserted-Identity condition exist
set profiles signaling sipAdaptorProfile CenturyLink_SMM rule 1 applyMatchHeader one
set profiles signaling sipAdaptorProfile CenturyLink_SMM rule 1 action 1 type header operation delete
set profiles signaling sipAdaptorProfile CenturyLink_SMM rule 1 action 1 to type header value P-Asserted-Identity
set profiles signaling sipAdaptorProfile CenturyLink_SMM rule 2 criterion 1 type message
set profiles signaling sipAdaptorProfile CenturyLink_SMM rule 2 criterion 1 message messageTypes requestAll
set profiles signaling sipAdaptorProfile CenturyLink_SMM rule 2 criterion 2 type header
set profiles signaling sipAdaptorProfile CenturyLink_SMM rule 2 criterion 2 header name P-Asserted-Identity condition absent
set profiles signaling sipAdaptorProfile CenturyLink_SMM rule 2 applyMatchHeader one
set profiles signaling sipAdaptorProfile CenturyLink_SMM rule 2 action 1 type header operation add headerPosition last
set profiles signaling sipAdaptorProfile CenturyLink_SMM rule 2 action 1 to type header value P-Asserted-Identity
set profiles signaling sipAdaptorProfile CenturyLink_SMM rule 2 action 1 from type value value <sip:4696806537@voip.centurylink.com>
set profiles signaling sipAdaptorProfile CenturyLink_SMM rule 3 criterion 1 type message
set profiles signaling sipAdaptorProfile CenturyLink_SMM rule 3 criterion 1 message messageTypes request methodTypes register
set profiles signaling sipAdaptorProfile CenturyLink_SMM rule 3 criterion 2 type header
set profiles signaling sipAdaptorProfile CenturyLink_SMM rule 3 criterion 2 header name Request-Line condition exist
set profiles signaling sipAdaptorProfile CenturyLink_SMM rule 3 criterion 3 type token
set profiles signaling sipAdaptorProfile CenturyLink_SMM rule 3 criterion 3 token condition exist tokenType urihostname
set profiles signaling sipAdaptorProfile CenturyLink_SMM rule 3 applyMatchHeader one
set profiles signaling sipAdaptorProfile CenturyLink_SMM rule 3 action 1 type token operation regsub
set profiles signaling sipAdaptorProfile CenturyLink_SMM rule 3 action 1 to type token tokenValue urihostname
set profiles signaling sipAdaptorProfile CenturyLink_SMM rule 3 action 1 regexp string voip.centurylink.com matchInstance all
set profiles signaling sipAdaptorProfile CenturyLink_SMM rule 3 action 1 from type value value 216.206.64.91
set profiles signaling sipAdaptorProfile CenturyLink_SMM rule 4 criterion 1 type message
set profiles signaling sipAdaptorProfile CenturyLink_SMM rule 4 criterion 1 message messageTypes all
set profiles signaling sipAdaptorProfile CenturyLink_SMM rule 4 criterion 2 type header
set profiles signaling sipAdaptorProfile CenturyLink_SMM rule 4 criterion 2 header name From condition exist
set profiles signaling sipAdaptorProfile CenturyLink_SMM rule 4 applyMatchHeader one
set profiles signaling sipAdaptorProfile CenturyLink_SMM rule 4 action 1 type header operation regsub
set profiles signaling sipAdaptorProfile CenturyLink_SMM rule 4 action 1 to type header value From
set profiles signaling sipAdaptorProfile CenturyLink_SMM rule 4 action 1 regexp string anonymous@216.110.2.227 matchInstance all
set profiles signaling sipAdaptorProfile CenturyLink_SMM rule 4 action 1 from type value value anonymous@anonymous.invalid
set profiles signaling sipAdaptorProfile CenturyLink_SMM rule 4 action 2 type header operation regsub
set profiles signaling sipAdaptorProfile CenturyLink_SMM rule 4 action 2 to type header value From
set profiles signaling sipAdaptorProfile CenturyLink_SMM rule 4 action 2 regexp string Restricted@216.110.2.227 matchInstance all
set profiles signaling sipAdaptorProfile CenturyLink_SMM rule 4 action 2 from type value value anonymous@anonymous.invalid
set profiles signaling sipAdaptorProfile CenturyLink_SMM rule 5 criterion 1 type message
set profiles signaling sipAdaptorProfile CenturyLink_SMM rule 5 criterion 1 message messageTypes all
set profiles signaling sipAdaptorProfile CenturyLink_SMM rule 5 criterion 2 type header
set profiles signaling sipAdaptorProfile CenturyLink_SMM rule 5 criterion 2 header name Contact condition exist
set profiles signaling sipAdaptorProfile CenturyLink_SMM rule 5 action 1 type header operation regsub
set profiles signaling sipAdaptorProfile CenturyLink_SMM rule 5 action 1 to type header value Contact
set profiles signaling sipAdaptorProfile CenturyLink_SMM rule 5 action 1 regexp string sip:anonymous@216.110.2.227:5060 matchInstance all
set profiles signaling sipAdaptorProfile CenturyLink_SMM rule 5 action 1 from type value value sip:anonymous@anonymous.invalid:5060
set profiles signaling sipAdaptorProfile CenturyLink_SMM rule 5 action 2 type header operation regsub
set profiles signaling sipAdaptorProfile CenturyLink_SMM rule 5 action 2 to type header value Contact
set profiles signaling sipAdaptorProfile CenturyLink_SMM rule 5 action 2 regexp string sip:Restricted@216.110.2.227:5060 matchInstance all
set profiles signaling sipAdaptorProfile CenturyLink_SMM rule 5 action 2 from type value value sip:anonymous@anonymous.invalid:5060
set profiles signaling sipAdaptorProfile CenturyLink_SMM rule 6 criterion 1 type message
set profiles signaling sipAdaptorProfile CenturyLink_SMM rule 6 criterion 1 message messageTypes requestAll
set profiles signaling sipAdaptorProfile CenturyLink_SMM rule 6 criterion 2 type header
set profiles signaling sipAdaptorProfile CenturyLink_SMM rule 6 criterion 2 header name From condition exist
set profiles signaling sipAdaptorProfile CenturyLink_SMM rule 6 applyMatchHeader one
set profiles signaling sipAdaptorProfile CenturyLink_SMM rule 6 action 1 type header operation regsub
set profiles signaling sipAdaptorProfile CenturyLink_SMM rule 6 action 1 to type header value From
set profiles signaling sipAdaptorProfile CenturyLink_SMM rule 6 action 1 regexp string 216.110.2.227 matchInstance all
set profiles signaling sipAdaptorProfile CenturyLink_SMM rule 6 action 1 from type value value voip.centurylink.com
set profiles signaling sipAdaptorProfile CenturyLink_SMM state enabled
commit
set profiles signaling sipAdaptorProfile CenturyLink_SMM2 rule 1 criterion 1 type message
set profiles signaling sipAdaptorProfile CenturyLink_SMM2 rule 1 criterion 1 message messageTypes requestAll
set profiles signaling sipAdaptorProfile CenturyLink_SMM2 rule 1 criterion 2 type header
set profiles signaling sipAdaptorProfile CenturyLink_SMM2 rule 1 criterion 2 header name P-Asserted-Identity condition exist
set profiles signaling sipAdaptorProfile CenturyLink_SMM2 rule 1 applyMatchHeader one
set profiles signaling sipAdaptorProfile CenturyLink_SMM2 rule 1 action 1 type header operation delete
set profiles signaling sipAdaptorProfile CenturyLink_SMM2 rule 1 action 1 to type header value P-Asserted-Identity
set profiles signaling sipAdaptorProfile CenturyLink_SMM2 rule 2 criterion 1 type message
set profiles signaling sipAdaptorProfile CenturyLink_SMM2 rule 2 criterion 1 message messageTypes requestAll
set profiles signaling sipAdaptorProfile CenturyLink_SMM2 rule 2 criterion 2 type header
set profiles signaling sipAdaptorProfile CenturyLink_SMM2 rule 2 criterion 2 header name P-Asserted-Identity condition absent
set profiles signaling sipAdaptorProfile CenturyLink_SMM2 rule 2 applyMatchHeader one
set profiles signaling sipAdaptorProfile CenturyLink_SMM2 rule 2 action 1 type header operation add headerPosition last
set profiles signaling sipAdaptorProfile CenturyLink_SMM2 rule 2 action 1 to type header value P-Asserted-Identity
set profiles signaling sipAdaptorProfile CenturyLink_SMM2 rule 2 action 1 from type value value <sip:4696806538@voip.centurylink.com>
set profiles signaling sipAdaptorProfile CenturyLink_SMM2 rule 3 criterion 1 type message
set profiles signaling sipAdaptorProfile CenturyLink_SMM2 rule 3 criterion 1 message messageTypes request methodTypes register
set profiles signaling sipAdaptorProfile CenturyLink_SMM2 rule 3 criterion 2 type header
set profiles signaling sipAdaptorProfile CenturyLink_SMM2 rule 3 criterion 2 header name Request-Line condition exist
set profiles signaling sipAdaptorProfile CenturyLink_SMM2 rule 3 criterion 3 type token
set profiles signaling sipAdaptorProfile CenturyLink_SMM2 rule 3 criterion 3 token condition exist tokenType urihostname
set profiles signaling sipAdaptorProfile CenturyLink_SMM2 rule 3 applyMatchHeader one
set profiles signaling sipAdaptorProfile CenturyLink_SMM2 rule 3 action 1 type token operation regsub
set profiles signaling sipAdaptorProfile CenturyLink_SMM2 rule 3 action 1 to type token tokenValue urihostname
set profiles signaling sipAdaptorProfile CenturyLink_SMM2 rule 3 action 1 regexp string voip.centurylink.com matchInstance all
set profiles signaling sipAdaptorProfile CenturyLink_SMM2 rule 3 action 1 from type value value 216.206.66.91
set profiles signaling sipAdaptorProfile CenturyLink_SMM2 rule 4 criterion 1 type message
set profiles signaling sipAdaptorProfile CenturyLink_SMM2 rule 4 criterion 1 message messageTypes all
set profiles signaling sipAdaptorProfile CenturyLink_SMM2 rule 4 criterion 2 type header
set profiles signaling sipAdaptorProfile CenturyLink_SMM2 rule 4 criterion 2 header name From condition exist
set profiles signaling sipAdaptorProfile CenturyLink_SMM2 rule 4 applyMatchHeader one
set profiles signaling sipAdaptorProfile CenturyLink_SMM2 rule 4 action 1 type header operation regsub
set profiles signaling sipAdaptorProfile CenturyLink_SMM2 rule 4 action 1 to type header value From
set profiles signaling sipAdaptorProfile CenturyLink_SMM2 rule 4 action 1 regexp string anonymous@216.110.2.227 matchInstance all
set profiles signaling sipAdaptorProfile CenturyLink_SMM2 rule 4 action 1 from type value value anonymous@anonymous.invalid
set profiles signaling sipAdaptorProfile CenturyLink_SMM2 rule 4 action 2 type header operation regsub
set profiles signaling sipAdaptorProfile CenturyLink_SMM2 rule 4 action 2 to type header value From
set profiles signaling sipAdaptorProfile CenturyLink_SMM2 rule 4 action 2 regexp string Restricted@216.110.2.227 matchInstance all
set profiles signaling sipAdaptorProfile CenturyLink_SMM2 rule 4 action 2 from type value value anonymous@anonymous.invalid
set profiles signaling sipAdaptorProfile CenturyLink_SMM2 rule 5 criterion 1 type message
set profiles signaling sipAdaptorProfile CenturyLink_SMM2 rule 5 criterion 1 message messageTypes all
set profiles signaling sipAdaptorProfile CenturyLink_SMM2 rule 5 criterion 2 type header
set profiles signaling sipAdaptorProfile CenturyLink_SMM2 rule 5 criterion 2 header name Contact condition exist
