Outbound automatic call by file is detected as ANSWERED, but really It isn´t!

1

Hi. This was my experience:

I created an extension:

softphone
deny=0.0.0.0/0.0.0.0
dtmfmode=rfc2833
canreinvite=no
context=xxxxxxxx
host=dynamic
trustrpid=yes
sendrpid=no
type=friend
nat=yes
qualify=yes
qualifyfreq=60
transport=udp
avpf=no
icesupport=no
encryption=no
callgroup=
pickupgroup=
permit=0.0.0.0/0.0.0.0
callcounter=yes
faxdetect=no

100
callerid=My Id <100>
dial=SIP/100
mailbox=100@default
secret=YyYyYyYyYyYyYyYy

And Its dialplan:

[marcador-automatico]
exten => _[123]XX, 1, NoOp(Llamada a una extension desde MARCADOR AUTOMATICO)
exten => _[123]XX, n, Dial(SIP/${EXTEN},${GLOBAL(DIAL_TIMEOUT)})
exten => _[123]XX, n, NoOp(Marcador Automatico el resultado d contestar fue: ${DIALSTATUS})
exten => _[123]XX, n, Hangup()

exten => _XXXNXXXXXX, 1, NoOp(Llamada a un celular desde MARCADOR AUTOMATICO)
exten => _XXXNXXXXXX, n, Dial(${GLOBAL(CANAL_SALIDA)}, ${GLOBAL(DIAL_TIMEOUT)}, D(w03${EXTEN}))
exten => _XXXNXXXXXX, n, Hangup()

[mensaje-automatico]
exten => s, 1, NoOp(Marcador Automatico el resultado d contestar fue: ${DIALSTATUS})
exten=> s, n, Background(en/silence/1)
exten=> s, n, AMD()
exten=> s, n, NoOp(El resultado del analisis ${AMDSTATUS})
exten=> s, n, GotoIf("${AMDSTATUS}"=“HUMAN”?humn:mach)
exten=> s, n(humn), NoOp(Es un humano!!!)
exten=> s, n, Playback(/var/lib/asterisk/sounds/${GLOBAL(ORGANIZACION)}/marcadorAutomatico/tmp/mensaje-marcador)
exten=> s, n, Hangup()
exten=> s, n(mach), NoOp(Es una Maquina!!!)
exten=> s, n, WaitForSilence(4500,2,16)
exten=> s, n, Hangup()

I created 2 call files, one for the extension, one for a mobile:

Channel: LOCAL/100@marcador-automatico
CallerID: Marcador NSS <888888>
MaxRetries: 1
RetryTime: 1
WaitTime: 10
Account: marcador-aut
Context: mensaje-automatico
Extension: s
Priority: 1
;##############################

hannel: LOCAL/XXXXXXXXXX@marcador-automatico
CallerID: “Marcador NSS” <888888>
MaxRetries: 1
RetryTime: 1
WaitTime: 25
Account: marcador-aut
Context: mensaje-automatico
Extension: s
Priority: 1
;#######################

This was the situation: When extensioncall file was enabled, that worked fine. This is the output:

