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:[email protected];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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