two INVITEs frmom ACM do SM

[robertklusek]

Hello
I'wondering if somebody help us with generation second invite after OK when call started from CM phone to SM?
INVITE sip:[email protected]:5060;transport=tcp SIP/2.0

From: <sip:[email protected]:5060>;tag=005D4F1C-CE7D-1315-9F97-0100007FAA77-121919

To: <sip:[email protected]>

Call-ID: [email protected]

CSeq: 1 INVITE

Content-Length: 233

Content-Type: application/sdp

Via: SIP/2.0/TCP 10.23.103.96:5060;branch=z9hG4bK005D4F44-CE7D-1315-9F97-0100007FAA77-4951

Contact: <sip:[email protected]:5060;transport=tcp>

X-Genesys-CallInfo: routed

PESEL: 8989898989898

TRANSFER:

CONNIDTM: 010d0240f6e4c0a0

CONNIDAL:

ORIGINATION: T

MSISDN: 600600600

Allow: ACK, BYE, CANCEL, INFO, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE

User-Agent: Avaya CM/R016x.02.0.823.0 AVAYA-SM-6.2.3.0.623006

Accept-Language: en;q=1

Alert-Info: <cid[email protected]>;avaya-cm-alert-type=priority

P-Asserted-Identity: "CMT_TST6001" <sip:[email protected]>

P-Location: SM;origlocname="Warszawa";origsiglocname="Warszawa ";termlocname="Warszawa";termsiglocname="Warsz awa"

Max-Forwards: 66

X-Genesys-CallUUID: 01KD106EFK9HB7SN04000VTAES000050

X-ISCC-CofId: location=SIP;cofid=382

Session-Expires: 1800;refresher=uac

Min-SE: 90

Supported: uui,timer



v=0

o=- 1393942195 1 IN IP4 10.23.221.180

s=-

c=IN IP4 10.23.221.180

b=AS:64

t=0 0

a=avf:avc=n prio=n

a=csup:avf-v0

m=audio 2050 RTP/AVP 8 0 101

a=rtpmap:8 PCMA/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

SIP/2.0 100 Trying

From: <sip:[email protected]:5060>;tag=005D4F1C-CE7D-1315-9F97-0100007FAA77-121919

To: <sip:[email protected]>

Call-ID: [email protected]

CSeq: 1 INVITE

Via: SIP/2.0/TCP 10.23.103.96:5060;branch=z9hG4bK005D4F44-CE7D-1315-9F97-0100007FAA77-4951

Content-Length: 0



SIP/2.0 180 Ringing

From: <sip:[email protected]:5060>;tag=005D4F1C-CE7D-1315-9F97-0100007FAA77-121919

To: <sip:[email protected]>;tag=9E5DD4-1DFE

Call-ID: [email protected]

CSeq: 1 INVITE

Via: SIP/2.0/TCP 10.23.103.96:5060;branch=z9hG4bK005D4F44-CE7D-1315-9F97-0100007FAA77-4951

Date: Tue, 11 Mar 2014 13:40:10 GMT

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER

Allow-Events: telephone-event

Remote-Party-ID: <sip:[email protected]>;party=called;scr een=no;privacy=off

Contact: <sip:[email protected]:5060;transport=tcp>

Server: Cisco-SIPGateway/IOS-12.x

Content-Length: 0



SIP/2.0 200 OK

From: <sip:[email protected]:5060>;tag=005D4F1C-CE7D-1315-9F97-0100007FAA77-121919

To: <sip:[email protected]>;tag=9E5DD4-1DFE

Call-ID: [email protected]

CSeq: 1 INVITE

Via: SIP/2.0/TCP 10.23.103.96:5060;branch=z9hG4bK005D4F44-CE7D-1315-9F97-0100007FAA77-4951

Date: Tue, 11 Mar 2014 13:40:10 GMT

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER

Allow-Events: telephone-event

Remote-Party-ID: <sip:[email protected]>;party=called;scr een=no;privacy=off

Contact: <sip:[email protected]:5060;transport=tcp>

Supported: replaces

Supported: sdp-anat

Supported: timer

Server: Cisco-SIPGateway/IOS-12.x

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 238



v=0

o=CiscoSystemsSIP-GW-UserAgent 3856 4739 IN IP4 10.23.221.241

s=SIP Call

c=IN IP4 10.23.221.241

t=0 0

m=audio 16384 RTP/AVP 8 101

c=IN IP4 10.23.221.241

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

ACK sip:[email protected]:5060;transport=tcp SIP/2.0

From: <sip:[email protected]:5060>;tag=005D4F1C-CE7D-1315-9F97-0100007FAA77-121919

To: <sip:[email protected]>;tag=9E5DD4-1DFE

Call-ID: [email protected]

CSeq: 1 ACK

Content-Length: 0

Via: SIP/2.0/TCP 10.23.103.96:5060;branch=z9hG4bK005D4F44-CE7D-1315-9F97-0100007FAA77-4952

Allow: INVITE, ACK, PRACK, CANCEL, BYE, REFER, INFO, UPDATE, MESSAGE, NOTIFY, OPTIONS

Max-Forwards: 66

User-Agent: Avaya CM/R016x.02.0.823.0 AVAYA-SM-6.2.3.0.623006

P-Location: SM;origlocname="Warszawa";origsiglocname="Warszawa ";termlocname="Warszawa";termsiglocname="Warsz awa"



INVITE sip:[email protected]:5060;transport=tcp SIP/2.0

From: <sip:[email protected]:5060>;tag=005D4F1C-CE7D-1315-9F97-0100007FAA77-121919

To: <sip:[email protected]>;tag=9E5DD4-1DFE

Call-ID: [email protected]

CSeq: 2 INVITE

Content-Length: 0

Via: SIP/2.0/TCP 10.23.103.96:5060;branch=z9hG4bK005D4F44-CE7D-1315-9F97-0100007FAA77-4953

Contact: <sip:[email protected]:5060;transport=tcp>

Allow: INVITE, ACK, PRACK, CANCEL, BYE, REFER, INFO, UPDATE, MESSAGE, NOTIFY, OPTIONS

X-Genesys-CallUUID: 01KD106EFK9HB7SN04000VTAES000050

User-Agent: Avaya CM/R016x.02.0.823.0 AVAYA-SM-6.2.3.0.623006

Accept-Language: en;q=1

P-Asserted-Identity: "CMT_TST6001" <sip:[email protected]>

P-Location: SM;origlocname="Warszawa";origsiglocname="Warszawa ";termlocname="Warszawa";termsiglocname="Warsz awa"

Max-Forwards: 66

Session-Expires: 1800;refresher=uac

Min-SE: 90

Supported: timer



SIP/2.0 100 Trying

From: <sip:[email protected]:5060>;tag=005D4F1C-CE7D-1315-9F97-0100007FAA77-121919

To: <sip:[email protected]>;tag=9E5DD4-1DFE

Call-ID: [email protected]

CSeq: 2 INVITE

Via: SIP/2.0/TCP 10.23.103.96:5060;branch=z9hG4bK005D4F44-CE7D-1315-9F97-0100007FAA77-4953

Content-Length: 0

[aviswanathan]

Second INVITE(no SDP) from CM is to shuffle the call. The far end on receving this invite need to send a 200 OK with SDP(with far end IP for media). Then CM will send a ACK(SDP with Phone IP for media)

[robertklusek]

ok is works

[rschira]

Hi aviswanathan ,

what do you mean by 'shuffle' the call?
Thanks.