I am using vicidialnow-ce-1.2_final on a dell poweredge 4400 with 3 gigs of ram and a digium 212 2 pri card card. I do not have a sound card in it so i am getting dsp errors in the trace below but i know how to resolve that and that is not my real problem.Tha calls from asterisk dial out and connect, but the dial that vicidial uses tries to dial out a sip or iax trunk from what i can see and doesn't use the extension.conf that asterisk uses....i cant find where vicidial is bypassing my zap trunks....I have all carriers inactive in vicidial, so i would think that it would use the only default trunks left to asterisk, but it doesn't seem to get that far in the dial progress.
Thanks in advance for any direction you can point me in.
I can dial out from asterisk and here is a trace of that..it connects to my phone no problem. No sound card...so i get dsp error...i will resolve that latter.
Dial From Console
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vici*CLI> dial 94309360
-- Executing AGI("OSS/dsp", "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 Dial("OSS/dsp", "Zap/g1/4309360||To") in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called g1/4309360
Aug 14 14:53:53 WARNING[9376]: chan_oss.c:585 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory
-- Zap/1-1 is proceeding passing it to OSS/dsp
Aug 14 14:53:53 WARNING[9376]: chan_oss.c:859 oss_indicate: Don't know how to display condition 15 on OSS/dsp
Aug 14 14:53:54 WARNING[9376]: chan_oss.c:585 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory
Aug 14 14:53:55 WARNING[9376]: chan_oss.c:585 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory
Aug 14 14:53:56 WARNING[9376]: chan_oss.c:585 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory
Aug 14 14:53:57 WARNING[9376]: chan_oss.c:585 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory
Aug 14 14:53:58 WARNING[9376]: chan_oss.c:585 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory
-- Zap/1-1 is making progress passing it to OSS/dsp
Aug 14 14:53:59 WARNING[9376]: chan_oss.c:859 oss_indicate: Don't know how to display condition 14 on OSS/dsp
Aug 14 14:53:59 WARNING[9376]: chan_oss.c:585 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory
-- Zap/1-1 is ringing
== 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
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
-- Zap/1-1 answered OSS/dsp
<< Console call has been answered >>
Aug 14 14:54:08 WARNING[9376]: chan_oss.c:585 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory
Aug 14 14:54:09 WARNING[9376]: chan_oss.c:585 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory
Aug 14 14:54:10 WARNING[9376]: chan_oss.c:585 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory
Aug 14 14:54:11 WARNING[9376]: chan_oss.c:585 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory
Aug 14 14:54:12 WARNING[9376]: chan_oss.c:585 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory
Aug 14 14:54:13 WARNING[9376]: chan_oss.c:585 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory
== Manager 'sendcron' logged off from 127.0.0.1
Aug 14 14:54:14 WARNING[9376]: chan_oss.c:585 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory
Aug 14 14:54:15 WARNING[9376]: chan_oss.c:585 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory
-- Channel 0/1, span 1 got hangup request, cause 16
-- Hungup 'Zap/1-1'
But when Vicidial tries to call i get
Vicidial Dialing
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-- Executing AGI("Local/917164309360@default-d76b,2", "agi://127.0.0.1:4577/call_log") in new stack
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing AGI("Local/917164309360@default-272d,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 Dial("Local/917164309360@default-d76b,2", "/17164309360||To") in new stack
Aug 14 14:56:18 WARNING[9829]: channel.c:2621 ast_request: No channel type registered for ''
Aug 14 14:56:18 NOTICE[9829]: app_dial.c:1076 dial_exec_full: Unable to create channel of type '' (cause 66 - Channel not implemented)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing Hangup("Local/917164309360@default-d76b,2", "") in new stack
== Spawn extension (default, 917164309360, 3) exited non-zero on 'Local/917164309360@default-d76b,2'
-- Executing DeadAGI("Local/917164309360@default-d76b,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----66-----CHANUNAVAIL----------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing Dial("Local/917164309360@default-272d,2", "/17164309360||To") in new stack
Aug 14 14:56:18 WARNING[9832]: channel.c:2621 ast_request: No channel type registered for ''
Aug 14 14:56:18 NOTICE[9832]: app_dial.c:1076 dial_exec_full: Unable to create channel of type '' (cause 66 - Channel not implemented)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing Hangup("Local/917164309360@default-272d,2", "") in new stack
== Spawn extension (default, 917164309360, 3) exited non-zero on 'Local/917164309360@default-272d,2'
-- Executing DeadAGI("Local/917164309360@default-272d,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----66-----CHANUNAVAIL----------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----66-----CHANUNAVAIL---------- completed, returning 0
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----66-----CHANUNAVAIL---------- completed, returning 0
== Manager 'sendcron' logged off from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
How can i make vicidial use the same medium to dial out that asterisk uses....
