[asterisk-bugs] [JIRA] (ASTERISK-28513) Should To: be rewritten when forwarding to a phone

[y (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-28513?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=254224#comment-254224 ] 

y commented on ASTERISK-28513:
------------------------------

The status of my License sign page says: You already have a license that is approved or pending review.

I am considering the following situation: Can we forward the message to multiple registered devices like Dial(${PJSIP_DIAL_CONTACTS(${EXTEN})}) does? and we set "To" in update_to according to MESSAGE(to) rather than the first parameter of MessageSend.
The question is MessageSend only accepting one destination, and the implementation inside asterisk ignores MESSAGE(to), the enhancement workloads might be too much. 

> Should To: be rewritten when forwarding to a phone
> --------------------------------------------------
>
>                 Key: ASTERISK-28513
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-28513
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Resources/res_pjsip_messaging
>    Affects Versions: 13.28.0
>            Reporter: Brian J. Murrell
>            Assignee: Unassigned
>              Labels: pjsip
>
> I have run into a problem with {{MESSAGE}} s and the [Linphone Android client|https://github.com/BelledonneCommunications/linphone-android].
> The [issue|https://github.com/BelledonneCommunications/linphone-android/issues/605] as described in their tracker is that when {{MESSAGE}} s come to the linphone client from the same sender, they can be "filed" into many different threads, rather than all in one chat/thread.  This is because linphone separates chats based on both the From: and To: headers.
> As I am sure you know, the {{To:}} header of a client can vary wildly based on the IP address it's connecting from.  This means that every time the IP address of the SIP client changes, a new chat for the same sender is created.
> But the problem is that Asterisk is setting the {{To:}} header of the {{MESSAGE}} to the {{user at ip_address}} of the remote SIP client and so this means that every time the IP address of the remote SIP client changes, a new To: header is created, and so is a new chat in the SIP client.
> Linphone defends this behaviour by insisting that the {{To:}} header value is a logical value of the recipient for a given domain and should always remain it's logical value no matter whether it's being forwarded on to a SIP client or not.
> So for example, if my Asterisk server is at pbx.example.com and somebody (my VOIP provider for example) send a {{MESSAGE}} to 555-555-1212 at ip-address-of-my-asterisk, when my Asterisk server receives that message and then wants to forward it on to a SIP client, the To: should be {{To:  _recipient_]@pbx.example.com}}, not {{To: _recipient_ at ip-address-of-SIP-client}}.
> They quote [RFC 3261 section 8.1.1.2|https://tools.ietf.org/html/rfc3261#section-8.1.1.2] further in defending this behaviour.  My reading of it doesn't leave me with much to argue against their defence.
> I don't see any way to make Asterisk (with PJSIP) to follow this behaviour.



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