[HELP] - Retrieve channel status

Asterisk Asterisk SIP
1

Hello everyone,
I’m newbie at asterisk, since I apologize now if I’m creating this topic in the wrong place…
I’m using Asterisk 14.6.2

I have a php application, which makes a connection via a local trunk to a number, when this connection is answered, I connect with an IVR.

In my last interaction in the dial plan, I dial with the option g through the SIP to an X exten.

My question is, when this call is terminated, is there any way to get channel status so that I can identify in php that this call was terminated?
Follows the last dial and log information.

 exten => 1,1,NoOp(CALL)
 same=>n,AGI(/var/www/html/tests_asterisk/return.php,TRANSFER,${id_dialing_list_number},${id_dialing_list})
 same=>n,Dial(SIP/g238/9998,60,g)
 same=>n,Log(NOTICE, ${DIALSTATUS})
 same=>n,Hangup()

    -- Executing [1@ivr:1] NoOp("Local/XXXXXXXXX@to-dialer-00000069;1", "CALL") in new stack
    -- Executing [1@ivr:2] AGI("Local/XXXXXXXXX@to-dialer-00000069;1", "/var/www/html/tests_asterisk/return.php,TRANSFER,,") in new stack
    -- Launched AGI Script /var/www/html/tests_asterisk/return.php
    -- <Local/XXXXXXXXX@to-dialer-00000069;1>AGI Script /var/www/html/tests_asterisk/return.php completed, returning 0
    -- Executing [1@ivr:3] Dial("Local/XXXXXXXXX@to-dialer-00000069;1", "SIP/g238/9998,,g") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/g238/9998
       > 0x7fa9c800a7f0 -- Strict RTP learning after remote address set to: 192.168.0.238:18318
    -- SIP/g238-000000a6 answered Local/XXXXXXXXX@to-dialer-00000069;1
    -- Channel SIP/g238-000000a6 joined 'simple_bridge' basic-bridge <1c0d6b80-e1d0-4b78-acb1-a4666ee91ace>
    -- Channel Local/XXXXXXXXX@to-dialer-00000069;1 joined 'simple_bridge' basic-bridge <1c0d6b80-e1d0-4b78-acb1-a4666ee91ace>
       > Move-swap optimizing Local/XXXXXXXXX@to-dialer-00000069;1 <-- SIP/g43-000000a5.
    -- Channel SIP/g43-000000a5 left 'simple_bridge' basic-bridge <77240d47-ba98-44e8-ae80-9794aa569c1b>
    -- Channel Local/XXXXXXXXX@to-dialer-00000069;1 left 'simple_bridge' basic-bridge <1c0d6b80-e1d0-4b78-acb1-a4666ee91ace>
    -- Channel SIP/g43-000000a5 swapped with Local/XXXXXXXXX@to-dialer-00000069;1 into 'simple_bridge' basic-bridge <1c0d6b80-e1d0-4b78-acb1-a4666ee91ace>
       > Bridge 1c0d6b80-e1d0-4b78-acb1-a4666ee91ace: switching from simple_bridge technology to native_rtp
       > Remotely bridged 'SIP/g43-000000a5' and 'SIP/g238-000000a6' - media will flow directly between them
    -- Channel Local/XXXXXXXXX@to-dialer-00000069;2 left 'simple_bridge' basic-bridge <77240d47-ba98-44e8-ae80-9794aa569c1b>
  == Spawn extension (to-dialer, XXXXXXXXX, 2) exited non-zero on 'Local/XXXXXXXXX@to-dialer-00000069;2'
       > 0x7fa9c800a7f0 -- Strict RTP learning after remote address set to: 192.168.0.238:18318
  == Spawn extension (ivr, 1, 3) exited non-zero on 'Local/XXXXXXXXX@to-dialer-00000069;1'
    -- Channel SIP/g238-000000a6 left 'native_rtp' basic-bridge <1c0d6b80-e1d0-4b78-acb1-a4666ee91ace>
    -- Channel SIP/g43-000000a5 left 'native_rtp' basic-bridge <1c0d6b80-e1d0-4b78-acb1-a4666ee91ace>
localhost*CLI>
2

You can use the h extension, it gets executed after a call has been terminated, from there you can send a notification to your php script that the calls has been terminated

3

Or you can use Hangup Handlers to notify external application about the channel state

–Satish Barot

4

Good morning,
First of all, thank you for the answers. The two answers worked perfectly, I make the calls, and I get the status handles, but I realize it did not work out the way I wanted it to.