[root@astsk-local ~]# cp /var/lib/asterisk/sounds/softfarr/marcadorAutomatico/scripts/call/100-archivo.call /var/spool/asterisk/outgoing/;exit exit
– Attempting call on LOCAL/100@marcador-automatico for s@mensaje-automatico:1 (Retry 1)
– Executing [100@marcador-automatico:1] NoOp(“Local/100@marcador-automatico-00000014;2”, “Llamada a una extension desde MARCADOR AUTOMATICO”) in new stack
– Executing [100@marcador-automatico:2] NoOp(“Local/100@marcador-automatico-00000014;2”, “MARCADOR AUTOMATICO - Paso 1”) in new stack
– Executing [100@marcador-automatico:3] Dial(“Local/100@marcador-automatico-00000014;2”, “SIP/100,25”) in new stack
== Using SIP RTP CoS mark 5
– Called SIP/100
– SIP/100-00000020 is ringing
– SIP/100-00000020 answered Local/100@marcador-automatico-00000014;2
> Channel Local/100@marcador-automatico-00000014;1 was answered
– Executing [s@mensaje-automatico:1] NoOp(“Local/100@marcador-automatico-00000014;1”, "Marcador Automatico el resultado d contestar fue: ") in new stack
– Executing [s@mensaje-automatico:2] BackGround(“Local/100@marcador-automatico-00000014;1”, “en/silence/1”) in new stack
– <Local/100@marcador-automatico-00000014;1> Playing ‘en/silence/1.gsm’ (language ‘en’)
> 0xa73cf80 – Probation passed - setting RTP source address to 192.168.1.150:65356
== Spawn extension (marcador-automatico, 100, 3) exited non-zero on ‘Local/100@marcador-automatico-00000014;2’
– Executing [s@mensaje-automatico:3] AMD(“SIP/100-00000020”, “”) in new stack
– AMD: SIP/100-00000020 888888 (N/A) (Fmt: slin)
– AMD: initialSilence [4000] greeting [9000] afterGreetingSilence [2250] totalAnalysisTime [15000] minimumWordLength [100] betweenWordsSilence [50] maximumNumberOfWords [5] silenceThreshold [256] maximumWordLength [5000]
– AMD: Channel [SIP/100-00000020]. Changed state to STATE_IN_SILENCE
– AMD: Channel [SIP/100-00000020]. ANSWERING MACHINE: silenceDuration:4000 initialSilence:4000
– Executing [s@mensaje-automatico:4] NoOp(“SIP/100-00000020”, “El resultado del analisis MACHINE”) in new stack
– Executing [s@mensaje-automatico:5] GotoIf(“SIP/100-00000020”, ““MACHINE”=“HUMAN”?humn:mach”) in new stack
– Goto (mensaje-automatico,s,6)
– Executing [s@mensaje-automatico:6] NoOp(“SIP/100-00000020”, “Es un humano!!!”) in new stack
– Executing [s@mensaje-automatico:7] Playback(“SIP/100-00000020”, “/var/lib/asterisk/sounds/softfarr/marcadorAutomatico/tmp/mensaje-marcador”) in new stack
– <SIP/100-00000020> Playing ‘/var/lib/asterisk/sounds/softfarr/marcadorAutomatico/tmp/mensaje-marcador.gsm’ (language ‘en’)
[2016-07-05 14:00:17] NOTICE[2005]: chan_sip.c:28236 handle_request_register: Registration from ‘“907” sip:907@179.33.205.204:5060’ failed for ‘89.163.242.218:5117’ - Wrong password
== Spawn extension (mensaje-automatico, s, 7) exited non-zero on ‘SIP/100-00000020’
[2016-07-05 14:00:31] NOTICE[30012]: pbx_spool.c:427 attempt_thread: Call completed to LOCAL/100@marcador-automatico

When I got enabled mobile call file, then this happened: Asterisk detected ANSWERED status, then the audio began to be played but the mobile wasn’t ringing. When audio had advanced some seconds, mobile began to ring and when I attended the call the message ended. Then I hung up.

This is the output:

