[asterisk-users] RHS of the To: address in MESSAGE transactions

[Brian J. Murrell]

On Thu, 2019-06-06 at 09:33 -0400, Brian J. Murrell wrote:
> I'm trying to use linphone-android with asterisk but there is an
> aspect
> of the way asterisk and linphone-android interact with MESSAGE
> transactions that is causing problems.
> 
> The linphone-android folks consider both the To: and From: address in
> MESSAGE transactions when deciding which "chat" to put a received
> MESSAGE into.  Every combination of To: and From: address are a
> separate "chat" in their messaging paradigm.
> 
> So when asterisk sends a MESSAGE transaction such as:
> 
> MESSAGE
> sip:my_sip_account@[2001:123:ab:123:51e2:cc83:ae66:8c70]:38915;transp
> ort=udp;app-id=755770037818;pn-type=firebase;pn-timeout=0;pn-
> tok==[redacted];pn-silent=1 SIP/2.0
> Via: SIP/2.0/UDP
> [2001:123:ab:123::2]:5060;rport;branch=z9hG4bKPj138c026c-4437-4b59-
> 982f-f991521d3cdc
> From: "5565551212" <sip:[redacted]@pbx.example.com>;tag=5b5fe395-
> ff22-44fa-aa6b-7f770f8e0026
> To: <sip:my_sip_account@[2001:123:ab:123:51e2:cc83:ae66:8c70];app-
> id=755770037818;pn-type=firebase;pn-timeout=0;pn-tok==[redacted];pn-
> silent=1>
> Contact: <sip:my_sip_account@[2001:123:ab:123::2]:5060>
> Call-ID: 5e4fc686-72ce-4c20-bd2f-7f82e232a9db
> CSeq: 29808 MESSAGE
> Max-Forwards: 70
> User-Agent: Asterisk PBX 13.26.0
> Content-Type: text/plain
> Content-Length:     4
> 
> hey!
> 
> it files it into the chat for the combination of:
> 
> From: "5565551212" <sip:[redacted]@pbx.example.com>;tag=5b5fe395-
> ff22-44fa-aa6b-7f770f8e0026
> To: <sip:my_sip_account@[2001:123:ab:123:51e2:cc83:ae66:8c70];app-
> id=755770037818;pn-type=firebase;pn-timeout=0;pn-tok==[redacted];pn-
> silent=1>
> 
> and because the To: includes the IP address of the client, every time
> the client moves networks (or even regenerates a new "Privacy
> Extensions" IPv6 address) a new chat is created for the same sender.
> 
> Their position is that the RHS of the To: should be the name of the
> Asterisk machine such as:
> 
> To: <sip:my_sip_account at pbx.example.com;app-id=755770037818;pn-
> type=firebase;pn-timeout=0;pn-tok==[redacted];pn-silent=1>
> 
> While that seems odd to me, from a common-sense perspective, I don't
> have a deep enough background in the SIP protocol to decide if it's
> wrong or not, and if it's not, how to make Asterisk do it, as in
> something similar to the pjsip from_domain endpoint parameter that
> can
> be used to set the domain of the From: header for MESSAGEs to that
> endpoint.
> 
> Any opinions, ideas or otherwise?

Nobody has any options either way on this one?

Cheers,
b.

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