In both attempts we can see two “CHANNEL LEFT” at the end, that’s where my operator turned off the call, that’s exactly what I need to send to php, I need to know when the call is active or not after I send it to the “operator” in the other GW

Below is the logs with h extension and using the Hangup Handdles

H Extension

cat full.H-extension.bkp
[2017-10-04 11:21:14.410] VERBOSE[10993] dial.c: Called 111111111@to-dialer
[2017-10-04 11:21:14.410] VERBOSE[10994][C-00000122] pbx.c: Executing [111111111@to-dialer:1] NoOp("Local/111111111@to-dialer-0000009a;2", "Registring call on Dialer - 111111111") in new stack
[2017-10-04 11:21:14.410] VERBOSE[10994][C-00000122] pbx.c: Executing [111111111@to-dialer:2] Dial("Local/111111111@to-dialer-0000009a;2", "SIP/g43/111111111,,g") in new stack
[2017-10-04 11:21:14.411] VERBOSE[10994][C-00000122] netsock2.c: Using SIP RTP CoS mark 5
[2017-10-04 11:21:14.412] VERBOSE[10994][C-00000122] app_dial.c: Called SIP/g43/111111111
[2017-10-04 11:21:17.723] VERBOSE[1436][C-00000122] res_rtp_asterisk.c: 0x7fa9e000af90 -- Strict RTP learning after remote address set to: 192.168.0.43:12510
[2017-10-04 11:21:17.723] VERBOSE[10994][C-00000122] app_dial.c: SIP/g43-000000f4 is making progress passing it to Local/111111111@to-dialer-0000009a;2
[2017-10-04 11:21:17.724] VERBOSE[10993] dial.c: Local/111111111@to-dialer-0000009a;1 is making progress
[2017-10-04 11:21:17.732] VERBOSE[10994][C-00000122] res_rtp_asterisk.c: 0x7fa9e000af90 -- Strict RTP switching to RTP target address 192.168.0.43:12510 as source
[2017-10-04 11:21:19.232] VERBOSE[10994][C-00000122] res_rtp_asterisk.c: 0x7fa9e000af90 -- Strict RTP learning complete - Locking on source address 192.168.0.43:12510
[2017-10-04 11:21:21.127] VERBOSE[10994][C-00000122] app_dial.c: SIP/g43-000000f4 answered Local/111111111@to-dialer-0000009a;2
[2017-10-04 11:21:21.128] VERBOSE[10993] dial.c: Local/111111111@to-dialer-0000009a;1 answered
[2017-10-04 11:21:21.129] VERBOSE[10993][C-00000123] pbx.c: Executing [s@ivr:1] NoOp("Local/111111111@to-dialer-0000009a;1", "CALL IN URA") in new stack
[2017-10-04 11:21:21.129] VERBOSE[10993][C-00000123] pbx.c: Executing [s@ivr:2] AGI("Local/111111111@to-dialer-0000009a;1", "/var/www/html/tests_asterisk/return.php,LISTENING,,") in new stack
[2017-10-04 11:21:21.129] VERBOSE[11016][C-00000122] bridge_channel.c: Channel SIP/g43-000000f4 joined 'simple_bridge' basic-bridge <5680a659-6236-449c-918b-3a8dabfb8af3>
[2017-10-04 11:21:21.137] VERBOSE[10993][C-00000123] res_agi.c: Launched AGI Script /var/www/html/tests_asterisk/return.php
[2017-10-04 11:21:21.147] VERBOSE[10994][C-00000122] bridge_channel.c: Channel Local/111111111@to-dialer-0000009a;2 joined 'simple_bridge' basic-bridge <5680a659-6236-449c-918b-3a8dabfb8af3>
[2017-10-04 11:21:21.234] VERBOSE[10993][C-00000123] res_agi.c: <Local/111111111@to-dialer-0000009a;1>AGI Script /var/www/html/tests_asterisk/return.php completed, returning 0
