Add Caller name to SIP header

Hi,
I have just setup a new SIP trunk with a local provider and we need to have the caller name added to the SIP packets. Here is our sip profile.
voice class sip-profiles 1
request INVITE sip-header To modify "@.*>" "@siptrunking.bell.ca>"
request INVITE sip-header Contact modify "@" ";tgrp=calgary_01_403XXXXXXX;trunk-context=siptrunking.bell.ca@"
request INVITE sip-header SIP-Req-URI modify "XXX.XXX.XXX.XXX" "siptrunking.bell.ca"
request INVITE sip-header P-Asserted-Identity modify "(@.*)>" "@domain.com;user=phone>"
request INVITE sip-header Diversion modify "@.*>" "@siptrunking.bell.ca;user=phone>"
request INVITE sip-header From modify "(@.*)>" "@domain.com;user=phone>"
Calls are going through with this and the following packet is being sent out.
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK1E09D1DFF
From: <sip:[email protected];user=phone>;tag=1090D060-117C
To: <sip:[email protected]>
Date: Fri, 09 Jan 2015 17:36:39 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 900
Cisco-Guid: 0015978767-2802909515-2130729220-0167905803
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1420824999
Contact: <sip:XXXXXXXXXX;tgrp=calgary_01_403XXXXXXX;trunk-context=[email protected]:5060>
Expires: 180
Allow-Events: telephone-event
P-Asserted-Identity: <sip:[email protected];user=phone>
Content-Length: 0
We need to have the From and the P-Asserted-Identity changed to
"Caller Name" <sip:[email protected];user=phone>
Does anyone know how to do this?

have you set the name for each DN? if so, it has to go out thru SIP Trunk.
if you want to set same caller name for all the calls, you can do it by SIP profile as below
request INVITE sip-header Remote-Party-ID modify "\"(.*)\" " "\"CallerName\" "
request INVITE sip-header From modify "\"(.*)\" " "\"Caller Name\" "

Similar Messages

How to assign the dynamic value of PV to Sip Header in ICM?

Hi everybody,
I would like to ask your help, please. We are working on Temporary IVR Handoff (ICM+CVP). I need to add/modify a customer Sip header in ICM transfer script. The value for that header is dynamic and stored in the one of ICM Call Peripherial Variables (PV9, for examle).
Is it possible somehow to assign the value of that PV9 to Sip header (Set Variable Call.SipHeader)? I tried to do it but unfortunally without any success. I can assign any static string, but how to assign the value of the PV??? That's the question...
For static string assignment the syntax of the Set Variable Node (Call.SipHeader) looks like that:
"IVR-Handoff~add~It's Cisco"
and it works fine.
For dynamic value (PV9) I tried:
"IVR-Handoff~add~Call.PeripherialVariable9" - it add Call.PeripherialVariable9 as a string, but not it's value;
"IVR-Handoff"~add~Call.PeripherialVariable9 - returns a syntax error;
Call.PeripherialVariable9 - no Sip header is added.
What do you think? Is it doable at all?
Any ideas, answers or examples how to do it would be much appreciated.
Thank you in advance.
Dmitriy.

Hi Senthil,
Yes! It works, but with the little difference. The right answer is:
"IVR-Handoff~add~"&Call.PeripherialVariable9
(Call.PeripherialVariable9 is without quotes, otherwise it insert the name of the variable, but not it's value)
The other solution is shown here (thanks to Paul Tindall):
http://developer.cisco.com/web/cvp/forums/-/message_boards/message/15627744?p_p_auth=6psgR8ML
concatenate("IVR-Handoff~add~",Call.PeripherialVariable9)
Thank you so much for the idea! I very appreciate your help and vote you.

How to add coloum name in a exel sheet download file?