[root@astsk-local ~]# cp /var/lib/asterisk/sounds/softfarr/marcadorAutomatico/scripts/call/100-archivo.call /var/spool/asterisk/outgoing/;exit exit
– Attempting call on LOCAL/100@marcador-automatico for s@mensaje-automatico:1 (Retry 1)
– Executing [100@marcador-automatico:1] NoOp(“Local/100@marcador-automatico-00000014;2”, “Llamada a una extension desde MARCADOR AUTOMATICO”) in new stack
– Executing [100@marcador-automatico:2] NoOp(“Local/100@marcador-automatico-00000014;2”, “MARCADOR AUTOMATICO - Paso 1”) in new stack
– Executing [100@marcador-automatico:3] Dial(“Local/100@marcador-automatico-00000014;2”, “SIP/100,25”) in new stack
== Using SIP RTP CoS mark 5
– Called SIP/100
– SIP/100-00000020 is ringing
– SIP/100-00000020 answered Local/100@marcador-automatico-00000014;2
> Channel Local/100@marcador-automatico-00000014;1 was answered
– Executing [s@mensaje-automatico:1] NoOp(“Local/100@marcador-automatico-00000014;1”, "Marcador Automatico el resultado d contestar fue: ") in new stack
– Executing [s@mensaje-automatico:2] BackGround(“Local/100@marcador-automatico-00000014;1”, “en/silence/1”) in new stack
– <Local/100@marcador-automatico-00000014;1> Playing ‘en/silence/1.gsm’ (language ‘en’)
> 0xa73cf80 – Probation passed - setting RTP source address to 192.168.1.150:65356
== Spawn extension (marcador-automatico, 100, 3) exited non-zero on ‘Local/100@marcador-automatico-00000014;2’
– Executing [s@mensaje-automatico:3] AMD(“SIP/100-00000020”, “”) in new stack
– AMD: SIP/100-00000020 888888 (N/A) (Fmt: slin)
– AMD: initialSilence [4000] greeting [9000] afterGreetingSilence [2250] totalAnalysisTime [15000] minimumWordLength [100] betweenWordsSilence [50] maximumNumberOfWords [5] silenceThreshold [256] maximumWordLength [5000]
– AMD: Channel [SIP/100-00000020]. Changed state to STATE_IN_SILENCE
– AMD: Channel [SIP/100-00000020]. ANSWERING MACHINE: silenceDuration:4000 initialSilence:4000
– Executing [s@mensaje-automatico:4] NoOp(“SIP/100-00000020”, “El resultado del analisis MACHINE”) in new stack
– Executing [s@mensaje-automatico:5] GotoIf(“SIP/100-00000020”, ““MACHINE”=“HUMAN”?humn:mach”) in new stack
– Goto (mensaje-automatico,s,6)
– Executing [s@mensaje-automatico:6] NoOp(“SIP/100-00000020”, “Es un humano!!!”) in new stack
– Executing [s@mensaje-automatico:7] Playback(“SIP/100-00000020”, “/var/lib/asterisk/sounds/softfarr/marcadorAutomatico/tmp/mensaje-marcador”) in new stack
– <SIP/100-00000020> Playing ‘/var/lib/asterisk/sounds/softfarr/marcadorAutomatico/tmp/mensaje-marcador.gsm’ (language ‘en’)
[2016-07-05 14:00:17] NOTICE[2005]: chan_sip.c:28236 handle_request_register: Registration from ‘“907” sip:907@179.33.205.204:5060’ failed for ‘89.163.242.218:5117’ - Wrong password
== Spawn extension (mensaje-automatico, s, 7) exited non-zero on ‘SIP/100-00000020’
[2016-07-05 14:00:31] NOTICE[30012]: pbx_spool.c:427 attempt_thread: Call completed to LOCAL/100@marcador-automatico
astsk-local*CLI> !bash
[root@astsk-local ~]#
[root@astsk-local ~]# cp /var/lib/asterisk/sounds/softfarr/marcadorAutomatico/scripts/call/softfarr-2016-06-27-3176362421.call /var/spool/asterisk/outgoing/;exit
exit
[2016-07-05 14:00:57] NOTICE[2005]: chan_sip.c:28236 handle_request_register: Registration from ‘“200” sip:200@179.33.205.204:5060’ failed for ‘89.163.242.218:5083’ - Wrong password
– Attempting call on LOCAL/3176362421@marcador-automatico for s@mensaje-automatico:1 (Retry 1)
– Executing [3176362421@marcador-automatico:1] NoOp(“Local/3176362421@marcador-automatico-00000015;2”, “Llamada a un celular desde MARCADOR AUTOMATICO”) in new stack
– Executing [3176362421@marcador-automatico:2] Dial(“Local/3176362421@marcador-automatico-00000015;2”, “SIP/201,25,D(w033176362421)”) in new stack
== Using SIP RTP CoS mark 5
– Called SIP/201
– SIP/201-00000021 is ringing
– SIP/201-00000021 answered Local/3176362421@marcador-automatico-00000015;2
– Sending DTMF ‘w033176362421’ to the called party.
> 0xa73cf80 – Probation passed - setting RTP source address to 192.168.1.165:5012
> Channel Local/3176362421@marcador-automatico-00000015;1 was answered
– Executing [s@mensaje-automatico:1] NoOp(“Local/3176362421@marcador-automatico-00000015;1”, "Marcador Automatico el resultado d contestar fue: ") in new stack
– Executing [s@mensaje-automatico:2] BackGround(“Local/3176362421@marcador-automatico-00000015;1”, “en/silence/1”) in new stack
– <Local/3176362421@marcador-automatico-00000015;1> Playing ‘en/silence/1.gsm’ (language ‘en’)
== Spawn extension (marcador-automatico, 3176362421, 2) exited non-zero on ‘Local/3176362421@marcador-automatico-00000015;2’
– Executing [s@mensaje-automatico:3] AMD(“SIP/201-00000021”, “”) in new stack
– AMD: SIP/201-00000021 888888 (N/A) (Fmt: slin)
– AMD: initialSilence [4000] greeting [9000] afterGreetingSilence [2250] totalAnalysisTime [15000] minimumWordLength [100] betweenWordsSilence [50] maximumNumberOfWords [5] silenceThreshold [256] maximumWordLength [5000]
– AMD: Channel [SIP/201-00000021]. Changed state to STATE_IN_SILENCE
– AMD: Channel [SIP/201-00000021]. HUMAN: silenceDuration:2260 afterGreetingSilence:2250
– Executing [s@mensaje-automatico:4] NoOp(“SIP/201-00000021”, “El resultado del analisis HUMAN”) in new stack
– Executing [s@mensaje-automatico:5] GotoIf(“SIP/201-00000021”, ““HUMAN”=“HUMAN”?humn:mach”) in new stack
– Goto (mensaje-automatico,s,6)
– Executing [s@mensaje-automatico:6] NoOp(“SIP/201-00000021”, “Es un humano!!!”) in new stack
– Executing [s@mensaje-automatico:7] Playback(“SIP/201-00000021”, “/var/lib/asterisk/sounds/softfarr/marcadorAutomatico/tmp/mensaje-marcador”) in new stack
– <SIP/201-00000021> Playing ‘/var/lib/asterisk/sounds/softfarr/marcadorAutomatico/tmp/mensaje-marcador.gsm’ (language ‘en’)
[2016-07-05 14:01:44] NOTICE[2005]: chan_sip.c:28236 handle_request_register: Registration from ‘“100” sip:100@179.33.205.204:5060’ failed for ‘89.163.242.218:5092’ - Wrong password
– Executing [s@mensaje-automatico:8] Hangup(“SIP/201-00000021”, “”) in new stack
== Spawn extension (mensaje-automatico, s, 8) exited non-zero on ‘SIP/201-00000021’
[2016-07-05 14:01:56] NOTICE[30558]: pbx_spool.c:427 attempt_thread: Call completed to LOCAL/3176362421@marcador-automatico

