Call is failing when sending it to GVP

sudheerreddy.alluru1 · April 22, 2014, 4:30pm

Hi,

I deployed GVP recently and tried to play a VXML script and it is failing initially.
Dialed a number 4000 from X-lite phone(pointed to SIP server:port). A trunk(4000) is configured and pointed to RM:port(192.168.50.110:5060) and prefix as 4. configured DID with range 4000-4010 and associated a IVR Profile. Getting error “Failed to establish a call” on X-Lite.

Could you please let me know what i am missing here.

Thanks

SIP Server logs:
21:25:40.729: ERROR: 10000002, SelectClientByNewRequest(*message,result), GSIPTransportImplementation.cpp,890
21:25:40.729: CGSIPTransportImplementationManager::Unable resolve client for 2
21:25:40.729: $+NET:SIP::0:0
21:25:40.729: SIPTR: Received [0,UDP] 405 bytes from 192.168.50.110:5064 <<<<<
OPTIONS sip:192.168.50.112:5065 SIP/2.0
Via: SIP/2.0/UDP 192.168.50.110:5064;branch=z9hG4bK1F0837185690ff
From: sip:GVP@192.168.50.110:5064;tag=22020102-84BE-40CA-F3AF-171072EA6979
To: sip:192.168.50.112:5065
Max-Forwards: 70
CSeq: 27349 OPTIONS
Call-ID: A3CC58E9-B7C9-4CBC-0986-8D02417E868F-5064@192.168.50.110
Contact: sip:GVP@192.168.50.110:5064
Content-Length: 0
Supported: timer, uui

21:25:40.729: Sending [0,UDP] 323 bytes to 192.168.50.110:5064 >>>>>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.50.110:5064;branch=z9hG4bK1F0837185690ff;received=192.168.50.110
From: sip:GVP@192.168.50.110:5064;tag=22020102-84BE-40CA-F3AF-171072EA6979
To: sip:192.168.50.112:5065
CSeq: 27349 OPTIONS
Call-ID: A3CC58E9-B7C9-4CBC-0986-8D02417E868F-5064@192.168.50.110
Content-Length: 0

21:25:40.729: $-NET:SIP::0:169

@21:25:41.0730 [BSYNC] Trace: Received [600]:
message RequestQuerySwitch
attr_#1000 131072
attr_#1001 3
attr_#1002 155
attr_#1003 1398196025
attr_#1004 541
attr_#1005 0
21:25:41.729: ERROR: 10000002, SelectClientByNewRequest(*message,result), GSIPTransportImplementation.cpp,890
21:25:41.729: CGSIPTransportImplementationManager::Unable resolve client for 2
21:25:41.729: $+NET:SIP::0:0
21:25:41.729: SIPTR: Received [0,UDP] 405 bytes from 192.168.50.110:5064 <<<<<
OPTIONS sip:192.168.50.112:5065 SIP/2.0
Via: SIP/2.0/UDP 192.168.50.110:5064;branch=z9hG4bK1F0846E0569100
From: sip:GVP@192.168.50.110:5064;tag=6189A2B3-BCB0-4DA2-CC85-ADD904E380BE
To: sip:192.168.50.112:5065
Max-Forwards: 70
CSeq: 27351 OPTIONS

RM Logs:
2014-04-22T21:21:59.845 Std 20029 EROR 00000000-00000000 2592 09400217 82B223ED-4844-4C45-D8AA-92A8C6DD843A Cannot allocate a resource type matching service type [Service Type: voicexml]
2014-04-22T21:21:59.845 Std 20126 EROR 00000000-00000000 2592 0940023F ResourceModule RequestResource failed: -10 Call Session: 82B223ED-4844-4C45-D8AA-92A8C6DD843A
2014-04-22T21:23:45.103 Std 20029 EROR 00000000-00000000 2592 09400217 A8F08846-320C-4ED5-318B-6894D9797702 Cannot allocate a resource type matching service type [Service Type: voicexml]
2014-04-22T21:23:45.103 Std 20126 EROR 00000000-00000000 2592 0940023F ResourceModule RequestResource failed: -10 Call Session: A8F08846-320C-4ED5-318B-6894D9797702
2014-04-22T21:25:49.389 Std 20029 EROR 00000000-00000000 2576 09400217 321520D2-B89F-4CA1-15B2-DA144EA999D6 Cannot allocate a resource type matching service type [Service Type: voicexml]
2014-04-22T21:25:49.389 Std 20126 EROR 00000000-00000000 2576 0940023F ResourceModule RequestResource failed: -10 Call Session: 321520D2-B89F-4CA1-15B2-DA144EA999D6

Regards,
Sudheer

sudheerreddy.alluru1 · April 22, 2014, 5:54pm

Created new trunk 4001 with same options contact and prefix and sent a call to SIP Server from XLite phone and observed below events in SIP Server logs