Dear all,
I amdownloading a excel sheet from a internal table.I am getting all data correctly.
but now i want to add coloumn name for each coloumn.
How can I do it?
i had try from work area and insert at first index but not get proper result.....
so if possible plz suggest sampl code also.
Regards
Ricky

Hi Ricky Maheswari,
                              I will send a sample code for u.check it once.I execute the below code that is executed successfully.
In my report u have check these things carefully "PEFORM APPEND_HEADER " and SELECT STATEMENT(I use appending table stmt there).Then ur problem will be resolved.
code:
*& Report YBDC_DOWNLOAD_MM01_XLS                                      *
*& DEVELOPER   : KIRAN KUMAR.G                                         *
*& PURPOSE     : FETCH DATA FROM DB AND PLACE THEM IN .XLS FILE        *
*& CREATION DT : 2/12/2007                                             *
*& REQUEST     : ERPK900035                                            *
*& NOTE        : MENTION PATH & MENTION FILE.XLS IN THE SELE-SCREEN    *
REPORT ybdc_download_mm01_xls MESSAGE-ID zbdcmsg.
Tables
TABLES: mara, "General Material Data
        makt. "Material Descriptions
Global Variables
DATA: gv_path TYPE string. "Hold Path Selection Information
Internal Table
DATA : BEGIN OF gt_data OCCURS 0,
        matnr(20),   " Material Number
        mbrsh(20),   " Industry Sector
        mtart(20),   " Material Type
        meins(20),   " Base Unit Of Measure
        maktx(20),   " Material Description
       END OF gt_data.
Selection-Screen
SELECTION-SCREEN : BEGIN OF BLOCK b1 WITH FRAME TITLE text-001.
PARAMETERS : p_file(90).
SELECTION-SCREEN : END OF BLOCK b1.
SELECTION-SCREEN : BEGIN OF BLOCK b2 WITH FRAME TITLE text-002.
SELECT-OPTIONS : s_matnr FOR mara-matnr,
                 s_mbrsh FOR mara-mbrsh,
                 s_mtart FOR mara-mtart,
                 s_maktx FOR makt-maktx,
                 s_meins FOR mara-meins.
SELECTION-SCREEN : END OF BLOCK b2.
Initialization
INITIALIZATION.
PERFORM initial.
Placing A File In The Directory
AT SELECTION-SCREEN ON VALUE-REQUEST FOR p_file.
PERFORM path_directory.
START-OF-SELECTION.
Append Some Header Information To XLS File
PERFORM append_header.
Fetching The Data
PERFORM fecth_data.
END-OF-SELECTION.
GUI_DOWNLOAD
PERFORM gui_download.
*&      Form path_directory
      text
--> p1        text
<-- p2        text
FORM path_directory .
CALL METHOD cl_gui_frontend_services=>directory_browse
    EXPORTING
      window_title         = 'Download A File'
      initial_folder       = 'c:/'
    CHANGING
      selected_folder      = gv_path
    EXCEPTIONS
      cntl_error           = 1
      error_no_gui         = 2
      not_supported_by_gui = 3
      OTHERS               = 4.
IF sy-subrc <> 0.
MESSAGE ID SY-MSGID TYPE SY-MSGTY NUMBER SY-MSGNO
           WITH SY-MSGV1 SY-MSGV2 SY-MSGV3 SY-MSGV4.
ENDIF.
CALL METHOD cl_gui_cfw=>flush
    EXCEPTIONS
      cntl_system_error = 1
      cntl_error        = 2
      OTHERS            = 3.
IF sy-subrc <> 0.
MESSAGE ID SY-MSGID TYPE SY-MSGTY NUMBER SY-MSGNO
           WITH SY-MSGV1 SY-MSGV2 SY-MSGV3 SY-MSGV4.
ENDIF.
p_file = gv_path.
CLEAR gv_path.
ENDFORM.                    " path_directory
*&      Form fecth_data
      text
--> p1        text
<-- p2        text
