[asterisk-dev] [Code Review] 3726: ari: Add message technology agnostic out of call text messaging

Matt Jordan reviewboard at asterisk.org
Tue Aug 5 15:10:59 CDT 2014

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(Updated Aug. 5, 2014, 3:10 p.m.)


This change has been marked as submitted.

Review request for Asterisk Developers.


Committed in revision 420089

Bugs: ASTERISK-23692 and ASTERISK-23969

Repository: Asterisk


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:asterisk at mycooldomain.org:

ari/endpoints/sendMessage?to=pjsip:alice&from=sip:asterisk at mycooldomain.org&body=Hello+There

This is equivalent to the following as well:

ari/endpoints/PJSIP/alice/sendMessage?from=sip:asterisk at 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 at>",
    "to": "pjsip:asterisk at",
    "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.


  /branches/12/tests/test_message.c PRE-CREATION 
  /branches/12/rest-api/api-docs/events.json 419870 
  /branches/12/rest-api/api-docs/endpoints.json 419870 
  /branches/12/res/stasis/app.c 419870 
  /branches/12/res/res_xmpp.c 419870 
  /branches/12/res/res_stasis.c 419870 
  /branches/12/res/res_pjsip_messaging.c 419870 
  /branches/12/res/res_ari_endpoints.c 419870 
  /branches/12/res/ari/resource_endpoints.c 419870 
  /branches/12/res/ari/resource_endpoints.h 419870 
  /branches/12/res/ari/resource_channels.c 419870 
  /branches/12/res/ari/ari_model_validators.c 419870 
  /branches/12/res/ari/ari_model_validators.h 419870 
  /branches/12/main/message.c 419870 
  /branches/12/main/json.c 419870 
  /branches/12/include/asterisk/vector.h 419870 
  /branches/12/include/asterisk/message.h 419870 
  /branches/12/include/asterisk/manager.h 419870 
  /branches/12/include/asterisk/json.h 419870 
  /branches/12/channels/chan_sip.c 419870 

Diff: https://reviewboard.asterisk.org/r/3726/diff/


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/


Matt Jordan

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