set profiles signaling sipAdaptorProfile CenturyLink_SMM2 rule 5 action 1 type header operation regsub
set profiles signaling sipAdaptorProfile CenturyLink_SMM2 rule 5 action 1 to type header value Contact
set profiles signaling sipAdaptorProfile CenturyLink_SMM2 rule 5 action 1 regexp string sip:anonymous@216.110.2.227:5060 matchInstance all
set profiles signaling sipAdaptorProfile CenturyLink_SMM2 rule 5 action 1 from type value value sip:anonymous@anonymous.invalid:5060
set profiles signaling sipAdaptorProfile CenturyLink_SMM2 rule 5 action 2 type header operation regsub
set profiles signaling sipAdaptorProfile CenturyLink_SMM2 rule 5 action 2 to type header value Contact
set profiles signaling sipAdaptorProfile CenturyLink_SMM2 rule 5 action 2 regexp string sip:Restricted@216.110.2.227:5060 matchInstance all
set profiles signaling sipAdaptorProfile CenturyLink_SMM2 rule 5 action 2 from type value value sip:anonymous@anonymous.invalid:5060
set profiles signaling sipAdaptorProfile CenturyLink_SMM2 rule 6 criterion 1 type message
set profiles signaling sipAdaptorProfile CenturyLink_SMM2 rule 6 criterion 1 message messageTypes requestAll
set profiles signaling sipAdaptorProfile CenturyLink_SMM2 rule 6 criterion 2 type header
set profiles signaling sipAdaptorProfile CenturyLink_SMM2 rule 6 criterion 2 header name From condition exist
set profiles signaling sipAdaptorProfile CenturyLink_SMM2 rule 6 applyMatchHeader one
set profiles signaling sipAdaptorProfile CenturyLink_SMM2 rule 6 action 1 type header operation regsub
set profiles signaling sipAdaptorProfile CenturyLink_SMM2 rule 6 action 1 to type header value From
set profiles signaling sipAdaptorProfile CenturyLink_SMM2 rule 6 action 1 regexp string 216.110.2.227 matchInstance all
set profiles signaling sipAdaptorProfile CenturyLink_SMM2 rule 6 action 1 from type value value voip.centurylink.com
set profiles signaling sipAdaptorProfile CenturyLink_SMM2 state enabled
commit

 

Test Results

 

Table : Test Results

S.NoTitleDescriptionTest SetupResultComment
g729-001Anonymous Call Rejection ActivatePBX User dials *77
PSTN Calls PBX User with Caller ID Block
Should receive an announcement
*77 is Dialed PBX and leaves PBX Phones gets an announcement 
Calling Party blocks caller ID
Calling party makes a call to PBX User Calling Party receives an announcement when PBX user is dialed
Passed 
g729-002Anonymous Call Rejection Deactivate PBX User dials *87
PSTN Calls PBX User with Caller ID block
Call Should Complete
*87 is dialed PBX User receives and announcement 
PSTN calls PBX User
PSTN Phone receives ringback
PBX Phone gets ringing
PBX Phone get Caller ID
PBX Phone answer the Call
2 way audio is received
PBX Phone releases Calls
PSTN receives a Bye
Passed 
g729-003Anonymous Call PBX-BW PBX sends anonymous call to BW
BW delivers the calls Private or unknown or anonymous to PSTN
PBX is configured to send a call to BW as anonymous with TN as PSTN
BW delivers the call to PSTN as Private or Anonymous
PSTN phone shows the call as Private or Anonymous
Call is answered by PSTN
PBX user hangs up the call
Passed 
g729-004Alien TNs A call PBX call originate where the from TN that is not part of the customer trunk group.  As long as the pilot number is identified in outgoing call by PAI, the BroadWorks will accept and route the call.After Alien TN is set up on a Trunk in CenturyLink Network
PBX User Places a Call to PSTN
PBX User receives ringback
PSTN receives ringing
PSTN receives caller id of the Alien TN
PSTN answers the call
2 way audio is received
PBX Phone releases Calls
PSTN receives a Bye
Passed 
g729-005Barge In Create a Pick Up Group with 2 PBX Users
PSTN Calls PBX User 1
PBX User 2 dials *33 +PBX User Ext
PSTN, User 1, and User 2 should be conf
PSTN calls PBX User 1
PSTN Phone receives ringback
PBX Phone gets ringing
PBX Phone get Caller ID
PBX Phone answer the Call
2 way audio is received
PBX User 2 Dials *33 + PBX User 1 Extension
PSTN, PBX User 1, and PBX User 2 are conferenced together
2 Way Audio is heard by all Legs
PBX User 1 drops from Call
2 way Audio is heard by PSTN and PBX User 2
PSTN drops call
PBX User 2 receives a Bye
Passed 
g729-006Barge In Exempt In the Portal Enable Barge In Exempt
Create a Pick Up Group with 2 PBX Users
PSTN Calls PBX User 1
PBX User 2 dials *33 +PBX User Ext
User 2 Should not be conf
Barge in Exempt is set on PBX user 1
PSTN calls PBX User 1
PSTN Phone receives ringback
PBX Phone gets ringing
PBX Phone get Caller ID
PBX Phone answer the Call
2 way audio is received
PBX User 2 Dials *33 + PBX User 1 Extension
PBX user 2 is not allowed to barge in
PSTN drops the call
PBX User 1 receives a Bye
Passed 
g729-007PSTN to BWAPSTN calls BWA Number
Enter Calling Number (2nd Phone Location)
Enter Called Number  (PSTN)
PSTN should Ring with Caller ID of 2nd Phone
Answer Call
BroadWorks Anywhere is set up in Portal
PSTN 1 Calls BWA Number
Announcement is received
Enter calling Number (2nd Phone created in BWA)
Announcement received
Enter Called Number (PSTN 2)
PSTN 1 receives ringback
PSTN 2 receives ringing
PSTN 2 receives caller ID of 2nd Phone (Not of PSTN 1)
PSTN 2 Answers Call
2 way audio is received
PSTN 2 releases Calls
PSTN receives a Bye
BlockedAnywhere service is not activated for test account
g729-008PSTN to PBX user with BWAPSTN Calls User with BWA
PBX User and 2nd Location should Ring
Answer phone for 2nd location
BroadWorks Anywhere is set up in Portal
PSTN 1 Calls BWA Number
PSTN 1 receives ringback
Both PBX User and 2nd Phone Location Number gets ringing
Both PBX User and 2nd Phone Location Number gets Caller ID of PSTN
Call is answered on Location 2
PBX User no longer gets ringing (cancel)
2 way Audio
Location 2 releases call
PSTN receives a Bye
BlockedAnywhere service is not activated for test account
g729-009Call Forwarding Always Activate PBX User dials *72
Enter the CFA Destination TN
PSTN calls PBX User with CFA
PBX User 1 Dials *72
Announcement is heard
PBX User enter PBX User 2 TN
Announcement is heard
PBX Receives a Bye
PSTN Calls PBX User 1
PSTN receives ringback
PBX User 1 does not ring
PBX User 2 gets ringing
PBX user 2 receives Caller ID (PSTN Originator Caller)
PBX User answers call
2 way Audio
PSTN releases call
PBX User 2 receives a Bye
Passed 
g729-010Call Forwarding Always InterrogatePBX User with CFA dials
*21*
Announcement received
PBX User 1 Dials *21*
Announcement is Heard
After announcement completes PBX User 1 receives a Bye
Passed 
g729-011Call Forwarding Always  Deactivate PBX User with CFA dials *73
PSTN Calls PBX User 
PBX User 1 Dials *73
Announcement is Heard
After announcement completes PBX User 1 receives a Bye
Passed 
g729-012Call Forwarding Always to Voicemail Activate PBX User Dials *21
PSTN Dials PBX User with CFA
Verify Call goes to Voicemail
PBX User 1 Dials *21
Announcement is received
When announcement completes PBX User receives a Bye
PSTN User Calls PBX User 1
Call should go directly to voicemail
Announcement is Heard
Leave voicemail
After leaving voicemail PSTN should receive a Bye
PBX User 1 should receive and MWI
Passed 
g729-013Call Forwarding Always to Voicemail Deactivate PBX User with CFA dial #21
PSTN dials PBX User verify Phone rings
PBX User 1 Dials #21
Announcement is Heard
After announcement completes PBX User 1 receives a Bye
Passed 
g729-014PSTN call is CFB to PSTN with ID RestrictedPBX configured to send CFB to BW for identified Station.
BW is configured with CFB to PSTN2.
PSTN 1 Calls PBX  with Caller ID Restricted
PSTN 1 hears ring back
PBX send 486 Busy to BW
BW forwards the call to PSTN2
PSTN 2 hears ringing
PSTN 2 Caller ID displays Private/Anonymous
PSTN 2 Answers the call.
Two way voice path is established between PSTN 1 and PSTN 2
PSTN 2 hangs up
PSTN2 should receive Private/Anonymous as CLIDPassed 
g729-015PSTN with Privacy call to PBX is CFA to PSTN PBX User  is configured  with CFA to PSTN 2
PSTN 1 Calls PBX  with Caller ID Restricted
PSTN 1 hears ring back
PBX sends a new call to BW with PSTN 2 Number, From as Anonymous and PAI set to Pilot Number
BW forwards the call to PSTN2
PSTN 2 hears ringing
PSTN 2 Caller ID displays Pilot Number
PSTN 2 Answers the call.
Two way voice path is established between PSTN 1 and PSTN 2
PSTN 2 hangs up
Pilot Number should be shown as CLID on PSTN2Passed 
g729-016Call Forwarding Busy ActivatePBX User dials *90
Enter the CFB Destination TN
PSTN calls PBX User with CFB
PBX User 1 Dials *90
Announcement is heard
PBX User enter PBX User 2 TN
Announcement is heard
PBX Receives a Bye
Busy PBX User 1
PSTN Calls PBX User 1
PSTN receives ringback
PBX User 1 does not ring
PBX User 2 gets ringing
PBX user 2 receives Caller ID (PSTN Originator Caller)
PBX User answers call
2 way Audio
PSTN releases call
PBX User 2 receives a Bye
Passed 
g729-017Call Forwarding Busy Interrogate PBX User with CFB dials
*67*
Announcement received
PBX User 1 Dials *67*
Announcement is Heard
After announcement completes PBX User 1 receives a Bye
Passed 
g729-018Call Forwarding Busy Deactivate PBX User with CFB dials *91
PSTN Calls PBX User 
PBX User 1 Dials *91
Announcement is Heard
After announcement completes PBX User 1 receives a Bye
Passed 
g729-019Call Forwarding Busy to Voicemail ActivatePBX User Dials *40
PSTN Dials PBX User with CFB
Verify Call goes to Voicemail
PBX User 1 Dials *40
Announcement is received
When announcement completes PBX User receives a Bye
Busy PBX User 1
PSTN User Calls PBX User 1
Call should go directly to voicemail
Announcement is Heard
Leave voicemail
After leaving voicemail PSTN should receive a Bye
PBX User 1 should receive and MWI
Passed 
g729-020Call Forwarding Busy to Voicemail Deactivate PBX User with CFB dial #40
PSTN dials PBX User verify Phone rings
PBX User 1 Dials #40
Announcement is Heard
After announcement completes PBX User 1 receives a Bye
Passed 
g729-021Call Forwarding No Answer Activate PBX User dials *92
Enter the CFNA Destination TN
PSTN calls PBX User with CFNA
PBX User 1 Dials *92
Announcement is heard
PBX User enters PBX User 2 TN
Announcement is heard
PBX Receives a Bye
PSTN Calls PBX User 1
PSTN receives ringback
PBX User 1 gets ringing
PBX User 1 receives Caller ID
After timer is RNA is received
PBX User 2 gets ringing
PBX user 2 receives Caller ID
PBX User 2 answers call
2 way Audio
PSTN releases call
PBX User 2 receives a Bye
Passed 
g729-022Call Forwarding No Answer- RNA TimerPBX User dials *610
Enter 1 #
PSTN calls PBX User with CFNA
Verify Call is forwarded
PBX User 1 Dials *610
Announcement is Heard
PBX User enter 1 for amount of Rings