FORM fecth_data .
SELECT a~matnr
         a~mbrsh
         a~mtart
         a~meins
         b~maktx
    FROM mara AS a
   INNER JOIN makt AS b ON amatnr = bmatnr
   APPENDING TABLE gt_data
   WHERE a~matnr IN s_matnr
   AND   a~mbrsh IN s_mbrsh
   AND   a~mtart IN s_mtart
   AND   b~maktx IN s_maktx
   AND   a~meins IN s_meins.
IF sy-subrc = 0.
    MESSAGE s000.
ENDIF.
ENDFORM.                    " fecth_data
*&      Form gui_download
      text
--> p1        text
<-- p2        text
FORM gui_download .
gv_path = p_file.
CALL FUNCTION 'GUI_DOWNLOAD'
    EXPORTING
   BIN_FILESIZE                  =
     filename                      = gv_path
     filetype                      = 'ASC'
   APPEND                        = ' '
     write_field_separator         = 'X'
   HEADER                        = '00'
   TRUNC_TRAILING_BLANKS         = ' '
   WRITE_LF                      = 'X'
   COL_SELECT                    = ' '
   COL_SELECT_MASK               = ' '
   DAT_MODE                      = ' '
IMPORTING
   FILELENGTH                    =
    TABLES
     data_tab                      = gt_data
   EXCEPTIONS
     file_write_error              = 1
     no_batch                      = 2
     gui_refuse_filetransfer       = 3
     invalid_type                  = 4
     no_authority                  = 5
     unknown_error                 = 6
     header_not_allowed            = 7
     separator_not_allowed         = 8
     filesize_not_allowed          = 9
     header_too_long               = 10
     dp_error_create               = 11
     dp_error_send                 = 12
     dp_error_write                = 13
     unknown_dp_error              = 14
     access_denied                 = 15
     dp_out_of_memory              = 16
     disk_full                     = 17
     dp_timeout                    = 18
     file_not_found                = 19
     dataprovider_exception        = 20
     control_flush_error           = 21
     OTHERS                        = 22
IF sy-subrc <> 0.
MESSAGE ID SY-MSGID TYPE SY-MSGTY NUMBER SY-MSGNO
        WITH SY-MSGV1 SY-MSGV2 SY-MSGV3 SY-MSGV4.
ENDIF.
ENDFORM.                    " gui_download
*&      Form append_header
      text
--> p1        text
<-- p2        text
FORM append_header .
REFRESH : gt_data.
CLEAR   : gt_data.
gt_data-matnr = 'MATERIAL NUMBER'.
gt_data-mbrsh = 'INDUSTRY SECTOR'.
gt_data-mtart = 'MATERIAL TYPE'.
gt_data-maktx = 'MATERIAL DESCRIPTION'.
gt_data-meins = 'BASE UNIT OF MEASURE'.
APPEND gt_data.
CLEAR gt_data.
ENDFORM.                    " append_header
*&      Form initial
      text
--> p1        text
<-- p2        text
FORM initial .
s_matnr-sign   = 'I'.
s_matnr-option = 'BT'.
s_matnr-low    = '800'.
s_matnr-high   = '100-200'.
APPEND s_matnr.
s_mbrsh-sign   = 'I'.
s_mbrsh-option = 'BT'.
s_mbrsh-low    = 'M'.
s_mbrsh-high   = ''.
APPEND s_mbrsh.
s_mtart-sign   = 'I'.
s_mtart-option = 'BT'.
s_mtart-low    = 'FERT'.
s_mtart-high   = 'HALB'.
APPEND s_mtart.
s_maktx-sign   = 'I'.
s_maktx-option = 'BT'.
s_maktx-low    = 'IRON'.
s_maktx-high   = 'STEEL'.
APPEND s_maktx.
s_meins-sign   = 'I'.
s_meins-option = 'BT'.
s_meins-low    = 'CM'.
s_meins-high   = 'KG'.
APPEND s_meins.
ENDFORM.                    " initial
Award points if helpful.
Kiran Kumar.G
                 Have a Nice Day..

