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Summary:ASTERISK-28270: Creating call using ARI, does not increasing/decreasing the call count(core show calls)
Reporter:sungtae kim (pchero)Labels:pjsip
Date Opened:2019-02-02 09:27:35.000-0600Date Closed:2019-02-04 09:04:12.000-0600
Priority:MinorRegression?
Status:Closed/CompleteComponents:PBX/General
Versions:16.1.1 Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) wrong_call_count.log
Description:Hi,

Creating call using ARI such as below, does not increasing/decreasing the call count.

* POST ari/channels
* POST ari/channels/{channel id}
Comments:By: Asterisk Team (asteriskteam) 2019-02-02 09:27:36.055-0600

Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution.

A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report.

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By: sungtae kim (pchero) 2019-02-02 09:34:46.172-0600

I was trying to fix this, but after reach to the `ast_pbx_start()`, then I was lost where should I fix it. Because it looks so much complicated.

If It's not that to much hard to fix it, I would like try to fix. May I get some advise?

Thank you.
By: George Joseph (gjoseph) 2019-02-04 07:33:58.378-0600

Seems to be working OK for me.   Do you have a test stub and a dialplan snippet I can try?

By: sungtae kim (pchero) 2019-02-04 08:25:41.390-0600

Is this might be helpful?

{noformat}
*CLI> [2019-02-04 15:22:01] NOTICE[26233]: res_ari.c:768 process_cors_request: Origin header 'chrome-extension://pfdhoblngboilpfeibdedpjgfnlcodoo' does not match an allowed origin.
Creating Stasis app 'test'
 == WebSocket connection from '192.168.100.1:64444' for protocol '' accepted using version '13'
   -- Called sipp_test/sip:test@127.0.0.1:5061
   -- PJSIP/sipp_test-00000001 is ringing
      > 0x7f75f800a440 -- Strict RTP learning after remote address set to: 127.0.1.1:6000
   -- PJSIP/sipp_test-00000001 is ringing
   -- PJSIP/sipp_test-00000001 answered
      > Launching Stasis(test) on PJSIP/sipp_test-00000001

*CLI> core show channels
Channel              Location             State   Application(Data)
PJSIP/sipp_test-0000 s@test:1             Up      Stasis(test)
1 active channel
0 active calls
0 calls processed
{noformat}

1 channel, but 0 call.
By: George Joseph (gjoseph) 2019-02-04 08:41:05.238-0600

So the reason the call count isn't updating for you is that the call counter is driven by the dialplan.  When you originate a call to a technology destination other than "Local", the dialplan isn't invoked.  In your scenario "PJSIP/test-voip/sip:testcall@localhost:5061" bypasses the dialplan.

To be honest, I'm not sure if  this is a bug nor not and I'm not sure how I'd fix it if it were.


By: sungtae kim (pchero) 2019-02-04 08:57:19.001-0600

I had a talk in the IRC, about the 'calls'.

the 'calls' things are not provide correct info anymore(Especially ARI). It's a historical things.

{noformat}
file:
15:43 the definition of "call" within Asterisk was the number of channels executing PBX logic
15:43 well, is
15:43 that hasn't been changed/altered/tweaked since probably the creation of Asterisk
15:44 even though there are new ways to create channels, which could be considered "calls"
15:44 it gets complicated because of how you can move channels around and connect them in ARI too
{noformat}