After announcement completes PBX User 1 receives a Bye
PSTN Calls PBX User 1
PSTN receives ringback
PBX User 1 gets ringing
PBX User 1 receives Caller ID
After timer is RNA is received
PBX User 2 gets ringing
PBX user 2 receives Caller ID
PBX User 2 answers call
2 way Audio
PSTN releases call
PBX User 2 receives a Bye
PassedMinimum is 0 or 2 rings which can be entered
g729-023Call Forwarding No Answer Interrogate PBX User with CFNA dials
*61*
Announcement received
PBX User 1 Dials *61*
Announcement is Heard
After announcement completes PBX User 1 receives a Bye
Passed 
g729-024Call Forwarding No Answer Deactivate PBX User with CFNA dials *93
PSTN Calls PBX User 
PBX User 1 Dials *93
Announcement is Heard
After announcement completes PBX User 1 receives a Bye
Passed 
g729-025Call Forwarding No Answer to Voicemail ActivatePBX User Dials *41
PSTN Dials PBX User with CFNA
Verify Call goes to Voicemail
PBX User 1 Dials *41
Announcement is received
When announcement completes PBX User receives a Bye
Busy PBX User 1
PSTN User Calls PBX User 1
Call should go to voicemail after RNA timer is reached
Announcement is Heard
Leave voicemail
After leaving voicemail PSTN should receive a Bye
PBX User 1 should receive and MWI
Passed 
g729-026Call Forwarding No Answer to Voicemail Deactivate PBX User with CFNA dial #41
PSTN dials PBX User verify Phone rings
PBX User 1 Dials #41
Announcement is Heard
After announcement completes PBX User 1 receives a Bye
Passed 
g729-027Call Forwarding Not Reachable Activate PBX User dials *94
Enter the CFNR Destination TN
Unregister Pilot TNs
PSTN calls PBX User with CFNR
Verify Call is forwarded
Register Pilot TNs
PBX User 1 Dials *94
Announcement is heard
PBX User enter PBX User 2 TN
Announcement is heard
PBX Receives a Bye
Unplug SBC Lan Cable
PSTN Calls PBX User 1
PSTN receives ringback
PBX User 1 does not ring
PSTN User 2 gets ringing
PSTN user 2 receives Caller ID (PSTN Originator Caller)
PSTN User answers call
2 way Audio
PSTN User 1 releases call
PSTN User 2 receives a Bye
Passed 
g729-028Call Forwarding Not Reachable Interrogate PBX User with CFNR dials
*63*
Announcement received
PBX User 1 Dials *63*
Announcement is Heard
After announcement completes PBX User 1 receives a Bye
Passed 
g729-029Call Forwarding Not Reachable DeactivatePBX User with CFNR dials *95
PSTN Calls PBX User 
PBX User 1 Dials *95
Announcement is Heard
After announcement completes PBX User 1 receives a Bye
Passed 
g729-030Call Forwarding Selective ActivateLog into Portal and set up Call forward selective User with a PSTN Number
PBX User with CFS enters #76
PSTN User calls PBX User with CFS
Call should be call forwarded
Log into Portal and set up Call forward selective User with a PSTN Number
PBX User with CFS dials #76
Announcement received
PBX User receives a Bye
From a Selected PSTN Dial PBX User 1
PBX User should not Ring
Call should be call forwarded to the CFS Destination
PSTN receives Ringback
Destination receives Ringing
Destination receives Caller ID (Originator PSTN)
Destination answers call
2 way Audio
PSTN ends the call
Destination receives a Bye
Passed 
g729-031Call Forwarding Selective DeactivatePBX User with CFS enters #77
PSTN User calls PBX User
Call should not be forwarded
PBX User 1 Dials #77
Announcement is heard
PBX Receives a Bye
PSTN Calls PBX User 1
PSTN receives ringback
PBX User 1 gets ringing
PBX user 1 receives Caller ID
PBX User answers call
2 way Audio
PSTN releases call
PBX User 2 receives a Bye
Passed 
g729-032Call Return by PBX User PBX User dials *69PSTN 1 Calls PBX User 1
PSTN 1 receives ringback
PBX User 1 receives ringing
PBX User 1 receives caller ID
PBX User 1 answers call
2 way Audio
PSTN 1 ends the call
PBX User 1 receives a Bye
PBX User 1 Dials *69
PBX User receives Ringback
PSTN 1 receives Ringing
PSTN receives Caller ID
PSTN answers
2 way Audio
PSTN releases call
PBX User 1 receives a Bye
Passed 
g729-033Consultative Transfer with SIP REFER PBX User Calls PSTN
PBX User transfers PSTN to PSTN2
PBX User has Audio with PSTNs
PSTN 1 has MOH
PBX User Transfers Call
PSTN and PSTN2 now have audio
 Not SupportedCUCM 11 does not support outbound SIP Transfer with Refer method
g729-034Unattended Transfer with SIP REFER PBX User Calls PSTN
PBX User transfers PSTN to PSTN2
During Ringback PBX User transfers
PSTN 1 has MOH
PSTN2 answers call
PSTN and PSTN2 now have audio
 Not SupportedCUCM 11 does not support outbound SIP Transfer with Refer method
g729-035Consultative Transfer  PBX User Calls PSTN
PBX User transfers PSTN to PBX User 2
PBX User 1 has Audio with PBX User 2
PSTN 1 has MOH
PBX User Transfers Call
PSTN and PBX 2 now have audio

PBX User 1 Calls PSTN
PBX User receives Ringback
PSTN 1 receives Ringing
PSTN 1 receives Caller ID
PSTN 1 answers
2 way Audio
PBX User transfers call to PBX User 2
PSTN User gets MOH
PBX User 1 gets Dial tone
PBX User 1 dials PBX User 2 Extension
PBX User 1 receives Ringback
PBX User 2 receives Ringing
PBX User 2 receives Caller ID of PBX User 1
PBX User 2 answers the Call
2 way Audio
PBX User 1 transfers the call
MOH Ends
PSTN 1 and PBX User 2 are now connected
2 Way Audio
PSTN 1 Ends the call
PBX User 2 receives the Bye
Passed 
g729-036Unattended Transfer PBX User Calls PSTN
PBX User transfers PSTN to PBX User 2
During Ringback PBX User transfers
PSTN 1 has MOH
PBX User 2 answers call
PSTN and PBX User 2 now have audio

PBX User 1 Calls PSTN
PBX User receives Ringback
PSTN 1 receives Ringing
PSTN 1 receives Caller ID
PSTN 1 answers
2 way Audio
PBX User transfers call to PBX User 2
PSTN User gets MOH
PBX User 1 gets Dial tone
PBX User 1 dials PBX User 2 Extension
PBX User 1 receives Ringback
PBX User 2 receives Ringing
PBX User 2 receives Caller ID of PSTN 1
PBX User 1 release call
PBX User 2 answers the Call
MOH Ends
2 way Audio
PSTN 1 release the call
PBX User 2 receives the Bye
Passed 
g729-037Call Waiting Persistent ActivatePBX User dials *43
PSTN Calls PBX User
PSTN 2 Calls PBX User
Verify Call Waiting Tone
PBX User 1 Dials *43
Announcement is heard
PBX Receives a Bye
PSTN User 1 Calls PBX User 1
PSTN User 1 receives ringback
PBX User 1 gets ringing
PBX user 1 receives Caller ID
PBX User answers call
2 way Audio
PSTN User 2 Calls PBX User 1
PSTN User 2 receives ringback
PBX User 1 receives caller ID
PBX User 1 hear Call Waiting Tone
PBX User Places PSTN User 1 on Hold
PSTN User 1 hears MOH
PBX User 1 answers Call from PSTN 2
2 way Audio
Verify PBX User 1 can swap between to callers
While on PBX User 1 and PSTN User 1
PSTN 1 releases Call
PBX User 1 receives a Bye
Call 2 should still be up with PSTN 2 hearing MOH
Passed 
g729-038Call Waiting Persistent DeactivatePBX User Dials #43
PSTN Calls PBX User
PSTN 2 Calls PBX User
Call 2 should go to voicemail
PBX User 1 Dials #43
Announcement is heard
PBX User 1 Receives a Bye after Announcement is completed
PSTN User Calls PBX User 1
PSTN User 1 Calls PBX User 1
PSTN User 1 receives ringback
PBX User 1 gets ringing
PBX user 1 receives Caller ID
PBX User 1 answers call
2 way Audio
PSTN User 1 releases the call
PBX User 1 receives a Bye
PassedCall Forwarding Busy to Voicemail  is activated to send PSTN User 2 to voicemail
g729-039Customer Originated Trace PSTN Calls PBX User
PBX User Answers the Call
PBX User Hangs up call
PBX User enters *57
Verify announcement
PSTN Calls PBX User 1
PSTN receives ringback
PBX User 1 gets ringing
PBX user 1 receives Caller ID
PBX User answers call
2 way Audio
PSTN releases call
PBX User 2 receives a Bye
PBX User 1 Dial *57
Announcement received
Announcement Completes PBX User receives a Bye
Passed 
g729-040Enhanced Call Logs Log into portal and verify Call logsLog into the portal for PBX User 1
On main screen verify calls Logs are displayed
Missed
Received
Placed
Passed 
g729-041Last Number Redial PBX User dials *66
The last number dialed should be called
PSTN Calls PBX User 1
PSTN receives ringback
PBX User 1 gets ringing
PBX user 1 receives Caller ID
PBX User answers call
2 way Audio
PSTN releases call
PBX User 1 receives a Bye
PBX User 1 Dial *66
PSTN receives ringback
PBX User 1 gets ringing
PBX user 1 receives Caller ID
PBX User answers call
2 way Audio
PSTN releases call
PBX User 1 receives a Bye
Passed 
g729-042MOH Verify MOH for conference, transfer, and holdPSTN Calls PBX User 1
PSTN receives ringback
PBX User 1 gets ringing
PBX user 1 receives Caller ID
PBX User answers call
2 way Audio
PBX User 1 Places call on Hold
PSTN receives MOH
PBX User retrieves call from Hold
2 way Audio
PSTN releases call
PBX User 2 receives a Bye
Passed 
g729-043Remote Office  - Like CFAProvision Remote office for a SIP Trunk user on the BroadWorks portal to use PSTN number A. Place a call from a PSTN number B to the SIP Trunk user's DID and verify that it is forwarded to PSTN number A (the destination configured in BroadWorks). Log into the portal for PBX User 1
Set up remote Office to PSTN Number 1
PSTN User 2 Calls PBX User 1
PSTN 2 receives ringback
PSTN User 1  gets ringing with PSTN 2 Caller ID and Diversion header for PBX User1
PSTN User 1 answers call
2 way Audio
PSTN 1 releases call
Passed 
g729-044Remote Office - Quick CallProvision Remote office for a SIP Trunk user 1 on the BroadWorks portal to use PSTN number A. On the BW portal,  Manage Users, select Configure Features of User 1, under Quick Call, add PSTN B number and click on the Call Button. PSTN A should Start Ringing with PBX User 1 Caller ID.  Log into the portal for PBX User 1
Set up remote Office to PSTN Number 1
Initiate a Quack Call to PSTN 2 on the portal
PSTN User 1  gets ringing with PBX User 1  Caller ID
PSTN user 1 answers the call.
Now PSTN2 should start ringing with PBX User1 as Caller ID.
PSTN 1 might hear ringback based on how long PSTN 2 rings.
PSTN 2 answers the call
2 way Audio
PSTN 1 releases call
Passed 
g729-045Remote Office - Click to  CallProvision Remote office for a SIP Trunk user 1 on the BroadWorks portal to use PSTN number A. On the BW portal,  Manage Users, select Configure Features of User 1, under Call Logs, select either incoming/outgoing/missed calls and  Click on a  Call under Phone Number Click To call column. PSTN A should Start Ringing with PBX User 1 Caller ID.  Log into the portal for PBX User 1
Set up remote Office to PSTN Number 1
Review call logs and identify a call log that needs to be called via Click to Call.
Click on the identified call log under Click to Call
PSTN User 1  gets ringing with PBX User 1  Caller ID
PSTN user 1 answers the call.
Now PSTN2 should start ringing with PBX User1 as Caller ID.
PSTN 1 might hear ringback based on how long PSTN 2 rings.
