Nailed up configuration with Avaya +SES

sitto · August 22, 2012, 11:52am

Hi,
I am configuring the nailed up of Genesys SIP Server 8.1 to interface with Avaya.

Environment:

Genesys SIP Server 8.1
Avaya act as Gateway
Using Avaya Phone for nailed up
Genesys SIP Server 8.1 interface with Avaya SES

Genesys SIP Server can set up the nailed up connection as 2 methods:

SIP Server calls the agent to start a session
The agent calls the contact center to start a session

It works fine for 2nd method because I created a routing strategy for agent to park call first.
But I got an issue for 1st method as Genesys SIP Server try to make a call to a nailed up extension but got some error as shown in the log file below:

18:58:42.122: Sending [0,UDP] 970 bytes to 10.100.6.41:5060 >>>>>
INVITE sip:87312@10.100.6.41:5060 SIP/2.0
From: “Anonymous” <sip:[email removed]>;tag=1B702C33-D41E-424F-8513-0C44A9A60BF8-4
To: sip:7800@10.100.72.42:5060
Call-ID: 23C18A27-B42E-404B-BF7D-AEE1521E4B10-3@10.100.72.42
CSeq: 1 INVITE
Content-Length: 195
Content-Type: application/sdp
Via: SIP/2.0/UDP 10.100.72.42:5060;branch=z9hG4bKDC64F3EE-94EA-462D-8771-98589516352C-3
Contact: sip:anonymous@10.100.72.42:5060
X-Genesys-CallInfo: routed
Allow: ACK, BYE, CANCEL, INFO, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE
P-Asserted-Identity: “Tanarat,Utogsiri” sip:10.100.6.10
Privacy: id
Max-Forwards: 69
X-Genesys-CallUUID: EJSK0PFV5H0QN3JLKSCA76CAHG000001
Session-Expires: 1800;refresher=uac
Min-SE: 90
Supported: uui,100rel,timer

v=0
o=- 1345636715 1 IN IP4 10.100.70.42
s=-
c=IN IP4 10.100.70.42
b=AS:64
t=0 0
m=audio 2058 RTP/AVP 18 127
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:127 telephone-event/8000

18:58:42.122: SipDialog: event SEND_INVITE, t=5, s=2, r=7, m=03491f9c
18:58:42.122 SIPCONN(87312): HandleSipDialogEvent(SEND_INVITE) - filtered
18:58:42.122 SIPCONN(87312): sdp state SDP_STATE_NULL, event SDP_OFFER_SENT
18:58:42.122 SIPCONN(87312): new sdp state SDP_OFFER_SENT, event SDP_OFFER_SENT
– thisCall by party
18:58:42.122 — CIFace::Request —
18:58:42.122 – deleted: CRequest@3498960 RequestPrivateService-IW_1801[468]/6
18:58:42.122: $-TLIB:CTI:Unknown:0:758

18:58:42.122: $+NET:SIP::0:0
18:58:42.122: SIPTR: Received [0,UDP] 426 bytes from 10.100.6.41:5060 <<<<<
SIP/2.0 100 Trying
From: “Anonymous” <sip:[email removed]>;tag=1B702C33-D41E-424F-8513-0C44A9A60BF8-4
To: sip:7800@10.100.72.42:5060
Call-ID: 23C18A27-B42E-404B-BF7D-AEE1521E4B10-3@10.100.72.42
CSeq: 1 INVITE
Via: SIP/2.0/UDP 10.100.72.42:5060;received=10.100.72.42;branch=z9hG4bKDC64F3EE-94EA-462D-8771-98589516352C-3
Content-Length: 0
Organization: 10.100.6.41
Server: Avaya SIP Enablement Services

18:58:42.122: SipDialog: event CALLING_RESPROV, t=5, s=2, r=5, m=03491f9c
18:58:42.122 SIPCONN(87312): HandleSipDialogEvent(CALLING_RESPROV)
18:58:42.122 SIPCONN(87312): Capabilities 61013f
18:58:42.122 SIPCONN(87312): reliable=0
18:58:42.122 SIPCONN(87312): store remote content
18:58:42.122 SIPCONN(87312): store remote content - trying ignored
18:58:42.122 SIPCONN(87312): Trying Timer for 32000 mlsec started…
18:58:42.122: $-NET:SIP::0:63

18:58:42.137: $+NET:SIP::0:0
18:58:42.137: SIPTR: Received [0,UDP] 493 bytes from 10.100.6.41:5060 <<<<<
SIP/2.0 403 Forbidden
From: “Anonymous” <sip:[email removed]>;tag=1B702C33-D41E-424F-8513-0C44A9A60BF8-4
To: sip:7800@10.100.72.42:5060;tag=C88E5D19E9CFE947C69B47B3A41C438F1345611140160114
Call-ID: 23C18A27-B42E-404B-BF7D-AEE1521E4B10-3@10.100.72.42
CSeq: 1 INVITE
Via: SIP/2.0/UDP 10.100.72.42:5060;psrrposn=1;received=10.100.72.42;branch=z9hG4bKDC64F3EE-94EA-462D-8771-98589516352C-3
Content-Length: 0
Organization: 10.100.6.41
Server: Avaya SIP Enablement Services

