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Change REFER to INVITE as ITSP does not support REFER message

Hi All,

 

I might be searching the wrong word but I cannot seem to get any answer on the following matter.

My ITSP does not support REFER thus when a call is transferred the CUBE receives the REFER message, but this needs to be converted into an INVITE for a new call leg as the CUBE needs to make a new call to the TRANSFERRED number.

So what is happening is:

Caller phones --> Genesys PureCloud number --> Call is answered and now Genesys Purecloud Agent Transfers the call to an external Party --> via a Cisco Cube which routes the call to the ITSP (Whom does not support REFER)

 

This below in short is what I get on the CUBE for the TRANSFER attempt. 

Do I apply some sort of header manipulation on the incoming dial-peer from Genesys Purecloud to change the REFER to an INVITE thus releasing the call from Genesys Purecloud and establish a new call from the Cube and remain in the call signaling until the transferred call also ended? Is this possible at all?

 

Received:
REFER sip:02122XXXXX@172.24.XXX.XXX:5060 SIP/2.0
To: "02122XXXXX" <sip:02122XXXXX@172.24.XXX.XXX>;tag=63A2FD25-1E93
From: <sip:0992XXXXX@172.24.XXX.XXX>;tag=6ug6Fvc
Call-ID: 868AD6C6-6AC11EB-887C98C3-2006AADE@172.24.11.60
Max-Forwards: 70
Via: SIP/2.0/UDP 172.24.XXX.XXX:5060;branch=z9hG4bK2243894105SU5JTi9pM3NpcC4t_2026709845_
Refer-To: <sip:0930XXXXX@172.24.XXX.XXX;user=phone>
Refer-Sub: true
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO
Supported: norefersub, timer
Accept: application/sdp, application/dtmf-relay
Contact: <sip:0992XXXXX@172.24.XXX.XXX:5060;transport=udp>
x-inin-cnv: f548f528-6090-4a42-84f9-2e0cd1140cc9
CSeq: 1 REFER
User-Agent: ININ-EDGE/1.0.0.9432
Content-Length: 0

 

Sent:
SIP/2.0 202 Accepted
Via: SIP/2.0/UDP 172.24.XXX.XXX:5060;branch=z9hG4bK2243894105SU5JTi9pM3NpcC4t_2026709845_
From: <sip:0992XXXXX@172.24.XXX.XXX>;tag=6ug6Fvc
To: "02122XXXXX" <sip:02122XXXXX@172.24.XXX.XXX>;tag=63A2FD25-1E93
Date: Tue, 06 Oct 2020 01:46:08 GMT
Call-ID: 868AD6C6-6AC11EB-887C98C3-2006AADE@172.24.XXX.XXX
Server: Cisco-SIPGateway/IOS-16.6.4
CSeq: 1 REFER
Content-Length: 0
Contact: <sip:02122XXXXX@172.24.XXX.XXX:5060>


Sent:
NOTIFY sip:0992XXXXX@172.24.XXX.XXX:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 172.24.XXX.XXX:5060;branch=z9hG4bK14F413199B
From: "02122XXXXX" <sip:02122XXXXX@172.24.XXX.XXX>;tag=63A2FD25-1E93
To: <sip:0992XXXXX@172.24.XXX.XXX>;tag=6ug6Fvc
Call-ID: 868AD6C6-6AC11EB-887C98C3-2006AADE@172.24.XXX.XXX
CSeq: 103 NOTIFY
Max-Forwards: 70
Date: Tue, 06 Oct 2020 01:46:08 GMT
User-Agent: Cisco-SIPGateway/IOS-16.6.4
Event: refer
Subscription-State: pending;expires=180
Contact: <sip:02122XXXXX@172.24.XXX.XXX:5060>
P-Asserted-Identity: "02122XXXXX" <sip:02122XXXXX@172.24.XXX.XXX>
Content-Type: message/sipfrag
Content-Length: 22