[2017-10-04 11:21:21.245] VERBOSE[10993][C-00000123] pbx.c: Executing [s@ivr:3] BackGround("Local/111111111@to-dialer-0000009a;1", "tt-monkeys") in new stack
[2017-10-04 11:21:21.246] VERBOSE[10993][C-00000123] file.c: <Local/111111111@to-dialer-0000009a;1> Playing 'tt-monkeys.gsm' (language 'en')
[2017-10-04 11:21:24.597] DTMF[11016][C-00000122] channel.c: DTMF begin '1' received on SIP/g43-000000f4
[2017-10-04 11:21:24.597] DTMF[11016][C-00000122] channel.c: DTMF begin passthrough '1' on SIP/g43-000000f4
[2017-10-04 11:21:24.597] DTMF[10993][C-00000123] channel.c: DTMF begin '1' received on Local/111111111@to-dialer-0000009a;1
[2017-10-04 11:21:24.597] DTMF[10993][C-00000123] channel.c: DTMF begin ignored '1' on Local/111111111@to-dialer-0000009a;1
[2017-10-04 11:21:25.087] DTMF[11016][C-00000122] channel.c: DTMF end '1' received on SIP/g43-000000f4, duration 560 ms
[2017-10-04 11:21:25.087] DTMF[11016][C-00000122] channel.c: DTMF end accepted with begin '1' on SIP/g43-000000f4
[2017-10-04 11:21:25.087] DTMF[11016][C-00000122] channel.c: DTMF end passthrough '1' on SIP/g43-000000f4
[2017-10-04 11:21:25.087] DTMF[10993][C-00000123] channel.c: DTMF end '1' received on Local/111111111@to-dialer-0000009a;1, duration 560 ms
[2017-10-04 11:21:25.087] DTMF[10993][C-00000123] channel.c: DTMF end passthrough '1' on Local/111111111@to-dialer-0000009a;1
[2017-10-04 11:21:25.087] VERBOSE[10993][C-00000123] pbx.c: Executing [1@ivr:1] NoOp("Local/111111111@to-dialer-0000009a;1", "CALL") in new stack
[2017-10-04 11:21:25.087] VERBOSE[10993][C-00000123] pbx.c: Executing [1@ivr:2] AGI("Local/111111111@to-dialer-0000009a;1", "/var/www/html/tests_asterisk/return.php,TRANSFER,,") in new stack
[2017-10-04 11:21:25.095] VERBOSE[10993][C-00000123] res_agi.c: Launched AGI Script /var/www/html/tests_asterisk/return.php
[2017-10-04 11:21:25.190] VERBOSE[10993][C-00000123] res_agi.c: <Local/111111111@to-dialer-0000009a;1>AGI Script /var/www/html/tests_asterisk/return.php completed, returning 0
[2017-10-04 11:21:25.201] VERBOSE[10993][C-00000123] pbx.c: Executing [1@ivr:3] Dial("Local/111111111@to-dialer-0000009a;1", "SIP/g238/9998,,g") in new stack
[2017-10-04 11:21:25.202] VERBOSE[10993][C-00000123] netsock2.c: Using SIP RTP CoS mark 5
[2017-10-04 11:21:25.204] VERBOSE[10993][C-00000123] app_dial.c: Called SIP/g238/9998
[2017-10-04 11:21:25.230] VERBOSE[1436][C-00000123] res_rtp_asterisk.c: 0x7fa9d800c220 -- Strict RTP learning after remote address set to: 192.168.0.238:10656
[2017-10-04 11:21:25.231] VERBOSE[10993][C-00000123] app_dial.c: SIP/g238-000000f5 answered Local/111111111@to-dialer-0000009a;1
[2017-10-04 11:21:25.232] VERBOSE[11031][C-00000123] bridge_channel.c: Channel SIP/g238-000000f5 joined 'simple_bridge' basic-bridge <ee598780-b34c-4cbb-b297-77c3a439104f>