How do I sync that the audio begins to be played only when the mobile is off the hook and not before as is happening now, even before the mobile starts ringing?

Thankyou in advance for your help.

2

The problem lies with your SIP peer. It is claiming the call is answered.

I’m not sure if this is a gateway device or an ITSP. If it is an analogue gateway, both it an the other end of the analogue line will need to support answer supervision and have it enabled.

Generally, for automated outgoing calls you either need a quality ITSP or an ISDN (digital) connection to the PSTN.

1 Like
3

The outgoing call is using an ATA (HT503) patched at my PSTN phone line (Movistar).Switching to a VoIP provider could work as expected? Is there anything to setup in ATA which helps positively?

4

Another “issue” I,ve detected: I had heard 2 tones when doing a call: mark the number, off the hook (1 tone). then a second tone, and then the phone begins to ring. Is it possible that Asterisk identify second tone as answered?

5

If it is a normal ATA (FXS interface) it should have no problem detecting answer. if that is the case, I can only assume it is providing some form of early media without using 183 Progress.

If it is a gateway (connected to exchange line), but it and the exchange need to support answer supervision.

Generally, though, this is a question about your ATA, not Asterisk.

6

Don’t try to make automate calls through pstn interfaces. It will not work.
If your provider doesn’t support answer supervision or answer on polarity reversal then you should check if your ata can progress the call. The HT503 might have some option but it doesn’t always work properly. Just avoid or use a sip trunk. It is not a problem of Asterisk although Asterisk doesn’t either have a mechanism to detect answer on PSTN interfaces.