ISCC not working.

sorav84 · August 11, 2014, 3:52pm

Hi all,

Please point me to post if this issue has already bee discussed. Here is the flow.

Call is dialed from extension 1001 on first switch.
Call reaches Route point 1101 which on first switch SIP_Switch_1 where a simple strategy is loaded to target agent based on skill which is voice >3.
Our agent voice_agent_1 is logged on to second switch which is SIP_Switch_2
On SIP_Switch_1 access code 2 has been configured pointing to SIP_Switch_2 with target type :Target ISCC.
ERP 201 has been configured on SIP_Switch_2 with access number 2201.Following are T-lib messages after first requestroutecall

message RequestRouteCall
AttributeThisDN ‘1101’
AttributeConnID 006b024e15dfa081
AttributeOtherDN ‘2001’
AttributeLocation ‘SIP_Switch_2’
AttributeExtensions [175] 00 08 00 00..
‘CUSTOMER_ID’ ‘Resources’
‘AGENT’ ‘voice_agent_1’
‘PLACE’ ‘Place_2001’
‘DN’ ‘2001’
‘ACCESS’ ‘2001’
‘SWITCH’ ‘SIP_Switch_2’
‘NVQ’ 1
‘TARGET’ ‘?:voice >3@Routing_Stat_Server.GA’
AttributeRouteType 0 (RouteTypeUnknown)
AttributeReason [14] 00 01 01 00..
‘RTR’ 109
AttributeReferenceID 509
21:07:34.530 Int 04543 Interaction message “RequestRouteCall” received from 372 (“Routing_Server_Primary”)
@21:07:34.5300 [ISCC] Request with xaction type ‘route’ will be processed by strategy ‘ROUTE_ORIGINATION_STRATEGY’
@21:07:34.5300 [ISCC] +? tep [006b024e15dfa081:Switch_2_SIP_TServer_Primary@SIP_Switch_2:006b024e15dfa081]
@21:07:34.5300 [ISCC] Created transaction [route O7012373 o::006b024e15dfa081:1101 d:Switch_2_SIP_TServer_Primary@SIP_Switch_2:006b024e15dfa081:2001 c:Routing_Server_Primary:0000000d@]
@21:07:34.5300 [ISCC] Send to location SIP_Switch_2
@21:07:34.5300 [ISCC] Send to server Switch_2_SIP_TServer_Primary@SIP_Switch_2/0: message ISCCRequestGetCallDataXferService
ISCCAttributeTrackingID 7012411 [006b003b]
ISCCAttributeToken ‘’
ISCCAttributeGlobalTransactionID ‘I1JIAPNJ5H6K7CRCR0P6R6V91O0000HE’
ISCCAttributeISLinkUUID ‘I1JIAPNJ5H6K7CRCR0P6R6V91O0000HG’
ISCCAttributeNetworkDestDN ‘SIP_Switch_2::2001’
ISCCAttributeFirstTransferOriginationDN ‘1101’
ISCCAttributeFirstTransferConnID 006b024e15dfa081
ISCCAttributeFirstTransferHomeLocation ‘Switch_1_SIP_TServer_Primary@SIP_Switch_1’
ISCCAttributeLastTransferOriginationDN ‘1101’
ISCCAttributeLastTransferConnID 006b024e15dfa081
ISCCAttributeLastTransferHomeLocation ‘Switch_1_SIP_TServer_Primary@SIP_Switch_1’
ISCCAttributeCallType Internal
ISCCAttributeCallDataXferType RouteNoToken
ISCCAttributeDestinationDN ‘2001’
ISCCAttributeReferenceID 7012373 [006b0015]
ISCCAttributeOriginationDN ‘1101’
ISCCAttributeConnID 006b024e15dfa081
ISCCAttributeExtensions
‘CUSTOMER_ID’ ‘Resources’
‘AGENT’ ‘voice_agent_1’
‘PLACE’ ‘Place_2001’
‘DN’ ‘2001’
‘ACCESS’ ‘2001’
‘SWITCH’ ‘SIP_Switch_2’
‘NVQ’ 1
‘TARGET’ ‘?:voice >3@Routing_Stat_Server.GA’
ISCCAttributeUpdateRevision 262251 [0004006b]
ISCCAttributeUserData
‘RTargetAgentGroup’ ‘?:voice >3’
‘RVQID’ ‘’
‘RVQDBID’ ‘’
‘RTargetTypeSelected’ ‘2’
‘RTargetRuleSelected’ ‘’
‘RTargetObjectSelected’ ‘?:voice >3’
‘RTargetObjSelDBID’ ‘’
‘RTargetAgentSelected’ ‘voice_agent_1’
‘RTargetPlaceSelected’ ‘Place_2001’
‘RTenant’ ‘Resources’
‘RStrategyName’ ‘simple voice’
‘RStrategyDBID’ ‘169’
‘CBR-actual_volume’ ‘’
‘CBR-Interaction_cost’ ‘’
‘CBR-contract_DBIDs’ ‘’
‘CBR-IT-path_DBIDs’ ‘’
‘RRequestedSkillCombination’ ‘’
‘RRequestedSkills’(list)
‘RTargetRequested’ ‘?:voice >3’
‘CustomerSegment’ ‘default’
‘ServiceType’ ‘default’
‘ServiceObjective’ ‘’
‘PegAG?:voice >3’ 1
@21:07:34.5300 [ISCC] [connection 02dfdc20] Sent to connection [400]2
@21:07:34.5460 [ISCC] [connection 02dfdc20] Received from connection [400]2
@21:07:34.5460 [ISCC] Received from server Switch_2_SIP_TServer_Primary@SIP_Switch_2/0: message ISCCEventCallDataXferService
ISCCAttributeCallDataXferType RouteNoToken
ISCCAttributeCallDataXferResource ‘2201’
ISCCAttributeReferenceID 7012373 [006b0015]
ISCCAttributeTrackingID 7340062 [0070001e]
@21:07:34.5460 [ISCC] Call is committed:
@ c:006b024e15dfa081,02f004e8 @ m:0000000000000000,00000000 p:2 i:00000081 nw:0000000000000000 t:1
p:0000000000000000,02f9b398 @ c:006b024e15dfa081,02f004e8 r:1 c---- n:1001:
p:0000000000000000,02fa9f30 @ c:006b024e15dfa081,02f004e8 r:2 c---- n:1101:
@21:07:34.5460 [ISCC] Message RequestRouteCall [Request:0000002d] is enqueued
@21:07:34.5770 [ISCC] Message RequestRouteCall [Request:0000002d] is dequeued
@21:07:34.5770 [ISCC] Sent to tserver [from ISCC:00000001@]: message RequestRouteCall
AttributeReferenceID 7012373
AttributeConnID 006b024e15dfa081
AttributeOtherDN ‘2201’
AttributeThisDN ‘1101’
AttributeLocation ‘SIP_Switch_2’
AttributeExtensions [175] 00 08 00 00..
‘CUSTOMER_ID’ ‘Resources’
‘AGENT’ ‘voice_agent_1’
‘PLACE’ ‘Place_2001’
‘DN’ ‘2001’
‘ACCESS’ ‘2001’
‘SWITCH’ ‘SIP_Switch_2’
‘NVQ’ 1
‘TARGET’ ‘?:voice >3@Routing_Stat_Server.GA’
AttributeRouteType 0 (RouteTypeUnknown)
AttributeReason [14] 00 01 01 00..
‘RTR’ 109
21:07:34.577 – created: CRequest@2de4218 RequestRouteCall-ISCC[-1]/7012373
21:07:34.577:(1) SIPS:LOGBLOCK:BEGIN:REQUEST:[
21:07:34.577 +++ CIFace::Request +++
– new invoke
– thisCall by party
Parsed: RequestRouteCall
From: ISCC[-1]/7012373
Numbers: +<1101> -<2201>
Calls: 2d9ff48:1 none
Parties: 1101.2fe8758-2d9ff48:1
none
Status: parsed:1 queued:0 sent:0 acked:0 preevent:0 event:0 context:0 transferred:0

