vicidial.org

Hi,

When I answer the phone of the called party by VD Autodial mode, the call doesnt seem to get passed to the free agent. My install is based on scratch docu, ztdummy, and SIP trunks only...

here's a sample of the asterisk screen

> Channel SIP/spa2001-08186a68 was answered.
== Manager 'sendcron' logged off from 127.0.0.1
-- Executing MeetMe("SIP/spa2001-08186a68", "8600051") in new stack
== Parsing '/etc/asterisk/meetme.conf': Found
-- Created MeetMe conference 1023 for conference '8600051'
-- Playing 'conf-onlyperson' (language 'en')
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing AGI("Local/912403961358@default-6005,2", "AGI(agi://127.0.0.1:4577/call_log") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/AGI(agi://127.0.0.1:4577/call_log
-- AGI Script AGI(agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing Dial("Local/912403961358@default-6005,2", "sip/12403961358@sipprovider|55|o") in new stack
-- Called 12403961358@sipprovider
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing AGI("Local/912403961358@default-bd72,2", "AGI(agi://127.0.0.1:4577/call_log") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/AGI(agi://127.0.0.1:4577/call_log
-- AGI Script AGI(agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing Dial("Local/912403961358@default-bd72,2", "sip/12403961358@sipprovider|55|o") in new stack
-- Called 12403961358@sipprovider
-- SIP/sipprovider-08183e48 is making progress passing it to Local/912403961358@default-6005,2
-- Got SIP response 503 "Service Unavailable" back from 64.192.112.13
-- SIP/sipprovider-081cd7b8 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing Hangup("Local/912403961358@default-bd72,2", "") in new stack
== Spawn extension (default, 912403961358, 3) exited non-zero on 'Local/912403961358@default-bd72,2'
-- Executing DeadAGI("Local/912403961358@default-bd72,2", "agi://127.0.0.1:4577/call_log") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing DeadAGI("Local/912403961358@default-bd72,2", "agi://127.0.0.1:4577/VD_hangup--HVcauses--PRI-----NODEBUG-----34-----CONGESTION----------)") in new stack
-- AGI Script agi://127.0.0.1:4577/VD_hangup--HVcause ... ----------) completed, returning 0
== Manager 'sendcron' logged off from 127.0.0.1
-- SIP/sipprovider-08183e48 is ringing
-- SIP/sipprovider-08183e48 is making progress passing it to Local/912403961358@default-6005,2
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
-- SIP/sipprovider-08183e48 answered Local/912403961358@default-6005,2
> Channel Local/912403961358@default-6005,1 was answered.
== Manager 'sendcron' logged off from 127.0.0.1
-- Executing AGI("Local/912403961358@default-6005,1", "agi://127.0.0.1:4577/call_log") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing AGI("Local/912403961358@default-6005,1", "agi-VDADtransfer.agi|8365") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDADtransfer.agi
== Manager 'sendcron' logged off from 127.0.0.1
-- AGI Script agi-VDADtransfer.agi completed, returning 0
-- Executing AGI("Local/912403961358@default-6005,1", "agi-VDADtransfer.agi|8365") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDADtransfer.agi
-- AGI Script agi-VDADtransfer.agi completed, returning 0
-- Executing AGI("Local/912403961358@default-6005,1", "agi-VDADtransfer.agi|8365") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDADtransfer.agi
-- AGI Script agi-VDADtransfer.agi completed, returning 0
-- Executing Hangup("Local/912403961358@default-6005,1", "") in new stack
== Spawn extension (default, 8365, 5) exited non-zero on 'Local/912403961358@default-6005,1'
-- Executing DeadAGI("Local/912403961358@default-6005,1", "agi://127.0.0.1:4577/call_log") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing DeadAGI("Local/912403961358@default-6005,1", "agi://127.0.0.1:4577/VD_hangup--HVcauses--PRI-----NODEBUG-----16---------------)") in new stack
-- AGI Script agi://127.0.0.1:4577/VD_hangup--HVcause ... ----------) completed, returning 0
== Spawn extension (default, 912403961358, 2) exited non-zero on 'Local/912403961358@default-6005,2'
-- Executing DeadAGI("Local/912403961358@default-6005,2", "agi://127.0.0.1:4577/call_log") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing DeadAGI("Local/912403961358@default-6005,2", "agi://127.0.0.1:4577/VD_hangup--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----18-----2)") in new stack
-- AGI Script agi://127.0.0.1:4577/VD_hangup--HVcause ... --18-----2) completed, returning 0


thanks for your help...

There you go, your asterisk said you have a problem with your sip provider.

kindly contact your sip provider.

Op3r wrote:There you go, your asterisk said you have a problem with your sip provider.

kindly contact your sip provider.