PSTN 2 answers the call
2 way Audio
PSTN 1 releases call
Passed 
g729-046Selective Call Acceptance Provision selective call acceptance in the BroadWorks portal. Place a call from an accepted TN to the SIP Trunk User. Verify that the call completes normally. Place a call from a TN that is not on the accept list and verify that the call does not reach the  SBC.Log into the portal for PBX User 1
Set up Selected Call Acceptance to PSTN Number 1
PSTN Calls PBX User 1
PSTN receives ringback
PBX User 1 gets ringing
PBX user 1 receives Caller ID
PBX User answers call
2 way Audio
PSTN releases Call
PBX User 1 receives a Bye
Passed 
g729-047Selective Call Rejection Provision selective call rejection in the BroadWorks portal. Place a call from a TN not on the reject list to the SIP Trunk User. Verify that the call completes normally. Place a call from a TN that is on the reject list and verify that the call does not reach the SBC.Log into the portal for PBX User 1
Set up Selected Call rejection to PSTN Number 1
PSTN Calls PBX User 1
Verify PSTN gets an announcement
PSTN receives a Bye
Passed 
g729-048Sequential RingProvision sequential ring in the BroadWorks portal. Place a call to the SIP trunk user. Verify that the numbers in the sequential ring list are dialed in order.Log into the Portal for PBX User 1
Set up Sequential Ring with PBX User 2 and PBX User 3
PSTN Calls PBX User 1
PSTN receives ringback
PBX User 1 gets ringing
PBX user 1 receives Caller ID
After RNA Timer PBX User 1 receives a Cancel
PBX User 2 gets ringing
PBX user 2 receives Caller ID
After RNA Timer PBX User 1 receives a Cancel
PBX User 3 gets ringing
PBX user 3 receives Caller ID
PBX User 3 answers call
2 way Audio
PSTN releases Call
PBX User 3 receives a Bye
Passed 
g729-049Simultaneous Ring Provision Simultaneous ring in the BroadWorks portal. Place a call to the SIP trunk user. Verify that the numbers in the Simultaneous ring list are dialed at once.Log into the Portal for PBX User 1
Set up Simultaneous Ring with PBX User 2 and PBX User 3
PSTN Calls PBX User 1
PSTN receives ringback
PBX User 1 gets ringing
PBX user 1 receives Caller ID
PBX User 2 gets ringing
PBX user 2 receives Caller ID
PBX User 3 gets ringing
PBX user 3 receives Caller ID
PBX User 3 Answers Call
PBX User 1 and 2 receive a Cancel
2 way Audio
PSTN releases Call
PSTN User 3 receives a Bye
Passed 
g729-050Third Party MWI Control NOTIFY  Provision Third Party MWI in the BroadWorks portal. Provision the CT Voice Mail system to notify BroadWorks of unread messages in the user's voice mail box. Confirm that the NOTIFY is sent to BroadWorks and that the NOTIFY is sent to the PBX.PSTN User Calls PBX User 1
Call should go to voicemail after RNA timer is reached
Announcement is Heard
Leave voicemail
After leaving voicemail PSTN should receive a Bye
PBX User 1 should receive and MWI
PBX User 1 dials *86
Log into Mailbox
Listen To Voicemail
Delete Voicemail
Verify MWI is gone
PBX User 1 ends the Call
PassedCall Forwarding No Answer to Voicemail is activated to send PSTN user 1 to voicemail after timer
g729-051Voice Mail ConsultationProvision Voice Mail n the BroadWorks or NYMPH portal. Provision the PBX to forward calls to an external voice mail system as the user's call coverage. Confirm the PBX user's capability to retrieve voice mail from the external Voice Mail system.PSTN User Calls PBX User 1
Call should go to voicemail after RNA timer is reached
Announcement is Heard
Leave voicemail
After leaving voicemail PSTN should receive a Bye
PBX User 1 should receive and MWI
PBX User 1 dials *86
Log into Mailbox
Listen To Voicemail
Delete Voicemail
Verify MWI is gone
PBX User 1 ends the Call
Passed 
g729-052PBX Initiate ConferencePBX User Calls PSTN
PBX User Conferences PBX User 2
PBX User 1 Calls PSTN
PBX User receives Ringback
PSTN 1 receives Ringing
PSTN 1 receives Caller ID
PSTN 1 answers
2 way Audio
PBX User conferences call to PBX User 2
PSTN User gets MOH
PBX User 1 gets Dial tone
PBX User 1 dials PBX User 2 Extension
PBX User 1 receives Ringback
PBX User 2 receives Ringing
PBX User 2 receives Caller ID of PBX User 1
PBX User 2 answers the Call
2 way Audio
PBX User 1 conferences the call
MOH Ends
PSTN 1, PBX User 1 and PBX User 2 are now connected
2 Way Audio
PBX User 1 Ends the call
PBX User 2 and PSTN receives the Bye
Passed 
g729-053PSTN Initiate ConferencePBX User calls PSTN
PSTN conferences PBX User2
PBX User 1 Calls PSTN
PBX User receives Ringback
PSTN 1 receives Ringing
PSTN 1 receives Caller ID
PSTN 1 answers
2 way Audio
PSTN User 1 conferences call to PBX User 2
PBX User 1 gets MOH
PSTN User 1 gets Dial tone
PSTN User 1 dials PBX User 2 Extension
PSTN User 1 receives Ringback
PBX User 2 receives Ringing
PBX User 2 receives Caller ID of PSTN User 1
PBX User 2 answers the Call
2 way Audio
PSTN User 1 conferences the call
MOH Ends
PSTN 1, PBX User 1 and PBX User 2 are now connected
2 Way Audio
PSTN User 1 Ends the call
PBX User 1 and PBX User 2 Still Have Audio
PBX User 1 End the Call
PBX User 2 receives a Bye
Passed 
g729-054Huntgroup Seq RingPSTN Calls Huntgroup Seq ring
Answer call on 2nd Member
Log into Admin Portal
Create Huntgroup with 3 members
PSTN Calls Huntgroup
PSTN receives ringback
PBX User 1 gets ringing
PBX user 1 receives Caller ID
After RNA Timer is reached
PBX User 2 gets ringing
PBX user 2 receives Caller ID
PBX User 2 Answers the call
2 way Audio
PSTN ends the call
PBX User 2 receives a Bye
BlockedHunt group is not activated for test account
g729-055Huntgroup Seq Ring RNA to VoicemailPSTN calls Huntgroup Seq ring
RNA to Voicemail
Log into Admin Portal
Create Huntgroup with 3 members
PSTN Calls Huntgroup
PSTN receives ringback
PBX User 1 gets ringing
PBX user 1 receives Caller ID
After RNA Timer is reached
PBX User 2 gets ringing
PBX user 2 receives Caller ID
After RNA Timer is reached
PBX User 3 gets ringing
PBX user 3 receives Caller ID
Call should go to voicemail after RNA timer is reached
Announcement is Heard
Leave voicemail
After leaving voicemail PSTN should receive a Bye
PBX User 1 should receive and MWI
PBX User 1 dials *86
Enter *#
Log into HuntGroup Mailbox
Listen To Voicemail
Delete Voicemail
PBX User 1 ends the Call
BlockedHunt group is not activated for test account
g729-056Huntgroup Sim RingPSTN calls Huntgroup Sim ring 3 members
Answer Call
Log into Admin Portal
Create Huntgroup with 3 members with Sequential Ring
PSTN Calls Huntgroup
PSTN receives ringback
PBX User 1 gets ringing
PBX user 1 receives Caller ID
PBX User 2 gets ringing
PBX user 2 receives Caller ID
PBX user 3 receives Caller ID
PBX User 3 Answers the call
PBX User 3 Answers the Call
2 way Audio
PSTN ends the call
PBX User 2 receives a Bye
BlockedHunt group is not activated for test account
g729-057PBX to PBXPBX User Calls PBX User2 Same Trunk
Verify RTP is dropped to SBC
PBX User 1 Calls PBX User 2
PBX  User 1 receives ringback
PBX User 2 gets ringing
PBX user 2 receives Caller ID
PBX User 2 answers call
2 way Audio RTP is on SBC/PBX
PBX User 1 End the call
PBX User 2 receives a Bye
Passed 
g729-058PSTN to PBXPSTN to PBX UserPSTN User 1 Calls PBX User 1
PSTN User 1 receives ringback
PBX User 1 gets ringing
PBX user 1 receives Caller ID
PBX User answers call
2 way Audio
PSTN User Ends The Call
PBX User 1 receives a Bye
Passed 
g729-059PBX to PSTNPBX User to PSTNPBX User 1 Calls PSTN User 1
PBX User 1 receives ringback
PSTN User 1 gets ringing
PSTN user 1 receives Caller ID
PSTN User answers call
2 way Audio
PSTN User Ends The Call
PBX User 1 receives a Bye
Passed 
g729-060PBX to PBX Different PBX (diff realm)PBX User to PBX User Different PBX (diff realm)PBX User 1 Calls PBX User 2 Diff Realm
PBX  User 1 receives ringback
PBX User 2 gets ringing
PBX user 2 receives Caller ID
PBX User 2 answers call
2 way Audio RTP
PBX User 1 End the call
PBX User 2 receives a Bye
Passed 
g729-061PSTN to PBXPSTN to PBX User Fax CallPSTN User 1 Fax Calls PBX User 1 Fax
PBX User 1 gets ringing
PBX user 1 receives Caller ID
PBX User 1 Fax answers call
Fax is received
PBX User Ends The Call
PSTN User 1 receives a Bye
Not SupportdSBC does not detect fax transmission so it depends on endpoints to send Re-Invite with correct codec
g729-062PBX to PSTNPBX User to PSTN Fax CallPBX User 1 Fax Calls PSTN User 1 Fax
PSTN User 1 gets ringing
PSTN user 1 receives Caller ID
PSTN User 1 Fax answers call
Fax is received
PSTN User Ends The Call
PBX User 1 receives a Bye
Not SupportdSBC does not detect fax transmission so it depends on endpoints to send Re-Invite with correct codec
g729-063PSTN to PBX -T38PSTN to PBX User Fax CallPSTN User 1 Fax Calls PBX User 1 Fax
PBX User 1 gets ringing
PBX user 1 receives Caller ID
PBX User 1 Fax answers call
Fax is received
PBX User Ends The Call
PSTN User 1 receives a Bye
Not SupportdSBC does not detect fax transmission so it depends on endpoints to send Re-Invite with correct codec
g729-064PBX to PSTN -T38PBX User to PSTN Fax CallPBX User 1 Fax Calls PSTN User 1 Fax
PSTN User 1 gets ringing
PSTN user 1 receives Caller ID
PSTN User 1 Fax answers call
Fax is received
PSTN User Ends The Call
PBX User 1 receives a Bye
Not SupportdSBC does not detect fax transmission so it depends on endpoints to send Re-Invite with correct codec
g729-065PBX to PSTN - Packet Marking for SIG packetsPBX to PSTN Call to verify that signaling packets are marked with DSCP = 24 (0x18)All outgoing SIP Signaling packets are marked with DSCP=24PassedSame as g729-059
g729-066PBX to PSTN - Packet Marking for RTP packetsPBX to PSTN Call to verify that rtp packets are marked with DSCP = 40 (0x28)All outgoing SIP RTP packets are marked with DSCP=40PassedSame as g729-059
g729-067PBX to PSTN - Directory assistancePBX User Calls PBX 411 and speaks with directory assistantPBX User 1 dials 411
Call is delivered to Directory Assistant for enquiry
Once the user hears an announcement or speaks with an operator, PBX user hangs up the call
Passed 
g729-068PBX to PSTN - Toll FreePBX User Calls 800.366.8201 to test toll free numbersPBX User 1 dials 800.366.8201 (CTL Support)
Call is delivered to CenturyLink Support
Once the user hears an announcement or speaks with an operator, PBX user hangs up the call
Passed 
g729-069PBX to PSTN - 911PBX User Calls 911 to get emergency supportPBX User 1 dials xxx-xxx-xxxx (CTL Rep)
Call is delivered to CenturyLink Rep
PBX User makes conferences 911 operator
PBX User, CTL rep and 911 operator are conferenced
????