– validate
– state check: ok
CIFace: Sent CRequest@2de4218 RequestRouteCall-ISCC[-1]/7012373
– aTmCall 006b024e15dfa081 SetCause: Distributed to 2201
FinishRequest CRequest@2de4218 RequestRouteCall-ISCC[-1]/7012373
IFace stats: q=0 s=0
– complete
– NAData ClRq removed
TNAEmulator::NotifyBackup()
21:07:34.577:(1) GetRegistration::Unable to resolve number for DN:2201
21:07:34.577:(1) Number:[2201] is not internal, looking for service
21:07:34.577:(1) Number:2201 did not find any gateway with matching prefix to send this call to
21:07:34.577:(1) Unable to resolve number for DN:2201
21:07:34.577:(1) SIPDM: failed to get registration info for 2201
21:07:34.577:(1) SIPDM: failed to get registration info for 2201
21:07:34.577:(1) ERROR: 10000006, ResolveNewDestinationDevice(call, party, destination, device), SipCallManagerCallControl.cpp,3187
Response (415): for CRequest@2de4218 RequestRouteCall-ISCC[-1]/7012373
21:07:34.577 Trc 36002 Request rejected: error code 415(Invalid Destination DN)
@21:07:34.5770 [0] 8.0.400.25 send_to_client: message EventError
(Invalid Destination DN)
AttributeEventSequenceNumber 0000000000000315
AttributeTimeinuSecs 577000
AttributeTimeinSecs 1407771454 (21:07:34)
AttributeExtensions [175] 00 08 00 00..
‘CUSTOMER_ID’ ‘Resources’
‘AGENT’ ‘voice_agent_1’
‘PLACE’ ‘Place_2001’
‘DN’ ‘2001’
‘ACCESS’ ‘2001’
‘SWITCH’ ‘SIP_Switch_2’
‘NVQ’ 1
‘TARGET’ ‘?:voice >3@Routing_Stat_Server.GA’
AttributeErrorCode 415
AttributeErrorMessage ‘Invalid Destination DN’
AttributeReason [14] 00 01 01 00..
‘RTR’ 109
AttributeRouteType 0 (RouteTypeUnknown)
AttributeLocation ‘SIP_Switch_2’
AttributeThisDN ‘1101’
AttributeOtherDN ‘2201’
AttributeConnID 006b024e15dfa081
AttributeReferenceID 7012373
AttributeClientID 1
21:07:34.577 Int 04545 Interaction message “EventError” sent to -1 (“ISCC”)
@21:07:34.5770 [ISCC] Send to location SIP_Switch_2
@21:07:34.5770 [ISCC] Send to server Switch_2_SIP_TServer_Primary@SIP_Switch_2/0: message ISCCRequestCancelCallDataXferService