18:58:42.137: Sending [0,UDP] 426 bytes to 10.100.6.41:5060 >>>>>
ACK sip:87312@10.100.6.41:5060 SIP/2.0
Via: SIP/2.0/UDP 10.100.72.42:5060;branch=z9hG4bKDC64F3EE-94EA-462D-8771-98589516352C-3
From: “Anonymous” <sip:[email removed]>;tag=1B702C33-D41E-424F-8513-0C44A9A60BF8-4
To: sip:7800@10.100.72.42:5060;tag=C88E5D19E9CFE947C69B47B3A41C438F1345611140160114
Call-ID: 23C18A27-B42E-404B-BF7D-AEE1521E4B10-3@10.100.72.42
CSeq: 1 ACK
Max-Forwards: 69
Content-Length: 0

18:58:42.137: SipDialog: event CALLING_RESREJECT, t=5, s=9, r=5, m=03491f9c
18:58:42.137 SIPCONN(87312): HandleSipDialogEvent(CALLING_RESREJECT)
18:58:42.137 SIPCONN(87312): Trying Timer deleted…
18:58:42.137 SIPCONN(87312): ConvertResponse: 403
18:58:42.137 SIPCONN(87312): ConvertResponse: no conversion configured for 403
18:58:42.137 SIPCONN(87312): Failed to obtain Media Server ID
18:58:42.137 SIPCONN(87312): state e:4,p:0,s:6,c:8,rc:403,m:1
18:58:42.137 SIPPARTY(87312): 4 verify update of party-connection state N-F
18:58:42.137 SIPPARTY(87312): 4 update party-connection state N-F
18:58:42.137 – thisCall by party
18:58:42.137 SetContext: for party 7800.3489f38-348aaa8:1
18:58:42.137 +++ CIFace::Event +++
+++ Pre-event +++
Type EventError
Devices: <7800/7800> <-/-> <-/->
Calls: 1/00850212e685d001/1.348aaa8/c:2/r:4 0/none
Parties: D7800/7800.3489f38-348aaa8:1/l:2/r:2/Queued,RtRequest,Destination
none
none
Cause: Busy/3, Info: 231
Flags: divert=0 hook=0 postCall=0 active=1 moveAll=1 callType=1 hideOtherPi=0 InternalOther=0
— Pre-event —
+++ Error +++
– thisCall by party
18:58:42.137 Trc 36002 Request rejected: error code 231(DN is busy)
@18:58:42.1370 [0] 8.1.000.81 send_to_client: message EventError
(DN is busy)
AttributeEventSequenceNumber 000000000000008d
AttributeTimeinuSecs 137000
AttributeTimeinSecs 1345636722 (18:58:42)
AttributeExtensions [322] 00 0f 00 00..
‘CUSTOMER_ID’ ‘Resources’
‘AGENT’ ‘1801’
‘PLACE’ ‘87312’
‘DN’ ‘87312’
‘ACCESS’ ‘87312’
‘SWITCH’ ‘SIPSwitch’
‘NVQ’ 1
‘TARGET’ ‘[email removed]’
‘iscc-ar-agent-dn’ ‘87312’
‘iscc-ar-agent-id’ ‘1801’
‘iscc-ar-place’ ‘87312’
‘iscc-ar-duration’ 15000
‘iscc-ar-priority’ 0
‘iscc-ar-priority-1’ 16
‘iscc-ar-priority-2’ 625
AttributeErrorCode 231
AttributeErrorMessage ‘DN is busy’
AttributeReferenceID 125
AttributeReason [14] 00 01 01 00..
‘RTR’ 118
AttributeRouteType 0 (RouteTypeUnknown)
AttributeOtherDN ‘87312’
AttributeConnID 00850212e685d001
AttributeThisDN ‘7800’
AttributeClientID 5
18:58:42.137 Int 04545 Interaction message “EventError” sent to 456 (“urs”)

Do you have any idea?
Thanks,
Sitto

bublepaw · August 22, 2012, 1:03pm

I am more SIPS expert but for me it looks like You didn’t configure proper routing on Avaya side. Avaya can send call to SIPS but when call arrives from SIPS to Avaya, Avaya is not able to send call to its final destination 87312

Pawel

Kubig · August 22, 2012, 1:29pm

It is a problem on Avaya site,because Avaya answer with SIP cause 403 Forbidden - that means server understood a request,but is refusing to fulfill it. Check the settings of DN on Avaya site.

sitto · August 22, 2012, 3:45pm

Many thanks both of you!!!

I will discuss with Avaya engineer tomorrow. Do you have any suggestion on Avaya configuration?

Regards,
Sitto

bublepaw · August 22, 2012, 6:57pm

One thing to check is codec configuration - I can see that You are only sending G729. You could extend codec list to include G711 which is by default always supported

sitto · August 23, 2012, 5:40am

Yes, Avaya just added it and it still faces the issue.

Fra · August 23, 2012, 9:46am

Check the signalling trunk on the Avaya, specifically whether the far end has been defined properly.

Fra

sitto · August 23, 2012, 11:24am

Thanks for all,

Just update I added the configuration in a nailed up extension as override-domain-from= or , it worked but only call is dialed from Avaya extension but it still is not working if I call the number via PSTN.

It seems like some security configuration on Avaya rejected our call.