PassedSame routing as for g729-067
g729-070PBX to PSTN - InternationalPBX User Calls international numberInternational Call is successfully established and torn down.Passed 
g711-001Anonymous Call Rejection ActivatePBX User dials *77
PSTN Calls PBX User with Caller ID Block
Should receive an announcement
*77 is Dialed PBX and leaves PBX Phones gets an announcement
Calling Party blocks caller ID
Calling party makes a call to PBX User Calling Party receives an announcement when PBX user is dialed
Passed 
g711-002Anonymous Call Rejection Deactivate PBX User dials *87
PSTN Calls PBX User with Caller ID block
Call Should Complete
*87 is dialed PBX User receives and announcement
PSTN calls PBX User
PSTN Phone receives ringback
PBX Phone gets ringing
PBX Phone get Caller ID
PBX Phone answer the Call
2 way audio is received
PBX Phone releases Calls
PSTN receives a Bye
Passed 
g711-003Anonymous Call PBX-BW PBX sends anonymous call to BW
BW delivers the calls Private or unknown or anonymous to PSTN
PBX is configured to send a call to BW as anonymous with TN as PSTN
BW delivers the call to PSTN as Private or Anonymous
PSTN phone shows the call as Private or Anonymous
Call is answered by PSTN
PBX user hangs up the call
Passed 
g711-004Alien TNs A call PBX call originate where the from TN that is not part of the customer trunk group.  As long as the pilot number is identified in outgoing call by PAI, the BroadWorks will accept and route the call.After Alien TN is set up on a Trunk in CenturyLink Network
PBX User Places a Call to PSTN
PBX User receives ringback
PSTN receives ringing
PSTN receives caller id of the Alien TN
PSTN answers the call
2 way audio is received
PBX Phone releases Calls
PSTN receives a Bye
Passed 
g711-005Barge In Create a Pick Up Group with 2 PBX Users
PSTN Calls PBX User 1
PBX User 2 dials *33 +PBX User Ext
PSTN, User 1, and User 2 should be conf
PSTN calls PBX User 1
PSTN Phone receives ringback
PBX Phone gets ringing
PBX Phone get Caller ID
PBX Phone answer the Call
2 way audio is received
PBX User 2 Dials *33 + PBX User 1 Extension
PSTN, PBX User 1, and PBX User 2 are conferenced together
2 Way Audio is heard by all Legs
PBX User 1 drops from Call
2 way Audio is heard by PSTN and PBX User 2
PSTN drops call
PBX User 2 receives a Bye
Passed 
g711-006Barge In Exempt In the Portal Enable Barge In Exempt
Create a Pick Up Group with 2 PBX Users
PSTN Calls PBX User 1
PBX User 2 dials *33 +PBX User Ext
User 2 Should not be conf
Barge in Exempt is set on PBX user 1
PSTN calls PBX User 1
PSTN Phone receives ringback
PBX Phone gets ringing
PBX Phone get Caller ID
PBX Phone answer the Call
2 way audio is received
PBX User 2 Dials *33 + PBX User 1 Extension
PBX user 2 is not allowed to barge in
PSTN drops the call
PBX User 1 receives a Bye
Passed 
g711-007PSTN to BWAPSTN calls BWA Number
Enter Calling Number (2nd Phone Location)
Enter Called Number  (PSTN)
PSTN should Ring with Caller ID of 2nd Phone
Answer Call
BroadWorks Anywhere is set up in Portal
PSTN 1 Calls BWA Number
Announcement is received
Enter calling Number (2nd Phone created in BWA)
Announcement received
Enter Called Number (PSTN 2)
PSTN 1 receives ringback
PSTN 2 receives ringing
PSTN 2 receives caller ID of 2nd Phone (Not of PSTN 1)
PSTN 2 Answers Call
2 way audio is received
PSTN 2 releases Calls
PSTN receives a Bye
BlockedAnywhere service is not activated for test account
g711-008PSTN to PBX user with BWAPSTN Calls User with BWA
PBX User and 2nd Location should Ring
Answer phone for 2nd location
BroadWorks Anywhere is set up in Portal
PSTN 1 Calls BWA Number
PSTN 1 receives ringback
Both PBX User and 2nd Phone Location Number gets ringing
Both PBX User and 2nd Phone Location Number gets Caller ID of PSTN
Call is answered on Location 2
PBX User no longer gets ringing (cancel)
2 way Audio
Location 2 releases call
PSTN receives a Bye
BlockedAnywhere service is not activated for test account
g711-009Call Forwarding Always Activate PBX User dials *72
Enter the CFA Destination TN
PSTN calls PBX User with CFA
PBX User 1 Dials *72
Announcement is heard
PBX User enter PBX User 2 TN
Announcement is heard
PBX Receives a Bye
PSTN Calls PBX User 1
PSTN receives ringback
PBX User 1 does not ring
PBX User 2 gets ringing
PBX user 2 receives Caller ID (PSTN Originator Caller)
PBX User answers call
2 way Audio
PSTN releases call
PBX User 2 receives a Bye
Passed 
g711-010Call Forwarding Always InterrogatePBX User with CFA dials
*21*
Announcement received
PBX User 1 Dials *21*
Announcement is Heard
After announcement completes PBX User 1 receives a Bye
Passed 
g711-011Call Forwarding Always  Deactivate PBX User with CFA dials *73
PSTN Calls PBX User 
PBX User 1 Dials *73
Announcement is Heard
After announcement completes PBX User 1 receives a Bye
Passed 
g711-012Call Forwarding Always to Voicemail Activate PBX User Dials *21
PSTN Dials PBX User with CFA
Verify Call goes to Voicemail
PBX User 1 Dials *21
Announcement is received
When announcement completes PBX User receives a Bye
PSTN User Calls PBX User 1
Call should go directly to voicemail
Announcement is Heard
Leave voicemail
After leaving voicemail PSTN should receive a Bye
PBX User 1 should receive and MWI
Passed 
g711-013Call Forwarding Always to Voicemail Deactivate PBX User with CFA dial #21
PSTN dials PBX User verify Phone rings
PBX User 1 Dials #21
Announcement is Heard
After announcement completes PBX User 1 receives a Bye
Passed 
g711-014PSTN with Privacy call to PBX is CFA to PSTN PBX User  is configured  with CFA to PSTN 2
PSTN 1 Calls PBX  with Caller ID Restricted
PSTN 1 hears ring back
PBX sends a new call to BW with PSTN 2 Number, From as Anonymous and PAI set to Pilot Number
BW forwards the call to PSTN2
PSTN 2 hears ringing
PSTN 2 Caller ID displays Pilot Number
PSTN 2 Answers the call.
Two way voice path is established between PSTN 1 and PSTN 2
PSTN 2 hangs up
Pilot Number should be shown as CLID on PSTN2Passed 
g711-015PSTN call is CFB to PSTN with ID RestrictedPBX configured to send CFB to BW for identified Station.
BW is configured with CFB to PSTN2.
PSTN 1 Calls PBX  with Caller ID Restricted
PSTN 1 hears ring back
PBX send 486 Busy to BW
BW forwards the call to PSTN2
PSTN 2 hears ringing
PSTN 2 Caller ID displays Private/Anonymous
PSTN 2 Answers the call.
Two way voice path is established between PSTN 1 and PSTN 2
PSTN 2 hangs up
PSTN2 should receive Private/Anonymous as CLIDPassed 
g711-016Call Forwarding Busy ActivatePBX User dials *90
Enter the CFB Destination TN
PSTN calls PBX User with CFB
PBX User 1 Dials *90
Announcement is heard
PBX User enter PBX User 2 TN
Announcement is heard
PBX Receives a Bye
Busy PBX User 1
PSTN Calls PBX User 1
PSTN receives ringback
PBX User 1 does not ring
PBX User 2 gets ringing
PBX user 2 receives Caller ID (PSTN Originator Caller)
PBX User answers call
2 way Audio
PSTN releases call
PBX User 2 receives a Bye
Passed 
g711-017Call Forwarding Busy Interrogate PBX User with CFB dials
*67*
Announcement received
PBX User 1 Dials *67*
Announcement is Heard
After announcement completes PBX User 1 receives a Bye
Passed 
g711-018Call Forwarding Busy Deactivate PBX User with CFB dials *91
PSTN Calls PBX User 
PBX User 1 Dials *91
Announcement is Heard
After announcement completes PBX User 1 receives a Bye
Passed 
g711-019Call Forwarding Busy to Voicemail ActivatePBX User Dials *40
PSTN Dials PBX User with CFB
Verify Call goes to Voicemail
PBX User 1 Dials *40
Announcement is received
When announcement completes PBX User receives a Bye
Busy PBX User 1
PSTN User Calls PBX User 1
Call should go directly to voicemail
Announcement is Heard
Leave voicemail
After leaving voicemail PSTN should receive a Bye
PBX User 1 should receive and MWI
Passed 
g711-020Call Forwarding Busy to Voicemail Deactivate PBX User with CFB dial #40
PSTN dials PBX User verify Phone rings
PBX User 1 Dials #40
Announcement is Heard
After announcement completes PBX User 1 receives a Bye
Passed 
g711-021Call Forwarding No Answer Activate PBX User dials *92
Enter the CFNA Destination TN
PSTN calls PBX User with CFNA
PBX User 1 Dials *92
Announcement is heard
PBX User enters PBX User 2 TN
Announcement is heard
PBX Receives a Bye
PSTN Calls PBX User 1
PSTN receives ringback
PBX User 1 gets ringing
PBX User 1 receives Caller ID
After timer is RNA is received
PBX User 2 gets ringing
PBX user 2 receives Caller ID
PBX User 2 answers call
2 way Audio
PSTN releases call
PBX User 2 receives a Bye
Passed 
g711-022Call Forwarding No Answer- RNA TimerPBX User dials *610
Enter 1 #
PSTN calls PBX User with CFNA
Verify Call is forwarded
PBX User 1 Dials *610
Announcement is Heard
PBX User enter 1 for amount of Rings
After announcement completes PBX User 1 receives a Bye
PSTN Calls PBX User 1
PSTN receives ringback
PBX User 1 gets ringing
PBX User 1 receives Caller ID
After timer is RNA is received
PBX User 2 gets ringing
PBX user 2 receives Caller ID
PBX User 2 answers call
2 way Audio
PSTN releases call
PBX User 2 receives a Bye
PassedMinimum is 0 or 2 rings which can be entered
g711-023Call Forwarding No Answer Interrogate PBX User with CFNA dials
*61*
Announcement received
PBX User 1 Dials *61*
Announcement is Heard
After announcement completes PBX User 1 receives a Bye
Passed 
g711-024Call Forwarding No Answer Deactivate PBX User with CFNA dials *93
PSTN Calls PBX User 
PBX User 1 Dials *93
Announcement is Heard
After announcement completes PBX User 1 receives a Bye
Passed 
g711-025Call Forwarding No Answer to Voicemail ActivatePBX User Dials *41
PSTN Dials PBX User with CFNA
Verify Call goes to Voicemail
PBX User 1 Dials *41
Announcement is received
When announcement completes PBX User receives a Bye
Busy PBX User 1
PSTN User Calls PBX User 1
Call should go to voicemail after RNA timer is reached
Announcement is Heard
Leave voicemail
After leaving voicemail PSTN should receive a Bye
PBX User 1 should receive and MWI
Passed 
g711-026Call Forwarding No Answer to Voicemail Deactivate PBX User with CFNA dial #41
PSTN dials PBX User verify Phone rings
PBX User 1 Dials #41
Announcement is Heard
After announcement completes PBX User 1 receives a Bye
Passed 
g711-027Call Forwarding Not Reachable Activate PBX User dials *94
Enter the CFNR Destination TN
Unregister Pilot TNs
PSTN calls PBX User with CFNR
Verify Call is forwarded
Register Pilot TNs
PBX User 1 Dials *94
Announcement is heard
PBX User enter PBX User 2 TN
Announcement is heard
PBX Receives a Bye
Unplug SBC Lan Cable
PSTN Calls PBX User 1
PSTN receives ringback
PBX User 1 does not ring
PSTN User 2 gets ringing
PSTN user 2 receives Caller ID (PSTN Originator Caller)
PSTN User answers call
2 way Audio
PSTN User 1 releases call
PSTN User 2 receives a Bye
Passed 
g711-028Call Forwarding Not Reachable Interrogate PBX User with CFNR dials
*63*
Announcement received
PBX User 1 Dials *63*
Announcement is Heard
After announcement completes PBX User 1 receives a Bye
Passed 
g711-029Call Forwarding Not Reachable DeactivatePBX User with CFNR dials *95
PSTN Calls PBX User 
PBX User 1 Dials *95
Announcement is Heard
After announcement completes PBX User 1 receives a Bye
Passed 
g711-030Call Forwarding Selective ActivateLog into Portal and set up Call forward selective User with a PSTN Number
PBX User with CFS enters #76
PSTN User calls PBX User with CFS
Call should be call forwarded
Log into Portal and set up Call forward selective User with a PSTN Number
PBX User with CFS dials #76
Announcement received
PBX User receives a Bye
From a Selected PSTN Dial PBX User 1
PBX User should not Ring
Call should be call forwarded to the CFS Destination
PSTN receives Ringback
Destination receives Ringing
Destination receives Caller ID (Originator PSTN)
Destination answers call
2 way Audio
PSTN ends the call
Destination receives a Bye
Passed 
g711-031Call Forwarding Selective DeactivatePBX User with CFS enters #77
PSTN User calls PBX User
Call should not be forwarded
PBX User 1 Dials #77
Announcement is heard