I have highlighted the error text , checked configuration more than once. Please let me know if anyone can point me in right direction.

Thanks
Saurabh

cavagnaro · August 11, 2014, 4:37pm

Message is kinda obvious and has nothing to do with ISCC. How does SIP_Server 1 can ever know how to reach SIP_Server 2? You need to create a Trunk againg SIP Server 2 on SS1 and viceversa.

sorav84 · August 11, 2014, 6:40pm

Hi Cavagnaro,

Thanks for the reply. I have configured two trunk from switch 1 to 2 and switch 2 to 1.But Still no luck. Below is snippet of what I got.

Please note I am using SJ Phone’s SIP direct method to call route point on switch 1.I hope this is by no means cause of issue.

@00:04:32.4100 [ISCC] Received from server Switch_2_SIP_TServer_Primary@SIP_Switch_2/0: message ISCCEventCallDataXferService
ISCCAttributeCallDataXferType RouteNoToken
ISCCAttributeCallDataXferResource ‘2201’
ISCCAttributeReferenceID 7012363 [006b000b]
ISCCAttributeTrackingID 7340081 [00700031]
@00:04:32.4100 [ISCC] Call is committed:
@ c:006b024e288f313d,02dc7670 @ m:0000000000000000,00000000 p:2 i:0000013d nw:0000000000000000 t:2
p:0000000000000000,02ef3420 @ c:006b024e288f313d,02dc7670 r:1 c---- n:Anonymous:
p:0000000000000000,02fb3370 @ c:006b024e288f313d,02dc7670 r:2 c---- n:1101:
@00:04:32.4100 [ISCC] Message RequestRouteCall [Request:00000014] is enqueued
@00:04:32.4260 [ISCC] Message RequestRouteCall [Request:00000014] is dequeued
@00:04:32.4260 [ISCC] Sent to tserver [from ISCC:00000001@]: message RequestRouteCall
AttributeReferenceID 7012363
AttributeConnID 006b024e288f313d
AttributeOtherDN ‘2201’
AttributeThisDN ‘1101’
AttributeLocation ‘SIP_Switch_2’
AttributeExtensions [175] 00 08 00 00..
‘CUSTOMER_ID’ ‘Resources’
‘AGENT’ ‘voice_agent_1’
‘PLACE’ ‘Place_2001’
‘DN’ ‘2001’
‘ACCESS’ ‘2001’
‘SWITCH’ ‘SIP_Switch_2’
‘NVQ’ 1
‘TARGET’ ‘?:voice >3@Routing_Stat_Server.GA’
AttributeRouteType 0 (RouteTypeUnknown)
AttributeReason [14] 00 01 01 00..
‘RTR’ 109
00:04:32.426 – created: CRequest@2ddb0c0 RequestRouteCall-ISCC[-1]/7012363
00:04:32.426:(1) SIPS:LOGBLOCK:BEGIN:REQUEST:[
00:04:32.426 +++ CIFace::Request +++
– new invoke
– thisCall by party
Parsed: RequestRouteCall
From: ISCC[-1]/7012363
Numbers: +<1101> -<2201>
Calls: 2610d00:1 none
Parties: 1101.30d0380-2610d00:1
none
Status: parsed:1 queued:0 sent:0 acked:0 preevent:0 event:0 context:0 transferred:0

