Asterisk call timeout 1 second

Hello, i setup an asterisk server running vicidial web system alongside goip4. calls from campigns made in vicidial do go through but they only last 1 second

There’s not much information to suggest any solution. You should provide Asterisk CLI output for one of such calls. However I feel you better off asking this question on Vicidial forum.

I setup and congifured asterisk with goip. whan i make calls using console dial. everything is normal but when i setup a campaign from vicidial the call hangups after 1 second

here is sip.conf

[asttecs]
disallow=all
allow=ulaw
allow=alaw
type=peer
;username=7021
;secret=a5tt3ctech7021
host=192.168.1.3
dtmfmode=RFC2833
qualify=yes
context=from-pstn
insecure=port,invite



extensions.conf

[general]
static=yes
writeprotect=no

[globals]
CONSOLE=Console/dsp                             ; Console interface for de$
TRUNK=DAHDI/r1                                    ; Trunk interface
TRUNKX=DAHDI/r2                                 ; 2nd trunk interface
TRUNKIAX=IAX2/ASTtest1:test@10.10.10.16:4569    ; IAX trunk interface
TRUNKIAX1=IAX2/ASTtest1:test@10.10.10.16:4569   ; IAX trunk interface
TRUNKBINFONE=IAX2/1112223333:PASSWORD@iax.binfone.com   ; IAX trunk interf$
SIPtrunk=SIP/1234:PASSWORD@sip.provider.net     ; SIP trunk

#include extensions-vicidial.conf

[from-pstn]
exten => _7021,1,AGI(agi-DID_route.agi)
exten => _7021,n,Hangup()

;;;;;;;;;;;;;;;;;;;;;;Outgoing Dialplan;;;;;;;;;;;;;;;;;;;;;;;;

;exten => _77XXXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log);
;exten => _77XXXXXXXXXXX,n,Set(_Test1=${CALLERID(all)});
;exten => _77XXXXXXXXXXX,n,Dial(SIP/asttecs/${EXTEN:2},,L(180000));
;exten => _77XXXXXXXXXXX,n,Hangup();

;exten => _77XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log);
;exten => _77XXXXXXXXXX,n,Set(_Test1=${CALLERID(all)});
;exten => _77XXXXXXXXXX,n,Dial(SIP/asttecs/${EXTEN:2},,L(180000));
;exten => _77XXXXXXXXXX,n,Hangup();

exten => _0XXXXXXXX.,1,AGI(agi://127.0.0.1:4577/call_log);
exten => _0XXXXXXXX.,n,Dial(SIP/asttecs/${EXTEN},,L(180000));

exten => _0XXXXXXXX.,n,Hangup();

exten => _XXXXXXXX.,1,AGI(agi://127.0.0.1:4577/call_log);
exten => _XXXXXXXX.,n,Dial(SIP/asttecs/0${EXTEN},,L(180000));
exten => _XXXXXXXX.,n,Hangup();

;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;

`       > Refreshing DNS lookups.
  == Manager 'sendcron' logged on from 127.0.0.1
    -- Executing [783596630@default:1] AGI("Local/783596630@default-00000001;2", "agi://127.0.0.1:4577/call_log") in new stack
    -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=vb123140))
    -- <Local/783596630@default-00000001;2>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
    -- Executing [783596630@default:2] Dial("Local/783596630@default-00000001;2", "SIP/asttecs/0783596630,,L(180000)") in new stack
    -- Setting call duration limit to 180.000 seconds.
  == Using SIP RTP CoS mark 5
    -- Called SIP/asttecs/0783596630
    -- SIP/asttecs-00000003 is making progress passing it to Local/783596630@default-00000001;2
    -- SIP/asttecs-00000003 is making progress passing it to Local/783596630@default-00000001;2
    -- SIP/asttecs-00000003 answered Local/783596630@default-00000001;2
       > Channel Local/783596630@default-00000001;1 was answered.
    -- Executing [8366@default:1] AGI("Local/783596630@default-00000001;1", "agi://127.0.0.1:4577/call_log") in new stack
    -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=vb123140))
    -- <Local/783595530@default-00000001;1>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
    -- Executing [8366@default:2] AGI("Local/783596630@default-00000001;1", "agi-VDAD_ALL_outbound.agi,SURVEYCAMP-----LB") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi
    -- Executing [h@default:1] AGI("Local/783596630@default-00000001;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----17-----0") in new stack
    -- <SIP/asttecs-00000003>AGI Script agi-VDAD_ALL_outbound.agi completed, returning 0
    -- Executing [8366@default:3] AGI("SIP/asttecs-00000003", "agi-VDAD_ALL_outbound.agi,SURVEYCAMP-----LB") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi
    -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Mar 10 12:33:37] WARNING[20453]: channel.c:3622 ast_waitfordigit_full: The FD we were waiting for has something waiting. Waitfordigit returning numeric 1
[Mar 10 12:33:37] ERROR[20453]: utils.c:1234 ast_carefulwrite: write() returned error: Broken pipe
    -- <SIP/asttecs-00000003>AGI Script agi-VDAD_ALL_outbound.agi completed, returning 0
    -- Executing [s@ivrtest:1] Answer("SIP/asttecs-00000003", "") in new stack
    -- Executing [s@ivrtest:2] AGI("SIP/asttecs-00000003", "agi-VDAD_inbound_calltime_check.agi,CALLMENU-----YES-----ivrtest-------------------------NO") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_inbound_calltime_check.agi
    -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
    -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
    -- <SIP/asttecs-00000003>AGI Script agi-VDAD_inbound_calltime_check.agi completed, returning 0
    -- Executing [s@ivrtest:3] Set("SIP/asttecs-00000003", "INVCOUNT=0") in new stack
    -- Executing [s@ivrtest:4] BackGround("SIP/asttecs-00000003", "") in new stack
[Mar 10 12:33:38] WARNING[20453]: pbx.c:10008 pbx_builtin_background: Background requires an argument (filename)
  == Spawn extension (ivrtest, s, 4) exited non-zero on 'SIP/asttecs-00000003'
    -- Executing [h@ivrtest:1] AGI("SIP/asttecs-00000003", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
    -- <SIP/asttecs-00000003>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0
  == Manager 'sendcron' logged off from 127.0.0.1
    -- <Local/783596630@default-00000001;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----17-----0 completed, returning 0
`

It tells you exactly why:

[Mar 10 12:33:38] WARNING[20453]: pbx.c:10008 pbx_builtin_background: Background requires an argument (filename)

You didn’t tell Background what sound to play, so it terminated the call as that is a fatal error.

i did select the ivr to play from vicidial

Vicidial is not a project or solution from Sangoma or the Asterisk project, so I can’t comment on what it does. I can only state what Asterisk was told to do.

ok, how can i tell background what to play from asterisk

If Vicidial is in control of your dialplan, then that’d be a Vicidial question. Otherwise I can say that it is at priority “4” of extension “s” in the “ivrtest” context. This is stated in the log message as well,

Executing [s@ivrtest:4]

thanks I will forward it to vicidial. by the way ivrtest is the recording(sound)