Disable calling name presentation on SPA-3102

Hi,
If I send a SIP INVITE to my SPA-3102, where the From header is like this -- (spaces inserted to stop the forum software treating it as an email address -- they're not there in the real invite)
From: Caller Name <01234567890 @ my.sip.server.net>;tag=as4b617ab1
-- the SPA-3102 generates a Caller ID spill on its FXS port with 'Caller Name' as the calling name, and '01234567890' as the calling number. That's all well and good.
If the From: header doesn't have a caller name, but is like this instead --
From: <01234567890 @ my.sip.server.net>;tag=as4b617ab1
-- the box sets the calling name to be 01234567890 as well.
Is there any way to turn that off, and have the SPA just not present a calling name at all?
If not, no bother! I'm just trying to get my box to behave a little more like BT with regards to caller ID presentation -- they don't ever send a reason for no calling *name*, but if the calling number is withheld or unavailable they will set the calling name to Withheld or Unavailable -- and set a reason for no calling number.
Many thanks!
Martin
Message was edited by: Martin Thorpe -- hopefully removed the auto-'email address' tagging! (Argh, no, it didn't. Bodged a different way.)

Hi Lindsey,
Thanks for the quick response. Here's a complete SIP invite -- I've changed the telephone number and put spaces around @ signs again, but everything else is unmodified.
INVITE sip:spa-line1 @ 81.2.113.115:5060 SIP/2.0
Via: SIP/2.0/UDP 81.187.239.177:5060;branch=z9hG4bK4062e0e9;rport
Max-Forwards: 70
From: ;tag=as75e22314
To:
Contact:
Call-ID: 445f75c33908fff74829a514159e9946 @ sentry.met24.net
CSeq: 102 INVITE
User-Agent: Asterisk
Date: Mon, 29 Oct 2012 19:51:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 286
So there is a contact field in there as well.
That's from a slightly patched Asterisk server, which doesn't put a calling name in if it's blank -- by default if you didn't set a calling name, Asterisk will also set the calling name from the calling number and you'd get this instead:
From: "01234567890" ;tag=as54c7bb08
I've done product management myself so I know one customer asking for it to work a little differently (as opposed to it doing something wrong!) isn't going to make a change -- that's no problem at all. If it were to be changed, I'd rather the ATA didn't generate a calling name field in the CLID spill at all, rather than 'Unknown'. But hey, that's just my opinion!
For the avoidance of doubt, the ATA is always generating the calling *number* field in the CLID spill correctly.
Thanks again!
All the best,
Martin

Show Subreport Name in the header part of the main Report subreport wise

Hi,
I want to create a Report which contains a collection of Subreport.
In header part of the main report I have one expression which shows specific text for specific subreport.
In Detail
I have three subreport named as
Sureport1
Subreport2
Subreport3
I have given page name for all the three subreport as subreport name.
Now I create one Main report. Where I drag one table and delete the group.
I created three rows in this tablix.
And in each textbox I have inserted one rectangle.
and place all the three subreport.
and here my expression is mentioned below.
=IIf(Globals!PageName Like "Subreport1","Subreport1",
IIf(Globals!PageName Like "Subreport2","Subreport2",
IIf(Globals!PageName Like "Subreport3","Subreport3","")))
This expression is working when I am placing all the subreports inside separate rectangle but not working when I am placing all three subreports in one tablix.
For both case it is showing in report designer window but in second case it is not showing in the pdf.
Thanks,
Aswini Rout

Hi,
As the description, I understand that you want to display the subreports’ name on the page header of the main report. When the page display a subreport, display the corresponding subreport name in the header part of the main report.
Base on my understanding, you should have page break on each subreport(suppose you add the subreports without Rectangle) since there is no group in the main report and there should not be a page break in the main report. If I have misunderstood, please correct
me.
I was not able to reproduce the issue currently. How did you define the page name of your main report. If it is possible, please upload your rdl file to your OneDrive and share the link here. You can also capture the screenshot so that I can understand the
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TechNet Community Support