– validate
– state check: ok
CIFace: Sent CRequest@2ddb0c0 RequestRouteCall-ISCC[-1]/7012363
– aTmCall 006b024e288f313d SetCause: Distributed to 2201
FinishRequest CRequest@2ddb0c0 RequestRouteCall-ISCC[-1]/7012363
IFace stats: q=0 s=0
– complete
– NAData ClRq removed
TNAEmulator::NotifyBackup()
00:04:32.426:(1) GetRegistration::Unable to resolve number for DN:2201
00:04:32.426:(1) Number:[2201] is not internal, looking for service
00:04:32.426:(1) Selected for Dn 2201(geo-loc:partitionId:cpdCapability): Service ISCC_Trunk_SS1_2 (geo-loc, priority[0], capacity 0 (0% of -2))
00:04:32.426 SIPCONN(Anonymous): re-invite-called-initiated
00:04:32.426:(1) transfer re-INVITE
00:04:32.426 SIPCONN(2201): set monitor 030e2268, 02f5b734
00:04:32.426:(1) SIPCALL(317): add party ‘2201’
– created party_info_tspp 2f2d0b8
– created aTmParty 30e4ac0
SetRole: Destination for 2201.30e4ac0-2610d00:1
– AddParty to 2610d00: 2201.30e4ac0-2610d00:1 after 1101.30d0380-2610d00:1
CreateParty new external: 2201.30e4ac0-2610d00:1
00:04:32.426:(1) Call 317 dn 2201 SetPartyId 644
00:04:32.426:(1) SIPTR(2870): Begin step 0 - SipTransactionTransferCall(2871)
00:04:32.426 SIPCONN(2201): re-invite-null
00:04:32.426:(1) SipDialog: set monitor 030e22dc
00:04:32.426 SIPCONN(2201): main dialog 0 created
00:04:32.426 SIPCONN(2201): Local contact: ‘sip:Anonymous@192.168.1.10:5060’
00:04:32.426 SIPCONN(2201): InitiateDialog: preparing a 3pcc request
00:04:32.426:(1) add party ‘2201’ state
00:04:32.426:(1) SIPDLG[636]: register TRN[642]
00:04:32.426:(1) SIPDLG[636]: TRN[642] flags set to 0x6
00:04:32.426:(1) Sending [392,UDP] 1126 bytes to 192.168.1.11:5060 >>>>>
INVITE sip:2201@192.168.1.11:5060 SIP/2.0
From: “unknown” sip:192.168.1.14;tag=B1E6A0F0-3473-438D-8F94-0ED060087B22-636
To: sip:1101@192.168.1.10:5060
Call-ID: 95B08F5C-947C-4CBC-80AB-18042C286392-317@192.168.1.10
CSeq: 1 INVITE
Content-Length: 356
Content-Type: application/sdp
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK4A049B11-D68C-4754-81F3-97CD747B815F-317
Contact: sip:Anonymous@192.168.1.10:5060
Allow: ACK, BYE, CANCEL, INFO, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE
X-Genesys-Orig: 1101
User-Agent: SJphone/1.65.377a (SJ Labs)
Max-Forwards: 69
X-Genesys-CallUUID: BQRG0A00NP33FCTKIURS4IGU5C00009T
X-ISCC-CofId: location=SIP_Switch_1;cofid=644
Session-Expires: 1800;refresher=uac
Min-SE: 90
Supported: 100rel,timer

v=0
o=- 1407776356 1 IN IP4 192.168.1.14
s=SJphone
c=IN IP4 192.168.1.14
t=0 0
m=audio 49206 RTP/AVP 3 97 98 8 0 101
c=IN IP4 192.168.1.14
a=setup:active
a=sendrecv
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=20
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