PBX Receives a Bye
PSTN Calls PBX User 1
PSTN receives ringback
PBX User 1 gets ringing
PBX user 1 receives Caller ID
PBX User answers call
2 way Audio
PSTN releases call
PBX User 2 receives a Bye
Passed 
g711-032Call Return by PBX User PBX User dials *69PSTN 1 Calls PBX User 1
PSTN 1 receives ringback
PBX User 1 receives ringing
PBX User 1 receives caller ID
PBX User 1 answers call
2 way Audio
PSTN 1 ends the call
PBX User 1 receives a Bye
PBX User 1 Dials *69
PBX User receives Ringback
PSTN 1 receives Ringing
PSTN receives Caller ID
PSTN answers
2 way Audio
PSTN releases call
PBX User 1 receives a Bye
Passed 
g711-033Consultative Transfer with SIP REFER PBX User Calls PSTN
PBX User transfers PSTN to PSTN2
PBX User has Audio with PSTNs
PSTN 1 has MOH
PBX User Transfers Call
PSTN and PSTN2 now have audio
 Not SupportedCUCM 11 does not support outbound SIP Transfer with Refer method
g711-034Unattended Transfer with SIP REFER PBX User Calls PSTN
PBX User transfers PSTN to PSTN2
During Ringback PBX User transfers
PSTN 1 has MOH
PSTN2 answers call
PSTN and PSTN2 now have audio
 Not SupportedCUCM 11 does not support outbound SIP Transfer with Refer method
g711-035Consultative Transfer  PBX User Calls PSTN
PBX User transfers PSTN to PBX User 2
PBX User 1 has Audio with PBX User 2
PSTN 1 has MOH
PBX User Transfers Call
PSTN and PBX 2 now have audio

PBX User 1 Calls PSTN
PBX User receives Ringback
PSTN 1 receives Ringing
PSTN 1 receives Caller ID
PSTN 1 answers
2 way Audio
PBX User transfers call to PBX User 2
PSTN User gets MOH
PBX User 1 gets Dial tone
PBX User 1 dials PBX User 2 Extension
PBX User 1 receives Ringback
PBX User 2 receives Ringing
PBX User 2 receives Caller ID of PBX User 1
PBX User 2 answers the Call
2 way Audio
PBX User 1 transfers the call
MOH Ends
PSTN 1 and PBX User 2 are now connected
2 Way Audio
PSTN 1 Ends the call
PBX User 2 receives the Bye
Passed 
g711-036Unattended Transfer PBX User Calls PSTN
PBX User transfers PSTN to PBX User 2
During Ringback PBX User transfers
PSTN 1 has MOH
PBX User 2 answers call
PSTN and PBX User 2 now have audio

PBX User 1 Calls PSTN
PBX User receives Ringback
PSTN 1 receives Ringing
PSTN 1 receives Caller ID
PSTN 1 answers
2 way Audio
PBX User transfers call to PBX User 2
PSTN User gets MOH
PBX User 1 gets Dial tone
PBX User 1 dials PBX User 2 Extension
PBX User 1 receives Ringback
PBX User 2 receives Ringing
PBX User 2 receives Caller ID of PSTN 1
PBX User 1 release call
PBX User 2 answers the Call
MOH Ends
2 way Audio
PSTN 1 release the call
PBX User 2 receives the Bye
Passed 
g711-037Call Waiting Persistent ActivatePBX User dials *43
PSTN Calls PBX User
PSTN 2 Calls PBX User
Verify Call Waiting Tone
PBX User 1 Dials *43
Announcement is heard
PBX Receives a Bye
PSTN User 1 Calls PBX User 1
PSTN User 1 receives ringback
PBX User 1 gets ringing
PBX user 1 receives Caller ID
PBX User answers call
2 way Audio
PSTN User 2 Calls PBX User 1
PSTN User 2 receives ringback
PBX User 1 receives caller ID
PBX User 1 hear Call Waiting Tone
PBX User Places PSTN User 1 on Hold
PSTN User 1 hears MOH
PBX User 1 answers Call from PSTN 2
2 way Audio
Verify PBX User 1 can swap between to callers
While on PBX User 1 and PSTN User 1
PSTN 1 releases Call
PBX User 1 receives a Bye
Call 2 should still be up with PSTN 2 hearing MOH
Passed 
g711-038Call Waiting Persistent DeactivatePBX User Dials #43
PSTN Calls PBX User
PSTN 2 Calls PBX User
Call 2 should go to voicemail
PBX User 1 Dials #43
Announcement is heard
PBX User 1 Receives a Bye after Announcement is completed
PSTN User Calls PBX User 1
PSTN User 1 Calls PBX User 1
PSTN User 1 receives ringback
PBX User 1 gets ringing
PBX user 1 receives Caller ID
PBX User 1 answers call
2 way Audio
PSTN User 1 releases the call
PBX User 1 receives a Bye
PassedCall Forwarding Busy to Voicemail  is activated to send PSTN User 2 to voicemail
g711-039Customer Originated Trace PSTN Calls PBX User
PBX User Answers the Call
PBX User Hangs up call
PBX User enters *57
Verify announcement
PSTN Calls PBX User 1
PSTN receives ringback
PBX User 1 gets ringing
PBX user 1 receives Caller ID
PBX User answers call
2 way Audio
PSTN releases call
PBX User 2 receives a Bye
PBX User 1 Dial *57
Announcement received
Announcement Completes PBX User receives a Bye
Passed 
g711-040Enhanced Call Logs Log into portal and verify Call logsLog into the portal for PBX User 1
On main screen verify calls Logs are displayed
Missed
Received
Placed
Passed 
g711-041Last Number Redial PBX User dials *66
The last number dialed should be called
PSTN Calls PBX User 1
PSTN receives ringback
PBX User 1 gets ringing
PBX user 1 receives Caller ID
PBX User answers call
2 way Audio
PSTN releases call
PBX User 1 receives a Bye
PBX User 1 Dial *66
PSTN receives ringback
PBX User 1 gets ringing
PBX user 1 receives Caller ID
PBX User answers call
2 way Audio
PSTN releases call
PBX User 1 receives a Bye
Passed 
g711-042MOH Verify MOH for conference, transfer, and holdPSTN Calls PBX User 1
PSTN receives ringback
PBX User 1 gets ringing
PBX user 1 receives Caller ID
PBX User answers call
2 way Audio
PBX User 1 Places call on Hold
PSTN receives MOH
PBX User retrieves call from Hold
2 way Audio
PSTN releases call
PBX User 2 receives a Bye
Passed 
g711-043Remote Office  - Like CFAProvision Remote office for a SIP Trunk user on the BroadWorks portal to use PSTN number A. Place a call from a PSTN number B to the SIP Trunk user's DID and verify that it is forwarded to PSTN number A (the destination configured in BroadWorks). Log into the portal for PBX User 1
Set up remote Office to PSTN Number 1
PSTN User 2 Calls PBX User 1
PSTN 2 receives ringback
PSTN User 1  gets ringing with PSTN 2 Caller ID and Diversion header for PBX User1
PSTN User 1 answers call
2 way Audio
PSTN 1 releases call
Passed 
g711-044Remote Office - Quick CallProvision Remote office for a SIP Trunk user 1 on the BroadWorks portal to use PSTN number A. On the BW portal,  Manage Users, select Configure Features of User 1, under Quick Call, add PSTN B number and click on the Call Button. PSTN A should Start Ringing with PBX User 1 Caller ID.  Log into the portal for PBX User 1
Set up remote Office to PSTN Number 1
Initiate a Quick Call to PSTN 2 on the portal
PSTN User 1  gets ringing with PBX User 1  Caller ID
PSTN user 1 answers the call.
Now PSTN2 should start ringing with PBX User1 as Caller ID.
PSTN 1 might hear ringback based on how long PSTN 2 rings.
PSTN 2 answers the call
2 way Audio
PSTN 1 releases call
Passed 
g711-045Remote Office - Click to  CallProvision Remote office for a SIP Trunk user 1 on the BroadWorks portal to use PSTN number A. On the BW portal,  Manage Users, select Configure Features of User 1, under Call Logs, select either incoming/outgoing/missed calls and  Click on a  Call under Phone Number Click To call column. PSTN A should Start Ringing with PBX User 1 Caller ID.  Log into the portal for PBX User 1
Set up remote Office to PSTN Number 1
Review call logs and identify a call log that needs to be called via Click to Call.
Click on the identified call log under Click to Call
PSTN User 1  gets ringing with PBX User 1  Caller ID
PSTN user 1 answers the call.
Now PSTN2 should start ringing with PBX User1 as Caller ID.
PSTN 1 might hear ringback based on how long PSTN 2 rings.
PSTN 2 answers the call
2 way Audio
PSTN 1 releases call
Passed 
g711-046Selective Call Acceptance Provision selective call acceptance in the BroadWorks portal. Place a call from an accepted TN to the SIP Trunk User. Verify that the call completes normally. Place a call from a TN that is not on the accept list and verify that the call does not reach the  SBC.Log into the portal for PBX User 1
Set up Selected Call Acceptance to PSTN Number 1
PSTN Calls PBX User 1
PSTN receives ringback
PBX User 1 gets ringing
PBX user 1 receives Caller ID
PBX User answers call
2 way Audio
PSTN releases Call
PBX User 1 receives a Bye
Passed 
g711-047Selective Call Rejection Provision selective call rejection in the BroadWorks portal. Place a call from a TN not on the reject list to the SIP Trunk User. Verify that the call completes normally. Place a call from a TN that is on the reject list and verify that the call does not reach the SBC.Log into the portal for PBX User 1
Set up Selected Call rejection to PSTN Number 1
PSTN Calls PBX User 1
Verify PSTN gets an announcement
PSTN receives a Bye
Passed 
g711-048Sequential RingProvision sequential ring in the BroadWorks portal. Place a call to the SIP trunk user. Verify that the numbers in the sequential ring list are dialed in order.Log into the Portal for PBX User 1
Set up Sequential Ring with PBX User 2 and PBX User 3
PSTN Calls PBX User 1
PSTN receives ringback
PBX User 1 gets ringing
PBX user 1 receives Caller ID
After RNA Timer PBX User 1 receives a Cancel
PBX User 2 gets ringing
PBX user 2 receives Caller ID
After RNA Timer PBX User 1 receives a Cancel
PBX User 3 gets ringing
PBX user 3 receives Caller ID
PBX User 3 answers call
2 way Audio
PSTN releases Call
PBX User 3 receives a Bye
Passed 
g711-049Simultaneous Ring Provision Simultaneous ring in the BroadWorks portal. Place a call to the SIP trunk user. Verify that the numbers in the Simultaneous ring list are dialed at once.Log into the Portal for PBX User 1
Set up Simultaneous Ring with PBX User 2 and PBX User 3
PSTN Calls PBX User 1
PSTN receives ringback
PBX User 1 gets ringing
PBX user 1 receives Caller ID
PBX User 2 gets ringing
PBX user 2 receives Caller ID
PBX User 3 gets ringing
PBX user 3 receives Caller ID
PBX User 3 Answers Call
PBX User 1 and 2 receive a Cancel
2 way Audio
PSTN releases Call
PSTN User 3 receives a Bye
Passed 
g711-050Third Party MWI Control NOTIFY  Provision Third Party MWI in the BroadWorks portal. Provision the CT Voice Mail system to notify BroadWorks of unread messages in the user's voice mail box. Confirm that the NOTIFY is sent to BroadWorks and that the NOTIFY is sent to the PBX.PSTN User Calls PBX User 1
Call should go to voicemail after RNA timer is reached
Announcement is Heard
Leave voicemail
After leaving voicemail PSTN should receive a Bye
PBX User 1 should receive and MWI
PBX User 1 dials *86
Log into Mailbox
Listen To Voicemail
Delete Voicemail
Verify MWI is gone
PBX User 1 ends the Call
Passed 
g711-051Voice Mail ConsultationProvision Voice Mail n the BroadWorks or NYMPH portal. Provision the PBX to forward calls to an external voice mail system as the user's call coverage. Confirm the PBX user's capability to retrieve voice mail from the external Voice Mail system.PSTN User Calls PBX User 1
Call should go to voicemail after RNA timer is reached
Announcement is Heard
Leave voicemail
After leaving voicemail PSTN should receive a Bye
PBX User 1 should receive and MWI
PBX User 1 dials *86
Log into Mailbox
Listen To Voicemail
Delete Voicemail
Verify MWI is gone
PBX User 1 ends the Call
Passed 
g711-052PBX Initiate ConferencePBX User Calls PSTN
PBX User Conferences PBX User 2
PBX User 1 Calls PSTN
PBX User receives Ringback
PSTN 1 receives Ringing
PSTN 1 receives Caller ID
PSTN 1 answers
2 way Audio
PBX User conferences call to PBX User 2
PSTN User gets MOH
PBX User 1 gets Dial tone
PBX User 1 dials PBX User 2 Extension
PBX User 1 receives Ringback
PBX User 2 receives Ringing
PBX User 2 receives Caller ID of PBX User 1
PBX User 2 answers the Call
2 way Audio
PBX User 1 conferences the call
MOH Ends
PSTN 1, PBX User 1 and PBX User 2 are now connected
2 Way Audio
PBX User 1 Ends the call
PBX User 2 and PSTN receives the Bye
PassedPBX user 2 and PSTN still have an audio on after PBX user 1 ends the call
g711-053PSTN Initiate ConferencePBX User calls PSTN
PSTN conferences PBX User2
PBX User 1 Calls PSTN
PBX User receives Ringback
PSTN 1 receives Ringing
PSTN 1 receives Caller ID
PSTN 1 answers
2 way Audio
PSTN User 1 conferences call to PBX User 2
PBX User 1 gets MOH
PSTN User 1 gets Dial tone
PSTN User 1 dials PBX User 2 Extension
PSTN User 1 receives Ringback