[2017-10-04 11:21:25.233] VERBOSE[10993][C-00000123] bridge_channel.c: Channel Local/111111111@to-dialer-0000009a;1 joined 'simple_bridge' basic-bridge <ee598780-b34c-4cbb-b297-77c3a439104f>
[2017-10-04 11:21:25.237] VERBOSE[10994][C-00000122] bridge.c: Move-swap optimizing Local/111111111@to-dialer-0000009a;1 <-- SIP/g43-000000f4.
[2017-10-04 11:21:25.237] VERBOSE[10994][C-00000122] bridge_channel.c: Channel SIP/g43-000000f4 left 'simple_bridge' basic-bridge <5680a659-6236-449c-918b-3a8dabfb8af3>
[2017-10-04 11:21:25.237] VERBOSE[10994][C-00000122] bridge_channel.c: Channel Local/111111111@to-dialer-0000009a;1 left 'simple_bridge' basic-bridge <ee598780-b34c-4cbb-b297-77c3a439104f>
[2017-10-04 11:21:25.237] VERBOSE[10994][C-00000122] bridge_channel.c: Channel SIP/g43-000000f4 swapped with Local/111111111@to-dialer-0000009a;1 into 'simple_bridge' basic-bridge <ee598780-b34c-4cbb-b297-77c3a439104f>
[2017-10-04 11:21:25.237] VERBOSE[10994][C-00000122] bridge.c: Bridge ee598780-b34c-4cbb-b297-77c3a439104f: switching from simple_bridge technology to native_rtp
[2017-10-04 11:21:25.238] VERBOSE[10994][C-00000122] bridge_native_rtp.c: Remotely bridged 'SIP/g43-000000f4' and 'SIP/g238-000000f5' - media will flow directly between them
[2017-10-04 11:21:25.239] VERBOSE[10994][C-00000122] bridge_channel.c: Channel Local/111111111@to-dialer-0000009a;2 left 'simple_bridge' basic-bridge <5680a659-6236-449c-918b-3a8dabfb8af3>
[2017-10-04 11:21:25.239] VERBOSE[10994][C-00000122] pbx.c: Spawn extension (to-dialer, 111111111, 2) exited non-zero on 'Local/111111111@to-dialer-0000009a;2'
[2017-10-04 11:21:25.241] VERBOSE[10993][C-00000123] pbx.c: Spawn extension (ivr, 1, 3) exited non-zero on 'Local/111111111@to-dialer-0000009a;1'
[2017-10-04 11:21:25.242] VERBOSE[10993][C-00000123] pbx.c: Executing [h@ivr:1] NoOp("Local/111111111@to-dialer-0000009a;1", "Testing the H Extension") in new stack
[2017-10-04 11:21:25.242] VERBOSE[10993][C-00000123] pbx.c: Executing [h@ivr:2] Log("Local/111111111@to-dialer-0000009a;1", "NOTICE, ANSWER") in new stack
[2017-10-04 11:21:25.242] NOTICE[10993][C-00000123] Ext. h:  ANSWER
[2017-10-04 11:21:25.242] VERBOSE[10993][C-00000123] pbx.c: Executing [h@ivr:3] Hangup("Local/111111111@to-dialer-0000009a;1", "") in new stack
[2017-10-04 11:21:25.242] VERBOSE[10993][C-00000123] pbx.c: Spawn extension (ivr, h, 3) exited non-zero on 'Local/111111111@to-dialer-0000009a;1'
[2017-10-04 11:21:25.242] VERBOSE[1436][C-00000123] res_rtp_asterisk.c: 0x7fa9d800c220 -- Strict RTP learning after remote address set to: 192.168.0.238:10656
[2017-10-04 11:21:30.915] VERBOSE[11031][C-00000123] bridge_channel.c: Channel SIP/g238-000000f5 left 'native_rtp' basic-bridge <ee598780-b34c-4cbb-b297-77c3a439104f>
[2017-10-04 11:21:30.916] VERBOSE[11016][C-00000122] bridge_channel.c: Channel SIP/g43-000000f4 left 'native_rtp' basic-bridge <ee598780-b34c-4cbb-b297-77c3a439104f>

