> On July 30, 2014, 10:31 a.m., opticron wrote:
> > /branches/12/rest-api/api-docs/endpoints.json, line 23
> > <https://reviewboard.asterisk.org/r/3726/diff/2/?file=65043#file65043line23>
> >
> >     Doesn't this conflict with /endpoints/{tech} and 
> > /endpoints/{tech}/{resource} below? There are currently no other paths that 
> > override a path parameter like this.
> 
> Matt Jordan wrote:
>     It is a direct operation on the endpoints resource itself. It doesn't 
> make sense for it to be on a {tech} or on a {resource}: you specify both the 
> tech and the resource in the "to" field.
>     
>     The only thing this could potentially conflict with is if someone created 
> a channel technology named "sendMessage", and only if there was a PUT 
> operation for /endpoints/{tech}. Since there isn't there is no conflict.

I should note as well: having a channel technology named "sendMessage" would be 
silly and would be a highly non-standard channel technology name. Further, we 
don't support a PUT operation on a channel technology via the endpoints 
resource: what are you PUTting? Even if we support dynamic endpoint creation 
(which would be cool), that would most likely be a POST to 
/endpoints/{tech}/{resource}, which still avoids any collisions here.


- Matt


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On July 27, 2014, 9:20 p.m., Matt Jordan wrote:
> 
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/3726/
> -----------------------------------------------------------
> 
> (Updated July 27, 2014, 9:20 p.m.)
> 
> 
> Review request for Asterisk Developers.
> 
> 
> Bugs: ASTERISK-23692 and ASTERISK-23969
>     https://issues.asterisk.org/jira/browse/ASTERISK-23692
>     https://issues.asterisk.org/jira/browse/ASTERISK-23969
> 
> 
> Repository: Asterisk
> 
> 
> Description
> -------
> 
> This patch adds the ability to send and receive text messages from various 
> technology stacks in Asterisk through ARI. This includes chan_sip (sip), 
> res_pjsip_messaging (pjsip), and res_xmpp (xmpp).
> 
> The following would send the message "Hello there" to PJSIP endpoint alice 
> with a display URI of sip:aster...@mycooldomain.org:
> 
> ari/endpoints/sendMessage?to=pjsip:alice&from=sip:aster...@mycooldomain.org&body=Hello+There
> 
> This is equivalent to the following as well:
> 
> ari/endpoints/PJSIP/alice/sendMessage?from=sip:aster...@mycooldomain.org&body=Hello+There
> 
> Both forms are available for message technologies that allow for arbitrary 
> destinations, such as chan_sip.
> 
> Inbound messages can now be received over ARI. An ARI application that 
> subscribes to endpoints will receive messages from those endpoints:
> 
> {
>   "type": "TextMessageReceived",
>   "timestamp": "2014-07-12T22:53:13.494-0500",
>   "endpoint": {
>     "technology": "PJSIP",
>     "resource": "alice",
>     "state": "online",
>     "channel_ids": []
>   },
>   "message": {
>     "from": "\"alice\" <sip:alice@127.0.0.1>",
>     "to": "pjsip:asterisk@127.0.0.1",
>     "body": "Watson, come here.",
>     "variables": []
>   },
>   "application": "testsuite"
> }
> 
> A few interesting things you could do with this:
> (1) Build your own XMPP to SIP gateway (without ever touching dialplan)
> (2) Make a conferencing application with built-in text messaging (speech to 
> text would be fun with this... probably should write that too)
> (3) WebRTC! SIP stacks in the browser can send MESSAGE requests. Why limit 
> yourself to just making calls when you can send arbitrary messages to a 
> communications application? (Note: if you can't mention WebRTC in a release, 
> you're not trying very hard)
> 
> The above was made possible due to some rather major changes in the message 
> core. This includes (but is not limited to):
> - Users of the message API can now register message handlers. A handler has 
> two callbacks: one to determine if the handler has a destination for the 
> message, and another to handle it.
> - All dialplan functionality of handling a message was moved into a message 
> handler provided by the message API.
> - Messages can now have the technology/endpoint associated with them. Various 
> other properties are also now more easily accessible.
> - A number of ao2 containers that weren't really needed were replaced with 
> vectors. Iteration over ao2_containers is expensive and pointless when the 
> lifetime of things is well defined and the number of things is very small.
> 
> res_stasis now has a new file that makes up its structure, messaging. The 
> messaging functionality implements a message handler, and passes received 
> messages that match an interested endpoint over to the app for processing.
> 
> Note that inadvertently while testing this, I reproduced ASTERISK-23969. 
> res_pjsip_messaging was incorrectly parsing out the 'to' field, such that 
> arbitrary SIP URIs mangled the endpoint lookup. This patch includes the fix 
> for that as well.
> 
> 
> Diffs
> -----
> 
>   /branches/12/tests/test_message.c PRE-CREATION 
>   /branches/12/rest-api/api-docs/events.json 419205 
>   /branches/12/rest-api/api-docs/endpoints.json 419205 
>   /branches/12/res/stasis/app.c 419205 
>   /branches/12/res/res_xmpp.c 419205 
>   /branches/12/res/res_stasis.c 419205 
>   /branches/12/res/res_pjsip_messaging.c 419205 
>   /branches/12/res/res_ari_endpoints.c 419205 
>   /branches/12/res/ari/resource_endpoints.c 419205 
>   /branches/12/res/ari/resource_endpoints.h 419205 
>   /branches/12/res/ari/resource_channels.c 419205 
>   /branches/12/res/ari/ari_model_validators.c 419205 
>   /branches/12/res/ari/ari_model_validators.h 419205 
>   /branches/12/main/message.c 419205 
>   /branches/12/main/json.c 419205 
>   /branches/12/include/asterisk/vector.h 419205 
>   /branches/12/include/asterisk/message.h 419205 
>   /branches/12/include/asterisk/manager.h 419205 
>   /branches/12/include/asterisk/json.h 419205 
>   /branches/12/channels/chan_sip.c 419205 
> 
> Diff: https://reviewboard.asterisk.org/r/3726/diff/
> 
> 
> Testing
> -------
> 
> Unit tests were added for the message core to make sure dialplan still worked.
> 
> Basic nominal tests have been added for the Asterisk Test Suite, and are up 
> for review at https://reviewboard.asterisk.org/r/3864/
> 
> 
> Thanks,
> 
> Matt Jordan
> 
>

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