Content

Page tree

Versions Compared

Key

  • This line was added.
  • This line was removed.
  • Formatting was changed.
Sv translation
languageen

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.

Code Block
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>


Info

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:

Code Block
titleSIP Trace / To open it click on the link -->
linenumberstrue
collapsetrue
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


Info
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.


Include Page
Howto Footer - uni-en
Howto Footer - uni-en

Content by Label
showLabelsfalse
max20
spacesPW
showSpacefalse
sorttitle
typepage
cqllabel in ("kb-how-to-article","kb-troubleshooting-article") and label = "deskphone" and type = "page"
labelsdect dect-multicell


Sv translation
languagede

Dies kann über die folgende SIP INFO Message erfolgen und muss während des bestehenden Gesprächs gesendet werden. Diese Nachricht sollte von der Telefonanlage gesendet werden.

Code Block
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>


Info

Hinweis: Falls Sie die INFO-Message selbst generieren, stellen Sie bitte sicher, dass:

  • Sie genau die CallID sowie den From- und To-Header aus den SIP-Messages des Gesprächs, das Sie ändern möchten, kopiert haben (Sie sollten die Header aus den '200 OK'- oder den 'ACK'-Messages kopieren und nicht aus der 'INVITE'-Message, da der Tag) im 'To'-Header der 'INVITE'-Nachricht fehlt)
  • Die CSeq-Nummer höher ist als die letzte Cseq, die innerhalb des Gesprächs, das Sie ändern möchten, gesendet wurde
  • Die Content-Length die richtige Summe der Zeichen im Inhalt enthält, wobei zu beachten ist, dass für jede Zeile 2 zusätzliche Zeichen gezählt werden müssen, da am Ende jeder Zeile ein \r\n steht, um den Zeilenumbruch zu repräsentieren, den wir nicht sehen (hier hat der eigentliche Text z. B. 41 Zeichen, und wir müssen für jede Zeile 2 weitere hinzufügen. Die Summe ist 41+4=45, also lautet der Header 'Content-Length: 45')


Hier ist ein vollständiger SIP-Trace eines Anrufs, gefolgt von einer SIP-INFO-Meldung im Dialog:

Code Block
titleSIP Trace / Zum Öffnen klicken Sie auf den Link -->
linenumberstrue
collapsetrue
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


Info
Bekannte Probleme: Der "From:"-Eintrag im SIP-INFO Message Body sollte immer den entfernten Gesprächspartner aktualisieren, unabhängig von der Richtung der Nachricht im Vergleich zum ursprünglichen INVITE. In einigen Versionen aktualisiert der "From:"-Eintrag stattdessen den lokalen Teilnehmer, was nicht korrekt ist. Bitte verwenden Sie die neueste verfügbare Version.


Include Page
Howto Footer - de
Howto Footer - de

Content by Label
showLabelsfalse
max20
spacesPW
showSpacefalse
sorttitle
typepage
cqllabel in ("kb-how-to-article","kb-troubleshooting-article") and label = "deskphone" and type = "page"
labelsdect dect-multicell