SIP/2.0 100 Trying


Oct 6 14:46:08.325 NZDT: //6037994/868950628876/CCAPI/ccSaveDialpeerTag:
Outgoing Dial-peer=0

 

Sent:
NOTIFY sip:0992XXXXX@172.24.XXX.XXX:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 172.24.XXX.XXX:5060;branch=z9hG4bK14F4141CF5
From: "02122XXXXX" <sip:02122XXXXX@172.24.XXX.XXX>;tag=63A2FD25-1E93
To: <sip:0992XXXXX@172.24.XXX.XXX>;tag=6ug6Fvc
Call-ID: 868AD6C6-6AC11EB-887C98C3-2006AADE@172.24.XXX.XXX
CSeq: 104 NOTIFY
Max-Forwards: 70
Date: Tue, 06 Oct 2020 01:46:08 GMT
User-Agent: Cisco-SIPGateway/IOS-16.6.4
Event: refer
Subscription-State: terminated;reason=noresource
Contact: <sip:02122XXXXX@172.24.XXX.XXX:5060>
P-Asserted-Identity: "02122XXXXX" <sip:02122XXXXX@172.24.XXX.XXX>
Content-Type: message/sipfrag
Content-Length: 25

SIP/2.0 404 Not Found

Best Regards
1 Accepted Solution

Accepted Solutions

Issue was resolved by TAC

Best Regards

View solution in original post

3 Replies 3

Issue was resolved by TAC

Best Regards

For the benefit of the community maybe you could provide a little bit of information about what TAC did to solve this?



Response Signature


Sure here in short what we have done:

 

So we first had to create separate dial-peer matching the REFER-TO number to go to the ISP.(SIP header manipulation had no effect)

Also create a Translation Pattern to change the calling number to a DDI the ISP allow over the SIP trunk.

Oct 6 14:46:08.325 NZDT: //6037994/868950628876/CCAPI/ccSaveDialpeerTag:
Outgoing Dial-peer=0

 

Sent:
NOTIFY sip:0992XXXXX@172.24.XXX.XXX:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 172.24.XXX.XXX:5060;branch=z9hG4bK14F4141CF5
From: "02122XXXXX" <sip:02122XXXXX@172.24.XXX.XXX>;tag=63A2FD25-1E93
To: <sip:0992XXXXX@172.24.XXX.XXX>;tag=6ug6Fvc
Call-ID: 868AD6C6-6AC11EB-887C98C3-2006AADE@172.24.XXX.XXX
CSeq: 104 NOTIFY
Max-Forwards: 70
Date: Tue, 06 Oct 2020 01:46:08 GMT
User-Agent: Cisco-SIPGateway/IOS-16.6.4
Event: refer
Subscription-State: terminated;reason=noresource
Contact: <sip:02122XXXXX@172.24.XXX.XXX:5060>
P-Asserted-Identity: "02122XXXXX" <sip:02122XXXXX@172.24.XXX.XXX>
Content-Type: message/sipfrag
Content-Length: 25

SIP/2.0 404 Not Found

 

Once we have done this the calls still failed:

 

++ The debug ccsip all shows that it was failing due to the below error:

                1797460: Oct  7 19:08:00.483 NZDT: //6075929/000000000000/SIP/Error/ccsip_call_setup_request:

                 It is a triggered INVITE, SDP Passthrough not supported, Initiating Disconnect

 

++ The CUBE was configured as below: (It is a shared CUBE so pass-thru content sdp cannot be disabled globally)

                voice service voip

                sip

                  pass-thru content sdp

 

++ These calls are working fine after configuring as below:

                dial-peer voice 121 voip (Outbound for REFER_TO to ISP) 

                no voice-class sip pass-thru content sdp 

                dial-peer voice 200 voip (Inbound to PureCloud)

                no voice-class sip pass-thru content sdp

                dial-peer voice 100 voip (Outbound for REFER_TO to ISP)

                no voice-class sip pass-thru content sdp

Best Regards
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