When I do manual calling it works fine. It looks like the call is not getting transferred to the correct conference. I think its the dial plan... I am still at a loss how to put entries in the extensions.conf. Is there any tutorial available out there besides wiki?...

What would be the correct entries for Autodial using SIP trunks?

have you tried the sip-silence Playback entries shown in the conf_examples extensions.conf?

mflorell wrote:have you tried the sip-silence Playback entries shown in the conf_examples extensions.conf?

I have these entries in my extensions.conf

;#### VDAD SIP UNREGISTERED TRANSFER ENTRIES ####
;#### Use these entries IN PLACE OF the entries above if you are using SIP trunks
;#### and are not registering your provider in sip.conf
; VICIDIAL_auto_dialer transfer script:
exten => 8365,1,Playback(sip-silence)
exten => 8365,2,AGI(agi://127.0.0.1:4577/call_log)
exten => 8365,3,AGI(agi-VDADtransfer.agi,${EXTEN})
exten => 8365,4,AGI(agi-VDADtransfer.agi,${EXTEN})
exten => 8365,5,Hangup

; VICIDIAL_auto_dialer transfer script SURVEY at beginning:
exten => 8366,1,Playback(sip-silence)
exten => 8366,2,AGI(agi://127.0.0.1:4577/call_log)
exten => 8366,3,AGI(agi-VDADtransferSURVEY.agi,${EXTEN})
exten => 8366,4,AGI(agi-VDADtransferSURVEY.agi,${EXTEN})
exten => 8366,5,Hangup

; VICIDIAL_auto_dialer transfer script Load Balance Overflow:
exten => 8367,1,Playback(sip-silence)
exten => 8367,2,AGI(agi://127.0.0.1:4577/call_log)
exten => 8367,3,AGI(agi-VDAD_LO_transfer.agi,${EXTEN})
exten => 8367,4,AGI(agi-VDAD_LO_transfer.agi,${EXTEN})
exten => 8367,5,Hangup

; VICIDIAL_auto_dialer transfer script Load Balanced:
exten => 8368,1,Playback(sip-silence)
exten => 8368,2,AGI(agi://127.0.0.1:4577/call_log)
exten => 8368,3,AGI(agi-VDAD_LB_transfer.agi,${EXTEN})
exten => 8368,4,AGI(agi-VDAD_LB_transfer.agi,${EXTEN})
exten => 8368,5,Hangup

; VICIDIAL_auto_dialer transfer script AMD with Load Balanced:
exten => 8369,1,Playback(sip-silence)
exten => 8369,2,AGI(agi://127.0.0.1:4577/call_log)
exten => 8369,3,AMD(3500|1500|300|5000|120|50|5|256)
exten => 8369,4,AGI(VD_amd.agi,${EXTEN})
exten => 8369,5,AGI(agi-VDAD_LB_transfer.agi,${EXTEN})
exten => 8369,6,AGI(agi-VDAD_LB_transfer.agi,${EXTEN})
exten => 8369,7,Hangup

; VICIDIAL auto-dial reminder script
exten => 8372,1,Playback(sip-silence)
exten => 8372,2,AGI(agi://127.0.0.1:4577/call_log)
exten => 8372,3,AGI(agi-VDADautoREMINDER.agi,${EXTEN})
exten => 8372,4,AGI(agi-VDADautoREMINDER.agi,${EXTEN})
exten => 8372,5,Hangup


what I am wondering is this part of the CLI output...(see below). it said it called the number... its making progress... and its passing it to "Local/912403961358@default-6005,2"

Shouldnt it be passing the call to the conference?

I am still reading up on how extensions.conf work. I am not sure if my entries there are correct or was entered in correct sequence...


== Manager 'sendcron' logged on from 127.0.0.1
-- Executing AGI("Local/912403961358@default-bd72,2", "AGI(agi://127.0.0.1:4577/call_log") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/AGI(agi://127.0.0.1:4577/call_log
-- AGI Script AGI(agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing Dial("Local/912403961358@default-bd72,2", "sip/12403961358@sipprovider|55|o") in new stack
-- Called 12403961358@sipprovider
-- SIP/sipprovider-08183e48 is making progress passing it to Local/912403961358@default-6005,2
-- Got SIP response 503 "Service Unavailable" back from 64.192.112.13
-- SIP/sipprovider-081cd7b8 is circuit-busy

can you show me sample .conf files for VICIDIAL in autodialmode using purely SIP trunks. thanks.

It's in the release you downloaded of vicidial/astGUIclient in the docs/example_configs folder.

thank you... I found my error though... mysql service failed at startup... everything works fine now...

really appreciate your help.
and when I get to know more I will take time to help others as well...
God Bless!