Hangup Handlers

cat full.hangup-handlers.bkp
[2017-10-04 11:18:59.932] VERBOSE[10610] dial.c: Called 111111111@to-dialer
[2017-10-04 11:18:59.932] VERBOSE[10611][C-00000120] pbx.c: Executing [111111111@to-dialer:1] NoOp("Local/111111111@to-dialer-00000099;2", "Registring call on Dialer - 111111111") in new stack
[2017-10-04 11:18:59.933] VERBOSE[10611][C-00000120] pbx.c: Executing [111111111@to-dialer:2] Set("Local/111111111@to-dialer-00000099;2", "CHANNEL(hangup_handler_push)=ivr,1,4(args)") in new stack
[2017-10-04 11:18:59.933] VERBOSE[10611][C-00000120] pbx.c: Executing [111111111@to-dialer:3] Dial("Local/111111111@to-dialer-00000099;2", "SIP/g43/111111111,,g") in new stack
[2017-10-04 11:18:59.934] VERBOSE[10611][C-00000120] netsock2.c: Using SIP RTP CoS mark 5
[2017-10-04 11:18:59.934] VERBOSE[10611][C-00000120] app_dial.c: Called SIP/g43/111111111
[2017-10-04 11:19:03.844] VERBOSE[1436][C-00000120] res_rtp_asterisk.c: 0x7fa9b400d730 -- Strict RTP learning after remote address set to: 192.168.0.43:18114
[2017-10-04 11:19:03.844] VERBOSE[10611][C-00000120] app_dial.c: SIP/g43-000000f2 is making progress passing it to Local/111111111@to-dialer-00000099;2
[2017-10-04 11:19:03.844] VERBOSE[10610] dial.c: Local/111111111@to-dialer-00000099;1 is making progress
[2017-10-04 11:19:03.856] VERBOSE[10611][C-00000120] res_rtp_asterisk.c: 0x7fa9b400d730 -- Strict RTP switching to RTP target address 192.168.0.43:18114 as source
[2017-10-04 11:19:05.356] VERBOSE[10611][C-00000120] res_rtp_asterisk.c: 0x7fa9b400d730 -- Strict RTP learning complete - Locking on source address 192.168.0.43:18114
[2017-10-04 11:19:07.482] VERBOSE[10611][C-00000120] app_dial.c: SIP/g43-000000f2 answered Local/111111111@to-dialer-00000099;2
[2017-10-04 11:19:07.482] VERBOSE[10610] dial.c: Local/111111111@to-dialer-00000099;1 answered
[2017-10-04 11:19:07.482] VERBOSE[10610][C-00000121] pbx.c: Executing [s@ivr:1] NoOp("Local/111111111@to-dialer-00000099;1", "CALL IN URA") in new stack
[2017-10-04 11:19:07.482] VERBOSE[10610][C-00000121] pbx.c: Executing [s@ivr:2] AGI("Local/111111111@to-dialer-00000099;1", "/var/www/html/tests_asterisk/return.php,LISTENING,,") in new stack
[2017-10-04 11:19:07.490] VERBOSE[10610][C-00000121] res_agi.c: Launched AGI Script /var/www/html/tests_asterisk/return.php
[2017-10-04 11:19:07.495] VERBOSE[10634][C-00000120] bridge_channel.c: Channel SIP/g43-000000f2 joined 'simple_bridge' basic-bridge <7a35c5dc-181d-45af-98d8-be1fd406b99a>
[2017-10-04 11:19:07.498] VERBOSE[10611][C-00000120] bridge_channel.c: Channel Local/111111111@to-dialer-00000099;2 joined 'simple_bridge' basic-bridge <7a35c5dc-181d-45af-98d8-be1fd406b99a>
[2017-10-04 11:19:07.583] VERBOSE[10610][C-00000121] res_agi.c: <Local/111111111@to-dialer-00000099;1>AGI Script /var/www/html/tests_asterisk/return.php completed, returning 0
[2017-10-04 11:19:07.595] VERBOSE[10610][C-00000121] pbx.c: Executing [s@ivr:3] BackGround("Local/111111111@to-dialer-00000099;1", "tt-monkeys") in new stack
[2017-10-04 11:19:07.595] VERBOSE[10610][C-00000121] file.c: <Local/111111111@to-dialer-00000099;1> Playing 'tt-monkeys.gsm' (language 'en')
[2017-10-04 11:19:10.144] DTMF[10634][C-00000120] channel.c: DTMF begin '1' received on SIP/g43-000000f2