CUCM 8.6 Dropped call transfers involving SIP phones

Hi All,
I am a developer who has been tasked with figuring out why call transfers are being dropped by Cisco CUCM when the original call comes from a SIP phone. This scenario works:
Cisco phone calls another Cisco phone, which transfers the original call to a SIP phone
These scenarios do not work:
SIP phone calls Cisco phone, which transfers the original call to another Cisco phone
SIP phone calls Cisco phone, which transfers the original call to another SIP phone
I have researched the Call Manager traces to the best of my ability, and I see some info in there that could potentially point to the source of the problem. I am just unable to understand what the trace means:
10:23:08.672 |//SIP/SIPCdpc(1,74,2342)/ci=24377698/ccbId=175645/scbId=0/active_CcDisconnReq: ccDisconnReq.onBehalfOf=Media : ccDisconnReq.s.sv=2 : ccDisconnReq.c.cv=47 |1,100,63,1.93259^10.10.10.85^*
10:23:08.672 |//SIP/Stack/Info/0x0/sipConstructContainerContext #### Created container=0xb0b42f58|1,100,71,1.1^*^*
10:23:08.672 |//SIP/SIPCdpc(1,74,2342)/ci=24377698/ccbId=175645/scbId=0/appendReasonHdr: appendReasonHdr - Invalid Disconnect Cause(cause=47), No Reason Header Appended|1,100,63,1.93259^10.10.10.85^*
10:23:08.672 |//SIP/SIPCdpc(1,74,2342)/ci=24377698/ccbId=175645/scbId=0/appendRPIDHdrForOriginalCalledParty: SIP device does not Support Orig Dialled Phone nego: 0|1,100,63,1.93259^10.10.10.85^*
I have been wondering whether this could be a codec issue, however the SIP phones we are using are configured with the following codecs:
G711U
G711A
G722
ILBC
GSM
and our SIP software is also set to accept the first codec offered by the remote side. It seems from the SIP client logs that G722 is being used as the codec to communicate with the Cisco phones, but perhaps I'm misinterpreting.
I have attached a CUCM trace of a call from a SIP phone (ext. 491) to a Cisco handset (ext. 170) where the Cisco handset attempts to transfer the call to another SIP phone (ext. 492). The trace snippet shown above is from this log.
I would really appreciate it if someone more experienced with VoIP/SIP/CUCM could take a look and offer any ideas on what the issue might be, and also how we might be able to address it. I can try to provide more info about our CUCM configuration if needed.
Thanks in advance!