I am doing a straight broadcast box with a pri... no sip or iax at all on the box...I want it to dial out and leave a message...and for me that is legal, I don't have to worry about telemarketing rules and dnc lists
My zaptel.conf
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# Span 1: TE2/0/1 "T2XXP (PCI) Card 0 Span 1" B8ZS/ESF RED
span=1,1,0,esf,b8zs
# termtype: te
bchan=1-23
dchan=24
# Span 2: TE2/0/2 "T2XXP (PCI) Card 0 Span 2" B8ZS/ESF RED
#span=2,2,0,esf,b8zs
# termtype: te
#bchan=25-47
#dchan=48
# Span 3: ZTDUMMY/1 "ZTDUMMY/1 (source: RTC) 1"
# Global data
loadzone = us
defaultzone = us
zapata.conf
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[color=blue]; Zapata telephony interface
;
; Configuration file
[trunkgroups]
[channels]
switchtype=national
context=local
signalling=pri_cpe
group=1
channel => 1-23
language=en
rxwink=300 ; Atlas seems to use long (250ms) winks
usedistinctiveringdetection=yes
callerid=asreceived
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=400
rxgain=0.0
txgain=0.0
callgroup=1
pickupgroup=1
immediate=no
; Span 2: WCTDM/0 "Wildcard TDM400P REV E/F Board 1"
;signalling=fxo_ks
;context=from-internal
;group=2
;channel => 25
;channel 26, WCTDM, inactive.
;signalling=fxs_ks
;context=from-zaptel
;group=2
; Span 2: WCTDM/0 "Wildcard TDM400P REV E/F Board 1"
;signalling=fxo_ks
;context=from-internal
;group=2
;channel => 25
;channel 26, WCTDM, inactive.
;signalling=fxs_ks
;context=from-zaptel
;group=2
;channel => 27
;signalling=fxs_ks
;context=from-zaptel
;group=2
;channel => 28
;faxdetect=both
faxdetect=incoming
;faxdetect=outgoing
;faxdetect=no
;Include genzaptelconf configs
#include zapata-auto.conf
group=1
;Include AMP configs
#include zapata_additional.conf
My Extensions.conf
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[general]
static=yes
writeprotect=no
[globals]
CONSOLE=Console/dsp ; Console interface for demo
TRUNK=Zap/g1 ; Trunk interface
;TRUNKX=Zap/g2 ; 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 interface
;SIPtrunk=SIP/1234:PASSWORD@sip.provider.net ; SIP trunk
;TRUNKloop = IAX2/ASTloop:test@127.0.0.1:40569 ; used for blind
;TRUNKblind = IAX2/ASTblind:test@127.0.0.1:41569 ; used for
#include extensions-vicidial.conf
[trunkinbound]
; agent dial-in:
exten => 2345,1,Answer ; Answer the line
exten => 2345,2,AGI(agi-AGENT_dial_in.agi)
exten => 2345,3,Hangup
; DID call routing process
exten => _X.,1,AGI(agi-DID_route.agi)
; FastAGI for VICIDIAL/astGUIclient call logging
exten => h,1,DeadAGI(agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME})
[outbound]
exten => _1.,1,Dial(${TRUNK}/${EXTEN:1},,To)
exten => _1.,2,Congestion
[local]
;
; Master context for local, toll-free, and iaxtel calls only
;
ignorepat => 9
include => default
include => trunklocal
include => trunktollfree
[default]
include => vicidial-auto
; Local agent alert extensions
; Local agent alert extensions
exten => _8600XXX*.,1,AGI(agi-VDADfixCXFER.agi)
exten => _78600XXX*.,1,AGI(agi-VDADfixCXFER.agi)
; Local blind monitoring
exten => _08600XXX,1,Dial(${TRUNKblind}/6${EXTEN:1},55,To)
;;;;;;;;;; BEGIN Voicemail and Prompts Section ;;;;;;;;;;;;;;;;;;;;;;;
; Give voicemail at extension 8500