[2017-10-04 11:19:10.144] DTMF[10634][C-00000120] channel.c: DTMF begin passthrough '1' on SIP/g43-000000f2
[2017-10-04 11:19:10.144] DTMF[10610][C-00000121] channel.c: DTMF begin '1' received on Local/111111111@to-dialer-00000099;1
[2017-10-04 11:19:10.144] DTMF[10610][C-00000121] channel.c: DTMF begin ignored '1' on Local/111111111@to-dialer-00000099;1
[2017-10-04 11:19:10.628] DTMF[10634][C-00000120] channel.c: DTMF end '1' received on SIP/g43-000000f2, duration 540 ms
[2017-10-04 11:19:10.628] DTMF[10634][C-00000120] channel.c: DTMF end accepted with begin '1' on SIP/g43-000000f2
[2017-10-04 11:19:10.628] DTMF[10634][C-00000120] channel.c: DTMF end passthrough '1' on SIP/g43-000000f2
[2017-10-04 11:19:10.629] DTMF[10610][C-00000121] channel.c: DTMF end '1' received on Local/111111111@to-dialer-00000099;1, duration 540 ms
[2017-10-04 11:19:10.629] DTMF[10610][C-00000121] channel.c: DTMF end passthrough '1' on Local/111111111@to-dialer-00000099;1
[2017-10-04 11:19:10.629] VERBOSE[10610][C-00000121] pbx.c: Executing [1@ivr:1] NoOp("Local/111111111@to-dialer-00000099;1", "CALL") in new stack
[2017-10-04 11:19:10.629] VERBOSE[10610][C-00000121] pbx.c: Executing [1@ivr:2] AGI("Local/111111111@to-dialer-00000099;1", "/var/www/html/tests_asterisk/return.php,TRANSFER,,") in new stack
[2017-10-04 11:19:10.637] VERBOSE[10610][C-00000121] res_agi.c: Launched AGI Script /var/www/html/tests_asterisk/return.php
[2017-10-04 11:19:10.725] VERBOSE[10610][C-00000121] res_agi.c: <Local/111111111@to-dialer-00000099;1>AGI Script /var/www/html/tests_asterisk/return.php completed, returning 0
[2017-10-04 11:19:10.736] VERBOSE[10610][C-00000121] pbx.c: Executing [1@ivr:3] Dial("Local/111111111@to-dialer-00000099;1", "SIP/g238/9998,,g") in new stack
[2017-10-04 11:19:10.738] VERBOSE[10610][C-00000121] netsock2.c: Using SIP RTP CoS mark 5
[2017-10-04 11:19:10.739] VERBOSE[10610][C-00000121] app_dial.c: Called SIP/g238/9998
[2017-10-04 11:19:10.755] VERBOSE[1436][C-00000121] res_rtp_asterisk.c: 0x7fa9b800e5e0 -- Strict RTP learning after remote address set to: 192.168.0.238:13546
[2017-10-04 11:19:10.756] VERBOSE[10610][C-00000121] app_dial.c: SIP/g238-000000f3 answered Local/111111111@to-dialer-00000099;1
[2017-10-04 11:19:10.757] VERBOSE[10644][C-00000121] bridge_channel.c: Channel SIP/g238-000000f3 joined 'simple_bridge' basic-bridge <ddfe251e-701a-4263-8658-859fa50f3488>
[2017-10-04 11:19:10.757] VERBOSE[10610][C-00000121] bridge_channel.c: Channel Local/111111111@to-dialer-00000099;1 joined 'simple_bridge' basic-bridge <ddfe251e-701a-4263-8658-859fa50f3488>
[2017-10-04 11:19:10.760] VERBOSE[10611][C-00000120] bridge.c: Move-swap optimizing Local/111111111@to-dialer-00000099;1 <-- SIP/g43-000000f2.
[2017-10-04 11:19:10.761] VERBOSE[10611][C-00000120] bridge_channel.c: Channel SIP/g43-000000f2 left 'simple_bridge' basic-bridge <7a35c5dc-181d-45af-98d8-be1fd406b99a>
[2017-10-04 11:19:10.761] VERBOSE[10611][C-00000120] bridge_channel.c: Channel Local/111111111@to-dialer-00000099;1 left 'simple_bridge' basic-bridge <ddfe251e-701a-4263-8658-859fa50f3488>
[2017-10-04 11:19:10.761] VERBOSE[10611][C-00000120] bridge_channel.c: Channel SIP/g43-000000f2 swapped with Local/111111111@to-dialer-00000099;1 into 'simple_bridge' basic-bridge <ddfe251e-701a-4263-8658-859fa50f3488>
