ASTERISK-18992: Asterisk From and To fields setup for SIP out of dialog MESSAGE method

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Summary: ASTERISK-18992: Asterisk From and To fields setup for SIP out of dialog MESSAGE method

Reporter: Yuri (ys) Labels:

Date Opened: 2011-12-08 06:41:54.000-0600 Date Closed: 2012-01-25 11:19:01.000-0600

Priority: Major Regression?

Status: Closed/Complete Components: Channels/chan_sip/General Channels/chan_sip/Messaging

Versions: SVN 10.0.0-rc2 Frequency of
Occurrence Constant

Related
Issues:
is related to ASTERISK-18917 Asterisk 10 Incorrectly Formats From Header

is related to ASTERISK-19058 Messagesend and SIPFROMUSER

Environment: Linux and FreeBSD machne asterisk-10.0.0-rc2 and asterisk trunk ( revision 347528) Attachments: ( 0) 347939.diff

Description: This issue are related with ASTERISK-18903, ASTERISK-18916, ASTERISK-18917

I,m dance some days with sip IM message service in asterisk and found the following bugs:

If I set MESSAGE_DATA(to) and/or MESSAGE_DATA(from) variable in dialplan, for example:

exten => 100,1,set(MESSAGE_DATA(to)=sip:100)
exten => 100,n,set(MESSAGE_DATA(from)=${MESSAGE(from)})
exten => 100,n,MessageSend(MESSAGE_DATA(to), ${MESSAGE_DATA(from)})

Asterisk send MESSAGE to destination w/two From: and To: headers!

MESSAGE sip:100@192.168.1.119:46944;rinstance=2c4db922ff368387;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.1:15072;branch=z9hG4bK533df9e5;rport
Max-Forwards: 70
From: " sip:100@192.168.1.1:15072;transport=UDP" <sip:asterisk@192.168.1.1:15072>;tag=as42cac1e2
To: <sip:100@192.168.1.119:46944;rinstance=2c4db922ff368387;transport=UDP>
Contact: <sip:asterisk@192.168.1.1:15072>
Call-ID: 1c6ae69362a502b264f246b8732a180f@192.168.1.1:0
CSeq: 102 MESSAGE
User-Agent: Asterisk PBX SVN-trunk-r347008
To: sip:100
From: sip:100@192.168.1.1:15072;transport=UDP
Content-Type: text/plain;charset=UTF-8
Content-Length: 4

test
---

SIP/2.0 400 Bad Request(Multiple values in single-value header To,Multiple values in single-value header From)
Via: SIP/2.0/UDP 192.168.1.1:15072;branch=z9hG4bK533df9e5;rport=15072
To: <sip:100@192.168.1.119:46944;rinstance=2c4db922ff368387;transport=UDP>;tag=dc6c4220
From: " sip:100@192.168.1.1:15072;transport=UDP" <sip:asterisk@192.168.1.1:15072>;tag=as42cac1e2
Call-ID: 1c6ae69362a502b264f246b8732a180f@192.168.1.1:0
CSeq: 102 MESSAGE
Content-Length: 0

The first (valid ?) "To" and "From" SIP header created by sip_msg_send()function and placed into "struct sip_pvt".
Inside transmit_message_with_msg(pvt, msg) function, call to initreqprep(); added valid "From" and "To" SIP header into sip request

The second (invalid?)"From"/"To" header added later in transmit_message_with_msg() inside interrator by add_header()

As a solution, we can disable adding standard SIP headers that we get from asterisk message engine (main/message.c).

Comments: By: Yuri (ys) 2011-12-10 04:54:37.584-0600

I'm attach file, where I make the following:

1. For out of dialog MESSAGE I remove check for content-type, that allow the indication transmisssion between the peer, for example:

MESSAGE sip:100@192.168.1.1:15072;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.119:20434;branch=z9hG4bK-d8754z-c6c10996151f0619-1---d8754z-
Max-Forwards: 70
To: <sip:200@192.168.1.1:15072;transport=UDP>
From: "100"<sip:100@192.168.1.1:15072;transport=UDP>;tag=f771a43c
Call-ID: YmI5N2Y1NDkzMDc5NTI3MDc2MzQ0MGQzYjc2MmIxMTM.
CSeq: 11 MESSAGE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/im-iscomposing+xml
User-Agent: Zoiper Communicator 2.05.11136 rev.11135
Allow-Events: presence, kpml
Content-Length: 323

<?xml version="1.0" encoding="UTF-8"?>
<isComposing xmlns="urn:ietf:params:xml:ns:im-iscomposing" xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance" xsi:schemaLocation="urn:ietf:params:xml:ns:im-composing iscomposing.xsd"> <state>active</state> <refresh>120</refresh>
</isComposing>

This work as expected.

2. Content-type may changed by dialplan funtion MESSAGE_DATA(), defaults is "text/plain;charset=UTF-8"
3. I add check for prevent re-add the SIP header "From" and "To" into SIP message.
4. Decrement "Max-Forwards" headers for forwarded messages.
5. "From" field passed to ast_message w/o ast_str_buffer() call.
6. Message send engine pass msg->from to sip level message transmitter, if this argument not set in dialplan function.

Affected files:
main/message.c
channels/chan_sip.c
By: Yuri (ys) 2011-12-14 12:54:50.252-0600

Also found:

create_addr(), that called from sip_msg_send(), overwrite fromuser in SIP dialog, what set in this function before, if fromuser= set in sip.conf.
This leads to the field "From" in outgoing OOD MESSAGE becoming invalid...

By: Richard Mudgett (rmudgett) 2012-01-20 16:03:00.603-0600

Posted a patch on reviewboard https://reviewboard.asterisk.org/r/1683/

* I removed the new feature to allow other incoming Content-Types. That can only be put on trunk. Also that feature needs to be configurable and should specify what other Content-Type values are allowed rather than allowing anything.

* Reading header items out of the incoming message (Content-Type and From) is not a good idea. The incoming message may have come from a different technology. Remember, the messaging system also supports XMPP (Jabber).