[asterisk-bugs] [JIRA] (ASTERISK-28802) ARI /dial function disconnect Inbound call automatically if Outbound is "busy" or "discard" in case delay "Answer" message.

Fri Apr 3 06:59:25 CDT 2020

     [ https://issues.asterisk.org/jira/browse/ASTERISK-28802?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Lee Yun Seok updated ASTERISK-28802:
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    Description: 
We use ARI /dial function to connect outbound in Bridge.
We don't send "Answer" message until outbound answer.

ARI /dial function disconnect Inbound call automatically if Outbound is "busy" or "discard" 
In case we don't send "Answer" message.

But ARI /dial function don't disconnect Inbound call if outbound is "busy" or "discard". 
In casessage first. we send "Answer" me

We must use delay send "Answer" message until outbound answer call because of "Billing Issue"(We can't bill Inbound user in case outbound don't answer call) 

ARI /originate function don't disconnect Inbound call if outbound is "busy" or "discard". 
Even if In case we don't send "Answer" message.

We think "/ originate" and "/ dial" are same operate in case of "No Answer"

We can't understand why the two functions operate separately.

Please update "/ dial" function to do not disconnect Inbound call if outbound is "busy" or "discard". 
Even if In case we don't send "Answer" message.

Below is the test result.
1. When dialstatus==“ANSWER” of 300(callee),
Even if the 300(callee) hangs up, the 100(caller) remains connected.

Dial from 100(caller) to 300(callee).
Answer 300(callee).
Hangup 300(callee).
Now 100(caller) stay connected until 100(caller) hangs up.
2. But dialstatus==“BUSY” of 300(callee),
After the busy tone is heard, the 100(caller) hangs up.

Dial from 100(caller) to 300(callee).
Busy 300(callee).
Listen busy tone.
And 100(caller) hangs up by asterisk. >>>> I want to stay connected(don’t hang up on 100(caller)).

in_channel.on_event(‘ChannelHangupRequest’, self.on_hangup)
out_channel.on_event(‘StasisEnd’, self.oo_stop)
out_channel.on_event(‘Dial’, self.oo_dial)
this Dial event print dialstatus only, nothing else.
Run this command
1)out_channel.dial(caller=in_channel.id)

Then the following events occur in order when callee busy.
And caller heard busy tone and off-hook tone.

<SIP/100-00000019> 5.Add the channels to the bridge.
stasis_start_cb
<SIP/100-00000019> 7.Now the dialing begin.

Dial event: self.oo_dial()
dialstatus = 

Dial event: self.oo_dial()
dialstatus = [RINGING]

Dial event: self.oo_dial()
dialstatus = [BUSY]

stasis_end_cb
<SIP/100-00000019> on_hangup

stasis_end_cb
<SIP/300-00000019> oo_stop

localhostCLI>
== Using SIP RTP CoS mark 5
localhostCLI>
– Channel SIP/100-0000001d joined ‘holding_bridge’ stasis-bridge <8caff83f-66aa-42bf-92e1-7f4721bf2589>
<< [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/100-0000001d]
– Channel SIP/100-0000001d left ‘holding_bridge’ stasis-bridge <8caff83f-66aa-42bf-92e1-7f4721bf2589>
<< [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/100-0000001d]
– Channel SIP/100-0000001d joined ‘simple_bridge’ stasis-bridge <1f270170-1440-41f2-b0d0-635c11114cc9>
<< [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/100-0000001d]
<< [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/100-0000001d]
– Channel SIP/300-0000001e joined ‘simple_bridge’ stasis-bridge <1f270170-1440-41f2-b0d0-635c11114cc9>
<< [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/300-0000001e]
– Got SIP response 486 “Busy Here” back from 192.168.0.137:14550
<< [ TYPE: Control (4) SUBCLASS: Unknown control ‘33’ (33) ] [SIP/300-0000001e]
<< [ TYPE: Control (4) SUBCLASS: Busy (5) ] [SIP/300-0000001e]
<< [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/100-0000001d]
<< [ HANGUP (NULL) ] [SIP/100-0000001d]
– Channel SIP/100-0000001d left ‘simple_bridge’ stasis-bridge <1f270170-1440-41f2-b0d0-635c11114cc9>
– Channel SIP/300-0000001e left ‘simple_bridge’ stasis-bridge <1f270170-1440-41f2-b0d0-635c11114cc9>
– Channel SIP/300-0000001e joined ‘holding_bridge’ stasis-bridge <11aab1b8-5328-40d8-b707-92d0111bc1a3>
<< [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/300-0000001e]
<< [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/300-0000001e]
<< [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/300-0000001e]
<< [ HANGUP (NULL) ] [SIP/300-0000001e]
– Channel SIP/300-0000001e left ‘holding_bridge’ stasis-bridge <11aab1b8-5328-40d8-b707-92d0111bc1a3>
localhost*CLI>

This is a very serious issue.

Please check it Thanks~~~

  was:
We use ARI /dial function to connect outbound in Bridge.
We don't send "Answer" message until outbound answer.

ARI /dial function disconnect Inbound call automatically if Outbound is "busy" or "discard" 
In case we don't send "Answer" message.

But ARI /dial function don't disconnect Inbound call if outbound is "busy" or "discard". 
In case we send "Answer" message first.

We must use delay send "Answer" message until outbound answer call because of "Billing Issue"(We can't bill Inbound user in case outbound don't answer call) 

ARI /originate function don't disconnect Inbound call if outbound is "busy" or "discard". 
Even if In case we don't send "Answer" message.

We think "/ originate" and "/ dial" are same operate in case of "No Answer"