[2017-10-04 11:19:10.761] VERBOSE[10611][C-00000120] bridge.c: Bridge ddfe251e-701a-4263-8658-859fa50f3488: switching from simple_bridge technology to native_rtp
[2017-10-04 11:19:10.761] VERBOSE[10611][C-00000120] bridge_native_rtp.c: Remotely bridged 'SIP/g43-000000f2' and 'SIP/g238-000000f3' - media will flow directly between them
[2017-10-04 11:19:10.762] VERBOSE[10611][C-00000120] bridge_channel.c: Channel Local/111111111@to-dialer-00000099;2 left 'simple_bridge' basic-bridge <7a35c5dc-181d-45af-98d8-be1fd406b99a>
[2017-10-04 11:19:10.762] VERBOSE[10611][C-00000120] pbx.c: Spawn extension (to-dialer, 111111111, 3) exited non-zero on 'Local/111111111@to-dialer-00000099;2'
[2017-10-04 11:19:10.763] VERBOSE[10611][C-00000120] app_stack.c: Local/111111111@to-dialer-00000099;2 Internal Gosub(ivr,1,4(args)) start
[2017-10-04 11:19:10.763] VERBOSE[10611][C-00000120] pbx.c: Executing [1@ivr:4] NoOp("Local/111111111@to-dialer-00000099;2", "Testing Hangup Handler") in new stack
[2017-10-04 11:19:10.763] VERBOSE[10611][C-00000120] pbx.c: Executing [1@ivr:5] Log("Local/111111111@to-dialer-00000099;2", "NOTICE, ANSWER") in new stack
[2017-10-04 11:19:10.764] NOTICE[10611][C-00000120] Ext. 1:  ANSWER
[2017-10-04 11:19:10.764] VERBOSE[10611][C-00000120] pbx.c: Executing [1@ivr:6] Hangup("Local/111111111@to-dialer-00000099;2", "") in new stack
[2017-10-04 11:19:10.764] VERBOSE[10611][C-00000120] app_stack.c: Spawn extension (ivr, 1, 6) exited non-zero on 'Local/111111111@to-dialer-00000099;2'
[2017-10-04 11:19:10.764] NOTICE[10611][C-00000120] app_stack.c: Local/111111111@to-dialer-00000099;2 Abnormal 'Gosub(ivr,1,4(args))' exit.  Popping routine return locations.
[2017-10-04 11:19:10.766] VERBOSE[10610][C-00000121] pbx.c: Spawn extension (ivr, 1, 3) exited non-zero on 'Local/111111111@to-dialer-00000099;1'
[2017-10-04 11:19:10.767] VERBOSE[1436][C-00000121] res_rtp_asterisk.c: 0x7fa9b800e5e0 -- Strict RTP learning after remote address set to: 192.168.0.238:13546
[2017-10-04 11:19:17.812] VERBOSE[10644][C-00000121] bridge_channel.c: Channel SIP/g238-000000f3 left 'native_rtp' basic-bridge <ddfe251e-701a-4263-8658-859fa50f3488>
[2017-10-04 11:19:17.813] VERBOSE[10634][C-00000120] bridge_channel.c: Channel SIP/g43-000000f2 left 'native_rtp' basic-bridge <ddfe251e-701a-4263-8658-859fa50f3488>
5

Use AMI (Asterisk Manager Interface) events.

6

I am using the AMI to originate calls.
Are you telling me to use the AMI at all times to check if the call is active or not?

7

If you want call status mid-call, AMI is probably the best option. You should be using events, rather than polling.

8

I need to know when the call is finished.
As with the logs above, the extension has hangup handlers, it does not seem to be working the way I need it, we can see that it works, but it seems like the “h extension” and hangup handlers are following in a different channel from the one that is finalized , is there any way to retrieve the status of this channel?

9

The only channel that will give you an interesting status is the B side. The A side is already up, so can either survive the B side or will give a normal clearing status.

If you want status on the A side of an originate that fails you need to use a local channel.