GVP TG | Issue in Routing the calls to IVR

Hi All,

I am facing issue in routing the call to IVR from DID. For one DID when i route the call to TG it works. For the other it does not work. Please see the below SIP logs and let me know.

13:23:39.755: IncrementRouteRequestCounter, option 100
13:23:39.755: GetRegistration::Unable to resolve number for DN:81740004
13:23:39.755: GetRtPartyId: call 16779276, party 16783216
13:23:39.755: GetRtPartyId: routingLinkId not set
13:23:39.755: GetRegistration::Unable to resolve number for DN:81740004
13:23:39.755: Number:[81740004] is not internal, looking for service
13:23:39.755: Selected for Dn 81740004(geo-loc:partitionId:cpdCapability): Service avaya_sm (geo-loc, priority[0], capacity 0 (0% of 0))
13:23:39.755: GetRtPartyId: call 16779276, party 16783216
13:23:39.755: GetRtPartyId: routingLinkId not set
13:23:39.755: GetRegistration::Unable to resolve number for DN:dummy
13:23:39.755 SIPCONN(+9093018664): re-invite-called-initiated
13:23:39.755: transfer oosp
13:23:39.755: SIPTR(23794): Begin step 0 - SipTransactionRefer(23795)
13:23:39.755: GetRegistration::Unable to resolve number for DN:dummy
13:23:39.755: GetRegistration::Unable to resolve number for DN:dummy
13:23:39.755 SIPCONN(+9093018664): re-invite-called-initiated
13:23:39.755: GetReferTo:+OOSP[0]: >> sip:81740004@10.138.31.123 >>
13:23:39.755: refer 81740004[lc] to +9093018664[lc] : 1=>1=>1
13:23:39.755 SIPCONN(+9093018664): GetOsspReferTo before params: ‘sip:81740004@10.138.31.123’, GetReferToParams from 56f2c70:‘81740004’ = ‘’
13:23:39.755: Sending [29,TCP] 953 bytes to 10.138.31.123:45819 >>>>>
SIP/2.0 302 Moved Temporarily
Via: SIP/2.0/TCP 10.138.31.123;branch=z9hG4bK880602180678708-AP;ft=10;received=10.138.31.123
Via: SIP/2.0/TCP 10.138.31.38:15060;rport=21380;ibmsid=local.1486656157680_1755823_1755810;branch=z9hG4bK880602180678708
Via: SIP/2.0/TCP 10.138.31.38:15060;rport;ibmsid=local.1486656157680_1755822_1755809;branch=z9hG4bK326242225547284
Via: SIP/2.0/TCP 10.138.31.123:5065;branch=z9hG4bK803ca06a2712e7122d15863e5e00-AP;ft=98085;received=10.138.31.123;rport=34648
Via: SIP/2.0/TCP 10.192.189.21:5065;branch=z9hG4bK803ca06a2712e7122d15863e5e00
Via: SIP/2.0/TCP 10.184.33.193;branch=z9hG4bK803ca06a2712e7122d15863e5e00
From: “IP Test Phone” <sip:[email removed]>;tag=803ca06a2712e7120d15863e5e00
To: sip:1740004@10.138.31.123;tag=006C069C-6D65-189B-97B0-6B15890AAA77-86810
Call-ID: 803ca06a2712e7121d15863e5e00
CSeq: 1 INVITE
Contact: sip:81740004@10.138.31.123
X-Genesys-CallInfo: routed
Content-Length: 0

13:23:39.755: SipDialog: event CALLED_RESREJECT, t=10295753, s=9, r=5, m=56f6fd8
13:23:39.755 SIPCONN(+9093018664): HandleSipDialogEvent(CALLED_RESREJECT) - filtered
13:23:39.755: SIPTR(23795): complete
13:23:39.755: SIPTR(23794): Step 0 - SipTransactionRefer(23795) complete
13:23:39.755: SIPTR(23794): Begin step 1 - SipTransactionClearParty(23796)
13:23:39.755: SIPCALL(16779276): add party ‘81740004’
13:23:39.755 SIPPARTY(81740004): Set Origination Info DN - +9093018664 ; Location - qa_sip_switch for party 81740004
– created party_info_tspp 56d3100
– created aTmParty 5633e30
SetRole: Destination for 81740004.5633e30-56d0ab0:1
– AddParty to 56d0ab0: 81740004.5633e30-56d0ab0:1 after 1740004.56d1cc0-56d0ab0:1
CreateParty new external: 81740004.5633e30-56d0ab0:1
13:23:39.755: Call 16779276 dn 81740004 SetPartyId 16783217
13:23:39.755 SIPPARTY(1740004): SipParty::TransferCall, state 5, flagsEx 0x12, cause 17
13:23:39.755: routing dialog ‘16783216’ terminated
13:23:39.755: Adding OtherTrunkName(avaya_sm): done
– thisCall by party
SetContext: for party 1740004.56d1cc0-56d0ab0:1
+++ CIFace::Event +++
+++ Pre-event +++
Type EventRouteUsed
Devices: <1740004/1740004> <-/81740004> <-/->
Calls: 16779276/006c02996375e80c/16779276.56d0ab0/c:2/r:1 0/none
Parties: D1740004/1740004.56d1cc0-56d0ab0:1/l:2/r:1/Queued,RtRequest,Destination
X81740004/81740004.5633e30-56d0ab0:1/l:2/r:0/Null,Destination
none
Cause: Distributed/38, Info: 0
Flags: divert=0 hook=0 postCall=0 active=1 moveAll=1 callType=1 hideOtherPi=0 InternalOther=0
— Pre-event —
+++ Diverted +++
SetRouteEnd: party 1740004.56d1cc0-56d0ab0:1, cause Distributed

TG? This Trunk point to your RM, correct?
What does RM says?
What does MCP says? Call arrives to it?

Full logs and architecture

Thanks for response. It is not reaching RM

So…logs? Come on… What you expect to happen after this?

Enviado de meu E6633 usando Tapatalk

Based on the SIP logs posted above. What do you think?

Some components is rejecting the request. Full SIP (and RM) logs will help.