00:04:32.426:(1) SipDialog: event SEND_INVITE, t=642, s=2, r=7, m=030e22dc
00:04:32.426 SIPCONN(2201): HandleSipDialogEvent(SEND_INVITE) - filtered
00:04:32.426 SIPCONN(2201): sdp state SDP_STATE_NULL, event SDP_OFFER_SENT
00:04:32.426 SIPCONN(2201): new sdp state SDP_OFFER_SENT, event SDP_OFFER_SENT
00:04:32.426 — CIFace::Request —
00:04:32.426:(1) 00:04:32.426:(1) SIPS:LOGBLOCK:END:REQUEST:]
@00:04:32.4260 [ISCC] Message [Request:00000014] is deleted
00:04:32.426:(1) SIPS:LOGBLOCK:BEGIN:SIPDATA:[
00:04:32.426:(1) Received [392,UDP] 1155 bytes from 192.168.1.11:5060 <<<<<
INVITE sip:2201@192.168.1.10:5060 SIP/2.0
From: “unknown” sip:192.168.1.14;tag=E3F15A63-B21B-4580-923E-EC95F561A350-1053
To: sip:2201@192.168.1.11:5060
Call-ID: E2D8E780-C065-45D5-9761-A94D9FEB6E0E-513@192.168.1.11
CSeq: 1 INVITE
Content-Length: 356
Content-Type: application/sdp
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK0DD64AA5-5993-480B-88A8-F491E35F06C2-513
Contact: sip:Anonymous@192.168.1.11:5060
Allow: ACK, BYE, CANCEL, INFO, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE
X-Genesys-Orig: Anonymous
X-Genesys-Orig: 1101
User-Agent: SJphone/1.65.377a (SJ Labs)
Max-Forwards: 68
X-Genesys-CallUUID: DPLSMD0REP7ED06SNRRA2JN9GO0000G2
X-ISCC-CofId: location=SIP_Switch_2;cofid=1026
Session-Expires: 1800;refresher=uac
Min-SE: 90
Supported: 100rel,timer

v=0
o=- 1407702589 1 IN IP4 192.168.1.14
s=SJphone
c=IN IP4 192.168.1.14
t=0 0
m=audio 49206 RTP/AVP 3 97 98 8 0 101
c=IN IP4 192.168.1.14
a=setup:active
a=sendrecv
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=20
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

00:04:32.426:(1) Unable to resolve number for DN:Anonymous
00:04:32.426:(1) Unable to resolve number for DN:Anonymous

sorav84 · August 11, 2014, 6:53pm

Hi Cavaganaro,

Thanks for the suggestion. It worked. Next thing i need to was change the access number for DN 201 which is ERP on switch 2 back to 201 , earlier it was 2201.

I thought it should be combination of 2(access code on switch 1 for switch 2) and 201(DN Number).

The query further to this is we also have ISCC configured for Avaya switches as well and access number is configured the way I had described. And we haven’t configured trunks between Avaya switches i.e. dummy switches we create in Genesys environment.

Thanks again for your help.

cavagnaro · August 11, 2014, 7:36pm

Some how Avaya switch should know how to reach SIP Server. That config won’t be done on Genesys level but on the PBX itself.

genesysguru · August 12, 2014, 12:46pm

Correct and the last time I did this a few years ago that was done through Avaya SES configuration.

sorav84 · August 17, 2014, 7:20am

Thank you for valuable suggestion

nomster123 · December 4, 2015, 3:38pm

I have done all this but still getting the error for internal dn posted above. Please help. Please mention what will be the prefixes for both the trunk in this case

Kubig · December 4, 2015, 3:40pm

Ufff..as was mentioned for many times. The prefix on tieline trunk should be equal to access code on ISCC switch level

skoom · December 4, 2015, 4:30pm

The lines you’ve highlighted aren’t problems.. The first two is the local switch not being able to find 2201 internally and the second two is the local switch, after recieving an INVITE from the remote switch not recognising the callers CLI (which is the text string Anonymos)..

It looks like the call is being sent to the remote switch and the remote switch doesn’t know what to do with it - and sends it right back..

Since its SIP I have used a text prefix in the past to route calls between SIP Servers. Using the direct (I think direct-uui?) route method the following should work:

SIP Server 1: In the SIP Server application object, add a connection to SIP Server 2 in the connections tab
SIP Server 1: Add a single switch access code to SIP Server 2 with the prefix “GenSIP2/” (without the quotes obviously)
SIP Server 1: Add a TrunkDN that will be used to send SIP traffic to SIP Server 2. Set the “prefix” option to “GenSIP2/” and “replace-prefix” as blank (this will remove the prefix GenSIP2/ before it gets to the second switch.)

SIP Server 2: In the SIP Server application object, add a connection to SIP Server 1 in the connections tab
SIP Server 2: Add a single switch access code to SIP Server 1 with the prefix “GenSIP1/”
SIP Server 2: Add a TrunkDN that will be used to send SIP traffic to SIP Server 2. Set the “prefix” option to “GenSIP1/” and “replace-prefix” as blank.