Leslie, so here is what I found from the traces....
To understand the difference we need to understand how cucm performs call transfers from a sccp signalling point and a sip signalling point
SCCP
When the transfer key is pressed
1. CUCM sends a CloseReceiveChannel and StopMediaTransmission to the IP phone involved in active media (referenced by the callids)
NB, here CUCM updates the call state on the phone to a call state of 8 which is "Hold"
2.CUCM tells the held party to listen MOH from MOH server
3.CUCM establishes newcall leg with the intended transfered destination..Once this call is connected
4.CUCM receives a new Transfer instruction from the transferring phone to connect the held party
5..CUCM sends a CloseReceiveChannel between the held phone and MOH server (to tear down the media)
6. Next CUCM sends a CloseReceiveChannel and StopMediaTransmission to the transfering party & transfered party to remove Xferring party from call
7. finally CUCM sends OpenReceiveChannel between the original called party and the transfered party..and call is done
For SIP signalling. when the first transfer key is pressed
1. CUCM sends invite (re-invite) with an inactive SDP (a=inactive) to indicate a break in media path
2. CUCM sends a Delayed offer to insert MOH or to resume Media stream
NB: CUCM expects a sendrecv offer with SDP to the DO. (NB:if cucm gets an inactive offer SDP in the 200 OK instead of providing a send-recv offer SDP, the media path remains in an inactive state and causes calls to dropcall will drop),CUCM sends an ACK with sendonly to the 200 OK
3.CUCM establishes newcall leg with the intended transfered destination..Once this call is connected
4.CUCM receives a new Transfer instruction from the transferring phone to connect the held party
5. Next CUCM sends a re-invite with an inactive SDP to indicate a break in media path to MOH (in attempt to complete transfer)
6.Next CUCM sends an inactive SDP to indicate a break in media path between transfering party & transfered party to remove Xferring party from call
7. Next CUCM sends a DO re-invite to connect the transfered party. The far end then sends 200 OK with the required SDP to connect the call
Now having explained all of these, we need to look at where the call is failing for SIP-----SCCP----SIP calls without MTP
lets look at succesful SCCP-----SCCP-----SIP without MTP
Point 4 above
++++++++Extension 170 presses the transfer button to connect the two calls (Callid=24378483)+++++++++++++
(0003395) SoftKeyEvent softKeyEvent=4(Trnsfer) lineInstance=1 callReference=24378483
Point 5 above
++++Next CUCM closed the media between extension 160 and MOH server callid=24378480(this is the only active call on this callid)+++
(0003396) CloseReceiveChannel conferenceID=24378480 passThruPartyID=16777845. myIP: IpAddr.type:0 ipv4Addr:0x0a0a0a89(10.10.10.137)
Point 6 Above
+++++Next cucm closes the call between extension 170 and 490 callid=(24378483)++++++++
(0003395) CloseReceiveChannel conferenceID=24378483 passThruPartyID=16777847. myIP: IpAddr.type:0 ipv4Addr:0x0a0a0a8b(10.10.10.139)
(0003395) StopMediaTransmission conferenceID=24378483 passThruPartyID=16777847. myIP: IpAddr.type:0 ipv4Addr:0x0a0a0a8b(10.10.10.139)
Point 6 above for the sip side (since the destination is SIP, to tear down media to SCCP phone, so as to connect the caller to the xfered party)
+++++++Next CUCM sends a re-invite with a=inactive SDP to the sip phone ++++++++++++
//SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.10.104 on port 62220 index 1890
[885626,NET]
INVITE sip:[email protected]:62220;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK23332dbee978
From: ;tag=192115~d8e94532-127d-4dca-bba0-64b1675da032-24378484
o=CiscoSystemsCCM-SIP 192115 2 IN IP4 10.10.10.195
s=SIP Call
c=IN IP4 0.0.0.0
m=audio 24560 RTP/AVP 9 101
a=rtpmap:9 G722/8000
a=ptime:20
a=inactive-----------------------------------------------------Inactive
Still part of Point 6 for SIP signalling
++++++++++++Next sip phone responds with a 200 OK recevonly SDP +++++++++++++++++++
//SIP/SIPTcp/wait_SdlReadRsp: Incoming SIP TCP message from 10.10.10.104 on port 62220 index 1890 with 683 bytes:
[885628,NET]
SIP/2.0 200 OK
v=0
o=- 18077 11099 IN IP4 10.10.10.104
s=yasdjip
c=IN IP4 10.10.10.104
t=0 0
a=ptime:20
a=recvonly-------------------------------------a=recvonly
Finally Point 7 above..
+++++++++++++=Next cucm sends a DO re-invite to extension 492-sip phone++++++++++
//SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.10.104 on port 62220 index 1890
[885630,NET]
INVITE sip:[email protected]:62220;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK233534ffec4a
From: ;tag=192115~d8e94532-127d-4dca-bba0-64b1675da032-24378484