ACK sip:4001@192.168.50.110:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.50.112:5065;branch=z9hG4bKAD13E65F-B828-4D42-A901-C37DAD6048C5-8
From: sip:87654@192.168.50.112:5065;tag=AFDF116B-066E-4C3F-80CC-13F9078E9CB4-16
To: sip:4001@192.168.50.112:5065;tag=A8BA40B0-5367-446E-0292-78C1E5A05EAC
Call-ID: 1020B806-D34F-4048-A85F-A25641F6F5B6-8@192.168.50.112
CSeq: 1 ACK
Max-Forwards: 69
User-Agent: X-Lite release 4.5.3 stamp 70569
Content-Length: 0

23:06:03.965: SipDialog: event CALLING_RESREJECT, t=9003, s=9, r=5, m=000000000313be08
23:06:03.965 SIPCONN(4001): HandleSipDialogEvent(CALLING_RESREJECT)
23:06:03.965 SIPCONN(4001): ConvertResponse: 480
23:06:03.965 SIPCONN(4001): ConvertResponse: no conversion configured for 480
23:06:03.965 SIPCONN(4001): state e:4,p:0,s:6,c:10,rc:480,m:1
23:06:03.965 SIPPARTY(4001): 1016 verify update of party-connection state N-F
23:06:03.965 SIPPARTY(4001): 1016 update party-connection state N-F
23:06:03.965: check call postponed until end of operation
23:06:03.965 SIPCONN(4001): terminate dialog
23:06:03.965: SipDialog: set monitor 0000000000000000
23:06:03.965: SipDialog::Terminate(state=9,reason=0)
23:06:03.965 SIPCONN(4001): state e:4,p:6,s:6,c:10,rc:480,m:1
23:06:03.965 SIPPARTY(4001): 1016 verify update of party-connection state F-F
23:06:03.965: SIPTR(68): failed
23:06:03.965: SIPTR(63): Step 4 - SipTransactionConnectNewParty(68) failed
23:06:03.965: SIPTR(63): failed
23:06:03.965: SIPCM: transaction SipScenario failed
23:06:03.965: PI: 00 S[IN]D[87654]C[*D[87654]]P[4001]
23:06:03.965: PI: 01 S[FN]D[4001]P[87654], busy scheduled
23:06:03.965: party cannot be recovered
23:06:03.965: CALLSTATE(a:2,d:0,i:0,e:2,r:0,o:0)
23:06:03.965: apply fast busy tone for device ‘4001’
23:06:03.965: ERROR: No treatment services are configured
23:06:03.965: ERROR: 1000001d, GetDeviceManager().ResolveServiceDevice(call, SIP_MEDIA_SERVICE_MUSIC, geoLocation.CStr(), result), SipMediaResourceManager.cpp,183
23:06:03.965: ERROR: 1000001d, CreateMusicDevice(context, call, party, endPoint, type, device), SipMediaResourceManager.cpp,197
23:06:03.965: ERROR: 1000001d, CreateMusicServiceInternal(context,call,party,type,endPoint,musicService), SipMediaResourceManager.cpp,242
23:06:03.965: cannot create music service 4
23:06:03.965: cannot attach busy tone. Please check configuration.
23:06:03.965: cannot apply tone, re-check the call
23:06:03.965: previous scenario is not cleaned up
23:06:03.965: SIPTR(63): Begin step 0 - SipTransactionRejectCall(69)
23:06:03.965 SIPCONN(87654): re-invite-called-initiated
23:06:04.012: Sending [0,UDP] 486 bytes to 192.168.50.33:26606 >>>>>
SIP/2.0 480 Temporarily not available
Via: SIP/2.0/UDP 192.168.50.33:26606;branch=z9hG4bK-d8754z-ec94f27cc3f2726f-1—d8754z-;rport;received=192.168.50.33
To: "4001"sip:4001@192.168.50.112:5065;tag=AFDF116B-066E-4C3F-80CC-13F9078E9CB4-15
From: sip:87654@192.168.50.112:5065;tag=794ac743
Call-ID: NjRmMjZlYjI3MDM0MzUyZjgyYjQxODljNWRhMWQ3YzA
CSeq: 1 INVITE
Warning: 399 192.168.50.110 “No matching resources for this service type [Service Type: voicexml]”
Content-Length: 0

cavagnaro · April 22, 2014, 6:17pm

The answer is there. READ your logs, not just scan:

Cannot allocate a resource type matching service type [Service Type: voicexml]

so you need to create a…service…of type…???

sudheerreddy.alluru1 · April 22, 2014, 6:31pm

You mean to create Resource Group by enabling VXML service type. If yes, i have already created RG for this service.
Regards,
Sudheer

cavagnaro · April 22, 2014, 7:19pm

“Service of type”…come on…you know the answer…
As you create a service of type Treatment for example…