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This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/3726/
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(Updated July 27, 2014, 9:20 p.m.)
Review request for Asterisk Developers.
Changes
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Updated description and testing
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 (updated)
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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
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/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 (updated)
-------
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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