To: ;tag=5B0E9816C2CA6D70F3166FB972EDE4C2
+++++++Next we get a 200 OK from sip phone with sdp=sendrecv+++++++++=
/SIP/SIPTcp/wait_SdlReadRsp: Incoming SIP TCP message from 10.10.10.104 on port 62220 index 1890 with 683 bytes:
[885634,NET]
SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK233534ffec4a
Contact:
From: ;tag=192115~d8e94532-127d-4dca-bba0-64b1675da032-24378484
Call-ID: [email protected]
v=0
o=- 18077 11099 IN IP4 10.10.10.104
s=yasdjip
c=IN IP4 10.10.10.104
t=0 0
m=audio 16574 RTP/AVP 9 101
a=rtpmap:101 TELEPHONE-EVENT/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
+Now CUCM sends an OpenReceiveChannel and start media xmission to sccp phone (callid=24378480) with media parameters of sip phone++++++
(0003396) OpenReceiveChannel conferenceID=24378480 passThruPartyID=16777848 millisecondPacketSize=20 compressionType=6(Media_Payload_G722_64k) RFC2833PayloadType=101 qualifierIn=? sourceIpAddr=IpAddr.type:0 ipAddr:0x0a0a0a68000000000000000000000000(10.10.10.104). myIP: IpAddr.type:0 ipv4Addr:0x0a0a0a89(10.10.10.137)
(0003396) startMediaTransmission conferenceID=24378480 passThruPartyID=16777848 remoteIpAddress=IpAddr.type:0 ipAddr:0x0a0a0a68000000000000000000000000(10.10.10.104)
remotePortNumber=16574 milliSecondPacketSize=20 compressType=6(Media_Payload_G722_64k) RFC2833PayloadType=101 qualifierOut=?. myIP: IpAddr.type:0 ipv4Addr:0x0a0a0a89(10.10.10.137)
+++++++++++=Next Phone sends its ACK+++++++++++++++
(0003396) OpenReceiveChannelAck Status=0, IpAddr=IpAddr.type:0 ipAddr:0x0a0a0a89000000000000000000000000(10.10.10.137), Port=20352, PartyID=16777848
+++++++++++=Next CUCM sends ACK to 200 OK from SIP Phone+++++++++++
//SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.10.104 on port 62220 index 1890
[885635,NET]
ACK sip:[email protected]:62220;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK23366067b8c0
From: ;tag=192115~d8e94532-127d-4dca-bba0-64b1675da032-24378484
To: ;tag=5B0E9816C2CA6D70F3166FB972EDE4C2
Date: Tue, 19 Feb 2013 21:44:45 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 103 ACK
Allow-Events: presence
Content-Type: application/sdp
Content-Length: 237
v=0
o=CiscoSystemsCCM-SIP 192115 3 IN IP4 10.10.10.195
s=SIP Call
c=IN IP4 10.10.10.137
b=TIAS:64000
b=AS:64
t=0 0
m=audio 20352 RTP/AVP 9 101
a=rtpmap:9 G722/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
Now at this point all is well...and the call is connected....
Now here is where the call is failing on the SIP-SCCP-SIP call without MTP
From Point 2 above, CUCM sends a DO to insert MOH, and then gets response, then sends an ACK to 200 Ok back to SIP Phone..
//SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.10.104 on port 53361 index 1810
[881160,NET]
ACK sip:[email protected]:53361;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK22035ecc1fcb
From: ;tag=190666~d8e94532-127d-4dca-bba0-64b1675da032-24378214
To: "492" ;tag=97C34E1FB9A11F83DD8D8F5BA4C87C57
Date: Tue, 19 Feb 2013 17:38:50 GMT
Call-ID: 1CCA5149B966DC89AE0F752B8EF86480BC7102DB
Max-Forwards: 70
CSeq: 102 ACK
o=CiscoSystemsCCM-SIP 190666 3 IN IP4 10.10.10.195
s=SIP Call
c=IN IP4 10.10.10.195---------------------------------------IP address of MOH server
t=0 0
m=audio 4000 RTP/AVP 0--------------------------------MOH port 4000
a=rtpmap:0 PCMU/8000
a=ptime:20
a=sendonly---------------------------------------------------------sendonly
+++++NOW Point 6 above (SIP) CUCM sends a=inactive to break media path to MOH server to connect caller and xfered party++++++
//SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.10.104 on port 53361 index 1810
[881161,NET]
INVITE sip:[email protected]:53361;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK22045bb7f918
From: ;tag=190666~d8e94532-127d-4dca-bba0-64b1675da032-24378214
To: "492" ;tag=97C34E1FB9A11F83DD8D8F5BA4C87C57
Date: Tue, 19 Feb 2013 17:39:04 GMT
Call-ID: 1CCA5149B966DC89AE0F752B8EF86480BC7102DB
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM8.6
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 103 INVITE
Max-Forwards: 70
Expires: 180
Allow-Events: presence
Remote-Party-ID: ;party=calling;screen=yes;privacy=off
Contact:
Content-Type: application/sdp
Content-Length: 164
v=0
o=CiscoSystemsCCM-SIP 190666 4 IN IP4 10.10.10.195
s=SIP Call
c=IN IP4 0.0.0.0--------------------------------------------------------------------Media IP is 0.0.0.0
t=0 0
m=audio 4000 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20
a=inactive---------------------------------------------------------------------media inactive
At this point, we should get a response back from the sip phone...
and here is what we got..
++Trying which is expected++++
//SIP/SIPTcp/wait_SdlReadRsp: Incoming SIP TCP message from 10.10.10.104 on port 53361 index 1810 with 331 bytes:
[881162,NET]
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK22045bb7f918
From: ;tag=190666~d8e94532-127d-4dca-bba0-64b1675da032-24378214
Call-ID: 1CCA5149B966DC89AE0F752B8EF86480BC7102DB
CSeq: 103 INVITE
To: "492" ;tag=97C34E1FB9A11F83DD8D8F5BA4C87C57
Content-Length: 0
++++++++Then we get a BYE+++++++++++++++
/SIP/SIPTcp/wait_SdlReadRsp: Incoming SIP TCP message from 10.10.10.104 on port 53361 index 1810 with 576 bytes:
[881163,NET]
BYE sip:[email protected]:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.10.10.104:53361;branch=z9hG4bKa2vdQvR7J9OiMyjU;rport
Contact:
Max-Forwards: 70
From: "492" ;tag=97C34E1FB9A11F83DD8D8F5BA4C87C57
Allow: OPTIONS, INVITE, ACK, REFER, CANCEL, BYE, NOTIFY
Supported: replaces, path
User-Agent: Acrobits Softphone Business/2.4.8
To: ;tag=190666~d8e94532-127d-4dca-bba0-64b1675da032-24378214
Call-ID: 1CCA5149B966DC89AE0F752B8EF86480BC7102DB
CSeq: 3 BYE
Content-Length: 0
So this is the root cause of the problem. Your SIP phone does not know how to respond to multiple media break between it and the MOH server.
The difference between this and the succesful SCCP-SIP--SCCP, is that the held party was a sccp phone, hence the sip phone only has to process one a=inactive SDP message, where as in the SIP-SCCP-SIP, the help party was sip, so the sip phone has to process two a=inactive SDP messages
Now what is the difference when MTP is involved! A Big difference. MTP stays in the media path. So there is never a break in media and no inactive SDP attribute is sent. The flow looks like below:
for the initial call...The SIP phone sends its media to MTP and likewise the SCCP phone
SIP------Media------MTP------------Media-------SCCP Phone
When the new destination is dialled and transfer is commited,
SIP-------------media----MTP--------media---------MTP
The final invoite sent to connect 492 and 491 has MTP as the IP address to connect Media to.
++++++++Ivite to 492 ++++++++++++++
INVITE sip:[email protected]:61303;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK231a3b24b862
From: ;tag=192048~d8e94532-127d-4dca-bba0-64b1675da032-24378472
To: ;tag=78FF5BF6C019A55EA020B69BB6A767E2
Date: Tue, 19 Feb 2013 21:24:59 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM8.6
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 102 INVITE
Max-Forwards: 70
Expires: 180
Allow-Events: presence
Remote-Party-ID: ;party=calling;screen=yes;privacy=off
Contact:
Content-Type: application/sdp
Content-Length: 214
v=0
o=CiscoSystemsCCM-SIP 192048 1 IN IP4 10.10.10.195
s=SIP Call
c=IN IP4 10.10.10.195---------------------------------------------------------------the MTP ip address
t=0 0
m=audio 25038 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
+++++++Invite to 491 +++++++++++++++++
//SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.10.94 on port 50376 index 1887
[885429,NET]
INVITE sip:[email protected]:50376;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK231b78d1b56
From: ;tag=192046~d8e94532-127d-4dca-bba0-64b1675da032-24378467
To: "491" ;tag=F13CE94DE942C47680356A647DC7F916
Date: Tue, 19 Feb 2013 21:24:59 GMT
Call-ID: AE7045FFB2D8D9C28D54651473A14A5D41B5B93C
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM8.6
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Max-Forwards: 70
Expires: 180
Allow-Events: presence
Remote-Party-ID: "Leslie2" ;party=calling;screen=no;privacy=off
Contact:
Content-Type: application/sdp
Content-Length: 237
v=0
o=CiscoSystemsCCM-SIP 192046 1 IN IP4 10.10.10.195
s=SIP Call
c=IN IP4 10.10.10.195----------------------------------------MTP
b=TIAS:64000
b=AS:64
t=0 0
m=audio 25030 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
Wao! That was a long one isnt it...It was fun too.
So now you can look at your sip phones and see if they can accept two inactive sdp messages within the same call. That way you can remove MTP. otherwise you will have MTP involved in every single call involving a sip phone, even if they do not involve transfers
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"opportunity is a haughty goddess who waste no time with those who are unprepared"

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TS3276 When I add a name to the "To" box or CC/BCC box and then change my mind about sending an email to them, Mail is ignoring the instruction to "remove" so that once the email has been sent, their name is still included and the email goes to them anywa

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Please remember, if you see a post that helped you please click "Vote As Helpful" and if it answered your question, please click "Mark As Answer"
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