FAQ/How to change the displayed source and destination data of a call

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Answer

This can be done via the following SIP INFO message and must be sent during the existing dialog. This message should be sent by the PBX.

INFO sip:431@10.10.12.87 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.59:1036;branch=z9hG4bK29c9b031;rport
From: "430" <sip:430@192.168.10.59>;tag=as2551b6cf
To: <sip:431@10.10.12.87:5060>;tag=ctxr9rd1t6
Contact: <sip:info@PBXserver>
Call-ID: 41304da47a3fde79141e862424996fa9@192.168.10.59
CSeq: 200 INFO
Max-Forwards: 70
User-Agent: snom320/testing_branch_7_3_2009_02_23_13_32_41
Content-Type: message/sipfrag
Content-Length: 45

To:
From: "New-ID" <sip:123@192.168.10.11>

Note: In case you are generating the INFO message yourself, please make sure that:

  • you have copied exactly the CallID, From header and To header from the SIP-messages of the dialog you want to change (you should copy the headers from the '200 OK' or the 'ACK' messages, and not from the 'INVITE' message, because the tag is missing in the 'To' header of the 'INVITE' message)
  • CSeq number is higher then the last Cseq sent within the dialog you want to change
  • Content-Length is the correct sum of characters in the content, keeping in mind that we must count 2 extra characters for each line because at the end of every line there is a \r\n to represent the newline which we don't see (for example here the actual text has 41 characters, and we must add 2 more for each line. The sum is 41+4=45, so the header is 'Content-Length: 45')


Here is a complete SIP trace of a call followed by an in-dialog SIP INFO message:

INVITE sip:432@10.10.12.88:1024;line=kdpizpq6 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.59:5060;branch=z9hG4bK111c0678;rport
From: "430" <sip:430@192.168.10.59>;tag=as34192650
To: <sip:432@10.10.12.88:1024;line=kdpizpq6>
Contact: <sip:430@192.168.10.59>
Call-ID: 7b0f73691b286eea690513eb7bb16b51@192.168.10.59
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 02 Jun 2010 09:52:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 264

v=0
o=root 1877 1877 IN IP4 192.168.10.59
s=session
c=IN IP4 192.168.10.59
t=0 0
m=audio 17940 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

Sent to udp:192.168.10.59:5060 at 2/6/2010 00:52:15:025 (503 bytes):

SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.10.59:5060;branch=z9hG4bK111c0678;rport=5060
From: "430" <sip:430@192.168.10.59>;tag=as34192650
To: <sip:432@10.10.12.88:1024;line=kdpizpq6>;tag=dwsvd8el2x
Call-ID: 7b0f73691b286eea690513eb7bb16b51@192.168.10.59
CSeq: 102 INVITE
Contact: <sip:432@10.10.12.88:1024;line=kdpizpq6>;reg-id=1
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Content-Length: 0

Sent to udp:192.168.10.59:5060 at 2/6/2010 00:52:15:539 (503 bytes):

SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.10.59:5060;branch=z9hG4bK111c0678;rport=5060
From: "430" <sip:430@192.168.10.59>;tag=as34192650
To: <sip:432@10.10.12.88:1024;line=kdpizpq6>;tag=dwsvd8el2x
Call-ID: 7b0f73691b286eea690513eb7bb16b51@192.168.10.59
CSeq: 102 INVITE
Contact: <sip:432@10.10.12.88:1024;line=kdpizpq6>;reg-id=1
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Content-Length: 0

Sent to udp:192.168.10.59:5060 at 2/6/2010 00:52:16:085 (848 bytes):

SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.10.59:5060;branch=z9hG4bK111c0678;rport=5060
From: "430" <sip:430@192.168.10.59>;tag=as34192650
To: <sip:432@10.10.12.88:1024;line=kdpizpq6>;tag=dwsvd8el2x
Call-ID: 7b0f73691b286eea690513eb7bb16b51@192.168.10.59
CSeq: 102 INVITE
Contact: <sip:432@10.10.12.88:1024;line=kdpizpq6>;reg-id=1
User-Agent: snom370/8.2.29
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Content-Type: application/sdp
Content-Length: 240

v=0
o=root 859232531 859232532 IN IP4 10.10.12.88
s=call
c=IN IP4 10.10.12.88
t=0 0
m=audio 49544 RTP/AVP 8 0 101
a=rtpmap:8 pcma/8000
a=rtpmap:0 pcmu/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

Received from udp:192.168.10.59:5060 at 2/6/2010 00:52:16:356 (402 bytes):

ACK sip:432@10.10.12.88:1024;line=kdpizpq6 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.59:5060;branch=z9hG4bK67746086;rport
From: "430" <sip:430@192.168.10.59>;tag=as34192650
To: <sip:432@10.10.12.88:1024;line=kdpizpq6>;tag=dwsvd8el2x
Contact: <sip:430@192.168.10.59>
Call-ID: 7b0f73691b286eea690513eb7bb16b51@192.168.10.59
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

Received from udp:192.168.10.59:1036 at 2/6/2010 01:00:44:322 (493 bytes):

INFO sip:432@10.10.12.88 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.59:1036;branch=z9hG4bK29c9b031;rport
From: "430" <sip:430@192.168.10.59>;tag=as34192650
To: <sip:432@10.10.12.88:1024;line=kdpizpq6>;tag=dwsvd8el2x
Contact: <sip:info@PBXserver>
Call-ID: 7b0f73691b286eea690513eb7bb16b51@192.168.10.59
CSeq: 200 INFO
Max-Forwards: 70
User-Agent: snom320/testing_branch_7_3_2009_02_23_13_32_41
Content-Type: message/sipfrag
Content-Length: 45

To:
From: "New-ID" <sip:123@192.168.10.59>

Sent to udp:192.168.10.59:1036 at 2/6/2010 01:00:44:430 (354 bytes):

SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.10.59:1036;branch=z9hG4bK29c9b031;rport=1036
From: "430" <sip:430@192.168.10.59>;tag=as34192650
To: <sip:432@10.10.12.88:1024;line=kdpizpq6>;tag=dwsvd8el2x
Call-ID: 7b0f73691b286eea690513eb7bb16b51@192.168.10.59
CSeq: 200 INFO
Contact: <sip:432@10.10.12.88:1024;line=kdpizpq6>;reg-id=1
Content-Length: 0

Known issues: The "From:" entry in the SIP INFO message body should always update the remote call party, regardless of the direction of the message compared to the initial INVITE. In some versions the "From:" entry updates the local party instead, which is incorrect. Please use the latest version available.

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