PBX User 2 receives Ringing
PBX User 2 receives Caller ID of PSTN User 1
PBX User 2 answers the Call
2 way Audio
PSTN User 1 conferences the call
MOH Ends
PSTN 1, PBX User 1 and PBX User 2 are now connected
2 Way Audio
PSTN User 1 Ends the call
PBX User 1 and PBX User 2 Still Have Audio
PBX User 1 End the Call
PBX User 2 receives a Bye
Passed 
g711-054Huntgroup Seq RingPSTN Calls Huntgroup Seq ring
Answer call on 2nd Member
Log into Admin Portal
Create Huntgroup with 3 members
PSTN Calls Huntgroup
PSTN receives ringback
PBX User 1 gets ringing
PBX user 1 receives Caller ID
After RNA Timer is reached
PBX User 2 gets ringing
PBX user 2 receives Caller ID
PBX User 2 Answers the call
2 way Audio
PSTN ends the call
PBX User 2 receives a Bye
BlockedHunt group is not activated for test account
g711-055Huntgroup Seq Ring RNA to VoicemailPSTN calls Huntgroup Seq ring
RNA to Voicemail
Log into Admin Portal
Create Huntgroup with 3 members
PSTN Calls Huntgroup
PSTN receives ringback
PBX User 1 gets ringing
PBX user 1 receives Caller ID
After RNA Timer is reached
PBX User 2 gets ringing
PBX user 2 receives Caller ID
After RNA Timer is reached
PBX User 3 gets ringing
PBX user 3 receives Caller ID
Call should go to voicemail after RNA timer is reached
Announcement is Heard
Leave voicemail
After leaving voicemail PSTN should receive a Bye
PBX User 1 should receive and MWI
PBX User 1 dials *86
Enter *#
Log into HuntGroup Mailbox
Listen To Voicemail
Delete Voicemail
PBX User 1 ends the Call
BlockedHunt group is not activated for test account
g711-056Huntgroup Sim RingPSTN calls Huntgroup Sim ring 3 members
Answer Call
Log into Admin Portal
Create Huntgroup with 3 members with Sequential Ring
PSTN Calls Huntgroup
PSTN receives ringback
PBX User 1 gets ringing
PBX user 1 receives Caller ID
PBX User 2 gets ringing
PBX user 2 receives Caller ID
PBX user 3 receives Caller ID
PBX User 3 Answers the call
PBX User 3 Answers the Call
2 way Audio
PSTN ends the call
PBX User 2 receives a Bye
BlockedHunt group is not activated for test account
g711-057PBX to PBXPBX User Calls PBX User2 Same Trunk
Verify RTP is dropped to SBC
PBX User 1 Calls PBX User 2
PBX  User 1 receives ringback
PBX User 2 gets ringing
PBX user 2 receives Caller ID
PBX User 2 answers call
2 way Audio RTP is on SBC/PBX
PBX User 1 End the call
PBX User 2 receives a Bye
Passed 
g711-058PSTN to PBXPSTN to PBX UserPSTN User 1 Calls PBX User 1
PSTN User 1 receives ringback
PBX User 1 gets ringing
PBX user 1 receives Caller ID
PBX User answers call
2 way Audio
PSTN User Ends The Call
PBX User 1 receives a Bye
Passed 
g711-059PBX to PSTNPBX User to PSTNPBX User 1 Calls PSTN User 1
PBX User 1 receives ringback
PSTN User 1 gets ringing
PSTN user 1 receives Caller ID
PSTN User answers call
2 way Audio
PSTN User Ends The Call
PBX User 1 receives a Bye
Passed 
g711-060PBX to PBX Different PBX (diff realm)PBX User to PBX User Different PBX (diff realm)PBX User 1 Calls PBX User 2 Diff Realm
PBX  User 1 receives ringback
PBX User 2 gets ringing
PBX user 2 receives Caller ID
PBX User 2 answers call
2 way Audio RTP
PBX User 1 End the call
PBX User 2 receives a Bye
Passed 
g711-061PSTN to PBX -PassthroughPSTN to PBX User Fax CallPSTN User 1 Fax Calls PBX User 1 Fax
PBX User 1 gets ringing
PBX user 1 receives Caller ID
PBX User 1 Fax answers call
Fax is received
PBX User Ends The Call
PSTN User 1 receives a Bye
Passed 
g711-062PBX to PSTN -PassthroughPBX User to PSTN Fax CallPBX User 1 Fax Calls PSTN User 1 Fax
PSTN User 1 gets ringing
PSTN user 1 receives Caller ID
PSTN User 1 Fax answers call
Fax is received
PSTN User Ends The Call
PBX User 1 receives a Bye
Passed 
g711-063PSTN to PBX -T38PSTN to PBX User Fax CallPSTN User 1 Fax Calls PBX User 1 Fax
PBX User 1 gets ringing
PBX user 1 receives Caller ID
PBX User 1 Fax answers call
Fax is received
PBX User Ends The Call
PSTN User 1 receives a Bye
Not SupportedSBC does not detect fax transmission so it depends on endpoints to send Re-Invite with correct codec
g711-064PBX to PSTN -T38PBX User to PSTN Fax CallPBX User 1 Fax Calls PSTN User 1 Fax
PSTN User 1 gets ringing
PSTN user 1 receives Caller ID
PSTN User 1 Fax answers call
Fax is received
PSTN User Ends The Call
PBX User 1 receives a Bye
Not SupportedSBC does not detect fax transmission so it depends on endpoints to send Re-Invite with correct codec
g711-065PBX to PSTN - Packet Marking for SIG packetsPBX to PSTN Call to verify that signaling packets are marked with DSCP = 24 (0x18)All outgoing SIP Signaling packets are marked with DSCP=24PassedSame as g711-059
g711-066PBX to PSTN - Packet Marking for RTP packetsPBX to PSTN Call to verify that rtp packets are marked with DSCP = 40 (0x28)All outgoing SIP RTP packets are marked with DSCP=40PassedSame as g711-059
g711-067PBX to PSTN - Directory assistancePBX User Calls PBX 411 and speaks with directory assistantPBX User 1 dials 411
Call is delivered to Directory Assistant for enquiry
Once the user hears an announcement or speaks with an operator, PBX user hangs up the call
Passed 
g711-068PBX to PSTN - Toll FreePBX User Calls 800.366.8201 to test toll free numbersPBX User 1 dials 800.366.8201 (CTL Support)
Call is delivered to CenturyLink Support
Once the user hears an announcement or speaks with an operator, PBX user hangs up the call
Passed 
g711-069PBX to PSTN - 911PBX User Calls 911 to get emergency supportPBX User 1 dials xxx-xxx-xxxx (CTL Rep)
Call is delivered to CenturyLink Rep
PBX User makes conferences 911 operator
PBX User, CTL rep and 911 operator are conferenced
????
PassedSame routing as for g711-067
g711-070PBX to PSTN - InternationalPBX User Calls international numberInternational Call is successfully established and torn down.Passed 
      
S.NoTitleDescriptionTest SetupResultComment
g729-001Configure Dual Trunk on PBXPBX is configured and connected to 2 PSTN GW/SBCsThe steps  will be  based on the type of PBX being utilized.
Ensure that trunks are configured between PBX and SBC.
Verify OPTIONS msgs from either PBX or SBC are being responded correctly by the other entity
Passed 
g729-002Configure Dual Trunk on ITSPITSP is configured and connected to 2 PSTN GW/SBCsThe steps  will be  based on the type of SBC being utilized.
Ensure the TWO SBCs are configured with individual trunks to ITSP
Passed 
g729-003Regitration of Dual TrunksEnsure that both trunks to ITSP are registered successfully using the individual trunk registration information1. Each SBC is configured with a trunk to ITSP and associated authentiation/digest and registration information.
2. Invoke a command on SBC to register the trunk with ITSP.
3. Verify that 200 OK is received from ITSP for both the trunks.
Passed 
g729-004Inbound PSTN calls pick correct trunk to SBCVerify that PSTN to PBX inbound calls arrive on both the trunks when multiple calls are made1.    Dial an inbound call to the PBX.
2.    Verify ringing is heard by calling and called parties.
3.    Verify the trace shows a valid ringing indication message
4.    Take called party phone off-hook.
5.    Verify that a media path is established in both directions.
6.    Hang up calling party
7.    Verify the IP/PBX receives a Bye message.
8.    Make a note of the Trunk on which the call arrived to the SBC and PBX.
9.    Repeat the above steps 3 more times (total 4 calls).
10.   Verify that calls to PBX arrive on both the trunks.
11.   Document Test Results.
12.   Save Trace.
Passed 
g729-005PBX calls are delivered to PSTN on both the trunksCalls from PBX to PSTN are delivered to ITSP/PSTN utilizing both the configured trunks1.    Dial an outbound call from the PBX.
2.    Verify ringing is heard by calling and called parties.
3.    Verify the trace shows a valid ringing indication message
4.    Take called party phone off-hook.
5.    Verify that a media path established in both directions.
6.    Hang up Calling Party
7.    Verify the IP/PBX sends a Bye message.
8.    Make a note of the Trunk on which the call was sent to ITSP.
9.    Repeat the above steps 3 more times (total 4 calls).
10.   Verify that calls from PBX are sent out on both the trunks to ITSP.
11.   Verify each call has PAI sent per the trunk configuration
12.   Document Test Results.
13.   Save Trace.
Passed 
g729-006Alien TN calls on 1st trunkVerify calls are successful with Alien TNs on 1st trunk1. After Alien TN is set up on a Trunk1 in CenturyLink Network
2. PBX User Places a Call to PSTN
3. PBX User receives ring back
4. PSTN receives ringing
5. PSTN receives caller id of the Alien TN
6. PSTN answers the call
7. 2 way audio is received
8. PBX Phone releases Calls
9. PSTN receives a Bye
Passed 
g729-007Alien TN calls on 2nd  trunkVerify calls are successful with Alien TNs on 2nd  trunk1. After Alien TN is set up on a Trunk2 in CenturyLink Network
2. PBX User Places a Call to PSTN
3. PBX User receives ring back
4. PSTN receives ringing
5. PSTN receives caller id of the Alien TN
6. PSTN answers the call
7. 2 way audio is received
8. PBX Phone releases Calls
9. PSTN receives a Bye
Passed 
g729-008Failover of 1st trunk WAN - PSTN-PBXEnsure that calls are delivered from PSTN to PBX when the first trunk has failed on the WAN side1. Down the WAN interface associated with Trunk 1.
2. Make 3 calls from PSTN to PBX (one after the other)
3. Ensure that all 3 calls are delivered to PBX utilizing Trunk 2
4. PBX user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
Passed 
g729-009Failover of 1st trunk WAN - PBX-PSTNEnsure that calls are delivered from PBX to PSTN when the first trunk has failed on the WAN side1. Down the WAN interface associated with Trunk 1.
2. Make 3 calls from PBX to PSTN (one after the other)
3. Ensure that all 3 calls are delivered to PSTN utilizing Trunk 2
4. PSTN user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
Passed 
g729-010Restore 1st trunk  WAN: PSTN-PBXEnsure that calls are delivered from PSTN to PBX when the first trunk has has been restored1. WAN interface associated with Trunk 1 is brought back into service.
2. Make 3 calls from PSTN to PBX (one after the other)
3. Ensure that at least one call is delivered to the PBX via Trunk 1
4. PBX user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
Passed 
g729-011Restore 1st trunk  WAN: PBX-PSTNEnsure that calls are delivered from PBX to PSTN when the first trunk has has been restored1. WAN interface associated with Trunk 1 is brought back into service.
2. Make 3 calls from PBX to PSTN (one after the other)
3. Ensure that at least one call is delivered to the PSTN via Trunk 1
4. PSTN user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
Passed 
g729-012Failover of 2nd trunk WAN: PSTN-PBXEnsure that calls are delivered from PSTN to PBX when the second trunk has failed on the WAN side1. Down the WAN interface associated with Trunk 2.
2. Make 3 calls from PSTN to PBX (one after the other)
3. Ensure that all 3 calls are delivered to PBX utilizing Trunk 1
4. PBX user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
Passed 
g729-013Failover of 2nd trunk WAN: PBX-PSTNEnsure that calls are delivered from PBX-PSTN when the second trunk has failed on the WAN side1. Down the WAN interface associated with Trunk 2.
2. Make 3 calls from PBX to PSTN (one after the other)
3. Ensure that all 3 calls are delivered to PBX utilizing Trunk 1
4. PSTN user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
Passed 
g729-014Restore 2nd trunk WAN: PSTN-PBXEnsure that calls are delivered from PSTN to PBX when the second trunk has has been restored1. WAN interface associated with Trunk 2 is brought back into service.
2. Make 3 calls from PSTN to PBX (one after the other)
3. Ensure that at least one call is delivered to the PBX via Trunk 2
4. PSTN user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
Passed 
g729-015Restore 2nd trunk WAN: PBX-PSTNEnsure that calls are delivered from PBX to PSTN when the second trunk has has been restored1. WAN interface associated with Trunk 2 is brought back into service.