exten => 8500,1,VoicemailMain
exten => 8500,2,Goto(s,6)
; this is used to allow the GUI to send you directly into voicemail
; don't forget to set GUI variable $voicemail_exten to this extension
exten => 8501,1,VoicemailMain(s${CALLERIDNUM})
exten => 8501,2,Hangup
; this is used to allow the GUI to send live calls directly into voicemail
; don't forget to set GUI variable $voicemail_dump_exten to this extension
exten => _85026666666666.,1,Wait(1)
exten => _85026666666666.,2,Voicemail(${EXTEN:14}|u)
exten => _85026666666666.,3,Hangup
; prompts for recording AGI script, ID is 4321
; first variable is format (gsm/wav)
; second variable is timeout in milliseconds (default is 720000 [12 minutes])
exten => 8167,1,Answer
exten => 8167,2,AGI(agi-record_prompts.agi,wav-----720000)
exten => 8167,3,Hangup
exten => 8168,1,Answer
exten => 8168,2,AGI(agi-record_prompts.agi,gsm-----720000)
exten => 8168,3,Hangup
; playback of recorded prompts
exten => _851XXXXX,1,Answer
exten => _851XXXXX,2,Playback(${EXTEN})
exten => _851XXXXX,3,Hangup
; this is used for playing a message to an answering machine forwarded from AMD in VICIDIAL
exten => _7851XXXXX,1,WaitForSilence(2000,2) ; AMD got machine. leave message after recording
exten => _7851XXXXX,2,Playback(${EXTEN:1})
exten => _7851XXXXX,3,AGI(VD_amd_post.agi,${EXTEN:1})
exten => _7851XXXXX,4,Hangup
;;;;;;;;;; END Voicemail and Prompts Section ;;;;;;;;;;;;;;;;;;;;;;;;;
; FastAGI for VICIDIAL/astGUIclient call logging
exten => h,1,DeadAGI(agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME})
; Example phone extensions
; Extension 2000 Sipura/Linksys ATA line 1
;exten => 2000,1,Dial(sip/spa2000,30,to) ; Ring, 30 secs max
;exten => 2000,2,Voicemail,u2000 ; Send to voicemail...
; Extension 2001 Sipura/Linksys ATA line 2
;exten => 2001,1,Dial(sip/spa2001,30,to) ; Ring, 30 secs max
;exten => 2001,2,Voicemail,u2001 ; Send to voicemail...
; Extension 2102 rings Grandstream phone
;exten => 2102,1,Dial(sip/gs102,30,to) ; Ring, 30 secs max
;exten => 2102,2,Voicemail,u2102 ; Send to voicemail...
; Extension 401 rings the firefly softphone
;exten => 401,1,Dial((IAX2/firefly01@firefly01/s||t)
;exten => 401,2,Hangup
; 100-350 phone extensions now auto-generated
; extensions for other SIP and IAX call center phones
; cc100-cc150 SIP Phones
;exten => _1[0-5]X,1,Dial(sip/cc${EXTEN},20,to)
; cc300-cc350 IAX Phones
;exten => _3[0-5]X,1,Dial(IAX2/cc${EXTEN},20,to)
; extensions if using a T1 channelbank
exten => _19XX,1,Dial(Zap/${EXTEN:2},30,o)
exten => _19XX,2,Hangup
exten => _9XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log),Macro(trunkdial,${trunk}/${EXTEN:1},${trunk_cid})
; Extension 4001 rings Zap phone (this example for FXS on Zap port 1)
exten => 4001,1,Dial(Zap/1,30,o) ; ring Zap device 1
exten => 4001,2,Voicemail,u4001 ; Send to voicemail...
; # timeout invalid rules
exten => #,1,Playback(invalid) ; "Thanks for trying the demo"
exten => #,2,Hangup ; Hang them up.
exten => t,1,Goto(#,1) ; If they take too long, give up
exten => i,1,Playback(invalid) ; "That's not valid, try again"
; Inbound call from BINFONE
; exten => 1112223333,1,AGI(agi://127.0.0.1:4577/call_log)
; exten => 1112223333,2,Dial(sip/gs102,55,o)