2. Make 3 calls from PBX to PSTN (one after the other)
3. Ensure that at least one call is delivered to the PBX via Trunk 2
4. PBX user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
Passed 
g729-016Failover of 1st trunk LAN - PBX to PSTNEnsure that calls are delivered from PBX to PSTN  when the first trunk has failed on the LAN side1. Down the LAN interface associated with Trunk 1.
2. Make 3 calls from PBX to PSTN (one after the other)
3. Ensure that all 3 calls are delivered to PSTN utilizing Trunk 2
4. PSTN user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
Passed 
g729-017Failover of 1st trunk LAN - PSTN to PBXEnsure that calls are delivered from PSTN to PBX  when the first trunk has failed on the LAN side1. Down the LAN interface associated with Trunk 1.
2. Make 3 calls from PSTN to PBX (one after the other)
3. Ensure that all 3 calls are delivered to PBX utilizing Trunk 2
4. PBX user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
Passed 
g729-018Restore 1st trunk  LAN - PBX to PSTNEnsure that calls are delivered from PBX to PSTN when the first trunk has has been restored1. LAN interface associated with Trunk 1 is brought back into service.
2. Make 3 calls from PBX to PSTN (one after the other)
3. Ensure that at least one call is delivered to the PSTN via Trunk 2
4. PSTN user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
Passed 
g729-019Restore 1st trunk  LAN - PSTN to PBXEnsure that calls are delivered from PSTN to PBX when the first trunk has has been restored1. LAN interface associated with Trunk 1 is brought back into service.
2. Make 3 calls from PSTN to PBX (one after the other)
3. Ensure that at least one call is delivered to the PBX via Trunk 2
4. PBX user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
Passed 
g729-020Failover of 2nd trunk LAN - PBX to PSTNEnsure that calls are delivered from PBX to PSTN  when the second trunk has failed on the LAN side1. Down the LAN interface associated with Trunk 2.
2. Make 3 calls from PBX to PSTN (one after the other)
3. Ensure that all 3 calls are delivered to PSTN utilizing Trunk 1
4. PSTN user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
Passed 
g729-021Failover of 2nd trunk LAN - PSTN to PBXEnsure that calls are delivered from PSTN to PBX  when the second trunk has failed on the LAN side1. Down the LAN interface associated with Trunk 2.
2. Make 3 calls from PSTN to PBX (one after the other)
3. Ensure that all 3 calls are delivered to PBX utilizing Trunk 1
4. PBX user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
Passed 
g729-022Restore 2nd trunk  LAN - PBX to PSTNEnsure that calls are delivered from PBX to PSTN when the second trunk has has been restored1. LAN interface associated with Trunk 2 is brought back into service.
2. Make 3 calls from PBX to PSTN (one after the other)
3. Ensure that at least one call is delivered to the PSTN via Trunk 2
4. PSTN user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
Passed 
g729-023Restore 2nd trunk  LAN - PSTN to PBXEnsure that calls are delivered from PSTN to PBX when the second trunk has has been restored1. LAN interface associated with Trunk 2 is brought back into service.
2. Make 3 calls from PSTN to PBX (one after the other)
3. Ensure that at least one call is delivered to the PBX via Trunk 2
4. PBX user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
Passed 
g711-001Configure Dual Trunk on PBXPBX is configured and connected to 2 PSTN GW/SBCsThe steps  will be  based on the type of PBX being utilized.
Ensure that trunks are configured between PBX and SBC.
Verify OPTIONS msgs from either PBX or SBC are being responded correctly by the other entity
Passed 
g711-002Configure Dual Trunk on ITSPITSP is configured and connected to 2 PSTN GW/SBCsThe steps  will be  based on the type of SBC being utilized.
Ensure the TWO SBCs are configured with individual trunks to ITSP
Passed 
g711-003Regitration of Dual TrunksEnsure that both trunks to ITSP are registered successfully using the individual trunk registration information1. Each SBC is configured with a trunk to ITSP and associated authentiation/digest and registration information.
2. Invoke a command on SBC to register the trunk with ITSP.
3. Verify that 200 OK is received from ITSP for both the trunks.
Passed 
g711-004Inbound PSTN calls pick correct trunk to SBCVerify that PSTN to PBX inbound calls arrive on both the trunks when multiple calls are made1.    Dial an inbound call to the PBX.
2.    Verify ringing is heard by calling and called parties.
3.    Verify the trace shows a valid ringing indication message
4.    Take called party phone off-hook.
5.    Verify that a media path is established in both directions.
6.    Hang up calling party
7.    Verify the IP/PBX receives a Bye message.
8.    Make a note of the Trunk on which the call arrived to the SBC and PBX.
9.    Repeat the above steps 3 more times (total 4 calls).
10.   Verify that calls to PBX arrive on both the trunks.
11.   Document Test Results.
12.   Save Trace.
Passed 
g711-005PBX calls are delivered to PSTN on both the trunksCalls from PBX to PSTN are delivered to ITSP/PSTN utilizing both the configured trunks1.    Dial an outbound call from the PBX.
2.    Verify ringing is heard by calling and called parties.
3.    Verify the trace shows a valid ringing indication message
4.    Take called party phone off-hook.
5.    Verify that a media path established in both directions.
6.    Hang up Calling Party
7.    Verify the IP/PBX sends a Bye message.
8.    Make a note of the Trunk on which the call was sent to ITSP.
9.    Repeat the above steps 3 more times (total 4 calls).
10.   Verify that calls from PBX are sent out on both the trunks to ITSP.
11.   Verify each call has PAI sent per the trunk configuration
12.   Document Test Results.
13.   Save Trace.
Passed 
g711-006Alien TN calls on 1st trunkVerify calls are successful with Alien TNs on 1st trunk1. After Alien TN is set up on a Trunk1 in CenturyLink Network
2. PBX User Places a Call to PSTN
3. PBX User receives ring back
4. PSTN receives ringing
5. PSTN receives caller id of the Alien TN
6. PSTN answers the call
7. 2 way audio is received
8. PBX Phone releases Calls
9. PSTN receives a Bye
Passed 
g711-007Alien TN calls on 2nd  trunkVerify calls are successful with Alien TNs on 2nd  trunk1. After Alien TN is set up on a Trunk2 in CenturyLink Network
2. PBX User Places a Call to PSTN
3. PBX User receives ring back
4. PSTN receives ringing
5. PSTN receives caller id of the Alien TN
6. PSTN answers the call
7. 2 way audio is received
8. PBX Phone releases Calls
9. PSTN receives a Bye
Passed 
g711-008Failover of 1st trunk WAN - PSTN-PBXEnsure that calls are delivered from PSTN to PBX when the first trunk has failed on the WAN side1. Down the WAN interface associated with Trunk 1.
2. Make 3 calls from PSTN to PBX (one after the other)
3. Ensure that all 3 calls are delivered to PBX utilizing Trunk 2
4. PBX user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
Passed 
g711-009Failover of 1st trunk WAN - PBX-PSTNEnsure that calls are delivered from PBX to PSTN when the first trunk has failed on the WAN side1. Down the WAN interface associated with Trunk 1.
2. Make 3 calls from PBX to PSTN (one after the other)
3. Ensure that all 3 calls are delivered to PSTN utilizing Trunk 2
4. PSTN user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
Passed 
g711-010Restore 1st trunk  WAN: PSTN-PBXEnsure that calls are delivered from PSTN to PBX when the first trunk has has been restored1. WAN interface associated with Trunk 1 is brought back into service.
2. Make 3 calls from PSTN to PBX (one after the other)
3. Ensure that at least one call is delivered to the PBX via Trunk 1
4. PBX user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
Passed 
g711-011Restore 1st trunk  WAN: PBX-PSTNEnsure that calls are delivered from PBX to PSTN when the first trunk has has been restored1. WAN interface associated with Trunk 1 is brought back into service.
2. Make 3 calls from PBX to PSTN (one after the other)
3. Ensure that at least one call is delivered to the PSTN via Trunk 1
4. PSTN user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
Passed 
g711-012Failover of 2nd trunk WAN: PSTN-PBXEnsure that calls are delivered from PSTN to PBX when the second trunk has failed on the WAN side1. Down the WAN interface associated with Trunk 2.
2. Make 3 calls from PSTN to PBX (one after the other)
3. Ensure that all 3 calls are delivered to PBX utilizing Trunk 1
4. PBX user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
Passed 
g711-013Failover of 2nd trunk WAN: PBX-PSTNEnsure that calls are delivered from PBX-PSTN when the second trunk has failed on the WAN side1. Down the WAN interface associated with Trunk 2.
2. Make 3 calls from PBX to PSTN (one after the other)
3. Ensure that all 3 calls are delivered to PBX utilizing Trunk 1
4. PSTN user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
Passed 
g711-014Restore 2nd trunk WAN: PSTN-PBXEnsure that calls are delivered from PSTN to PBX when the second trunk has has been restored1. WAN interface associated with Trunk 2 is brought back into service.
2. Make 3 calls from PSTN to PBX (one after the other)
3. Ensure that at least one call is delivered to the PBX via Trunk 2
4. PSTN user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
Passed 
g711-015Restore 2nd trunk WAN: PBX-PSTNEnsure that calls are delivered from PBX to PSTN when the second trunk has has been restored1. WAN interface associated with Trunk 2 is brought back into service.
2. Make 3 calls from PBX to PSTN (one after the other)
3. Ensure that at least one call is delivered to the PBX via Trunk 2
4. PBX user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
Passed 
g711-016Failover of 1st trunk LAN - PBX to PSTNEnsure that calls are delivered from PBX to PSTN  when the first trunk has failed on the LAN side1. Down the LAN interface associated with Trunk 1.
2. Make 3 calls from PBX to PSTN (one after the other)
3. Ensure that all 3 calls are delivered to PSTN utilizing Trunk 2
4. PSTN user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
Passed 
g711-017Failover of 1st trunk LAN - PSTN to PBXEnsure that calls are delivered from PSTN to PBX  when the first trunk has failed on the LAN side1. Down the LAN interface associated with Trunk 1.
2. Make 3 calls from PSTN to PBX (one after the other)
3. Ensure that all 3 calls are delivered to PBX utilizing Trunk 2
4. PBX user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
Passed 
g711-018Restore 1st trunk  LAN - PBX to PSTNEnsure that calls are delivered from PBX to PSTN when the first trunk has has been restored1. LAN interface associated with Trunk 1 is brought back into service.
2. Make 3 calls from PBX to PSTN (one after the other)
3. Ensure that at least one call is delivered to the PSTN via Trunk 2
4. PSTN user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
Passed 
g711-019Restore 1st trunk  LAN - PSTN to PBXEnsure that calls are delivered from PSTN to PBX when the first trunk has has been restored1. LAN interface associated with Trunk 1 is brought back into service.
2. Make 3 calls from PSTN to PBX (one after the other)
3. Ensure that at least one call is delivered to the PBX via Trunk 2
4. PBX user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
Passed 
g711-020Failover of 2nd trunk LAN - PBX to PSTNEnsure that calls are delivered from PBX to PSTN  when the second trunk has failed on the LAN side1. Down the LAN interface associated with Trunk 2.
2. Make 3 calls from PBX to PSTN (one after the other)
3. Ensure that all 3 calls are delivered to PSTN utilizing Trunk 1
4. PSTN user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
Passed 
g711-021Failover of 2nd trunk LAN - PSTN to PBXEnsure that calls are delivered from PSTN to PBX  when the second trunk has failed on the LAN side1. Down the LAN interface associated with Trunk 2.
2. Make 3 calls from PSTN to PBX (one after the other)
3. Ensure that all 3 calls are delivered to PBX utilizing Trunk 1
4. PBX user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
Passed 
g711-022Restore 2nd trunk  LAN - PBX to PSTNEnsure that calls are delivered from PBX to PSTN when the second trunk has has been restored1. LAN interface associated with Trunk 2 is brought back into service.
2. Make 3 calls from PBX to PSTN (one after the other)
3. Ensure that at least one call is delivered to the PSTN via Trunk 2
4. PSTN user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
Passed 
g711-023Restore 2nd trunk  LAN - PSTN to PBXEnsure that calls are delivered from PSTN to PBX when the second trunk has has been restored1. LAN interface associated with Trunk 2 is brought back into service.
2. Make 3 calls from PSTN to PBX (one after the other)
3. Ensure that at least one call is delivered to the PBX via Trunk 2
4. PBX user answers the call
5. Verify two way voice path is established
6. Called Party hangs up
7. Both Calling and Called parties are disconnected
8. Document results
9. Save traces
Passed 
 

 


Conclusion

These Application Notes describe the configuration steps required for Sonus SBC SWe Series to successfully interoperate with CenturyLink SIP Trunk. All feature and serviceability test cases were completed and passed with the exceptions/observations noted in Test Results.