vicidial.org

by **djarum** » Wed Aug 26, 2009 11:07 am

Hi
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

Code: Select all: 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

Code: Select all: -- 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

Code: Select all: # 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

Code: Select all: [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

Code: Select all: [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)

by **williamconley** » Thu Sep 17, 2009 12:00 am

1) in your zapata.conf you will want to change the context to "trunkinbound" (you'll need it when you GET a call)

2) in your extensions.conf, which you posted, there is a certain amount of weirdness. Right about here:

; 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})

For starters, each "exten" is a little program. The lines of the program are "held together" by the actual extension. In this case the extension is "_19XX". Adding "_9XXXXXXXXXX" to this little group does not in any way "connect it" to the "_19XX" extension program, which is good because if I recall it is for another purpose anyway.

Also, you seem to have modified the dial plan to run in "local" instead of "default". This is not advisable until you have some experience with a working vicidial, so you can see what happens when you make changes to the extensions.conf (which you do NOT want to do ... really ... unless instructed and then MINOR changes ...!)

So, revert to the ORIGINAL extensions.conf file (or follow this and see if you can fix the one you have!), and you will find:

Code: Select all: ; dial a long distance outbound number ;This 'o' Dial flag is VERY important for VICIDIAL on outbound cal exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log) exten => _91NXXNXXXXXX,2,Dial(${TRUNKX}/${EXTEN:1},,To) exten => _91NXXNXXXXXX,3,Hangup

Now you need to EITHER change "TRUNKX" to "TRUNK" (which will point to ZAP/g1) or just change TRUNKX=ZAP/g2 to TRUNKX=ZAP/g1 in the [GLOBALS] at the beginning of the file.

This is how it will THEN work:
1) You dail "91{AREA}{PHONE}" (total 12 digits)
2) this will match the "pattern" (so defined by the _ in front of it) 91NXXNXXXXXX, so this three line program will be used to "route" your call
3) on line 1, it will activate the Vicidial system to monitor the call
4) on line two it will dial the call and use "TRUNKX" ...
5) The variable TRUNKX (in globals) will be translated to "Zap/g1"
6) The call will then be made through group 1 of your T1 (defined in zapata.conf).
Note: If you need to remove the "1" because they are local calls, make ANOTHER three line program, just like this one, for EACH AREA CODE that is local. But in each of these three, there will be a minor difference: Let us say your area code is "608":

Code: Select all: ; dial a long distance outbound number ;This 'o' Dial flag is VERY important for VICIDIAL on outbound cal exten => _91608NXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log) exten => _91608NXXXXXX,2,Dial(${TRUNKX}/${EXTEN:2},,To) exten => _91608NXXXXXX,3,Hangup

The obvious difference is the "608" which will be a closer "pattern" match and catch the 608 area code calls automatically. Then the not-so-obvious change of "EXTEN:2" will drop the "1" before sending the call to the carrier.

The reason for this is to allow the calls to be treated the SAME in vicidial, but allow asterisk to strip the "1". You don't have to change the campaign to "local dialing" or enter the leads with the "1" missing. This way Vicidial neither knows nor cares that they are local, but the calls have the 1 removed IF going out through this carrier. If you have another carrier for VOIP (dial 8?), the call would be dialed the same by Vicidial, but NOT have the "1" stripped because there would be no "_81608NXXXXXX" dial plan to catch them and strip the 1. So that call would succeed as well, with NO changes in Vicidial or the List.

by **djarum** » Mon Sep 28, 2009 12:15 pm

I made the changes suggested and tried a campaign and this is the reading from my cli.

I tried a test number in a variety of formats...and I think i changed the default to local in the extensions.conf the first time to try to circumvent the below problem.
I am trying to have a dialer using a pri card and just remote agents, to broadcast a series of messages out my pri trunks, and I think vicidial is still looking for sip trunks to use...all carriers are turned off in the admin pages...
TIA if anyone can point me in the right direction to get this running or if anybody has implemented this themselves and can give me a hint where to go next.
I just don't know if vicidial can do what i need it to do.

Code: Select all: Sep 17 15:50:03 NOTICE[5172]: chan_local.c:526 local_alloc: No such extension/context 4309360@default creating local channel Sep 17 15:50:03 NOTICE[5172]: channel.c:2514 __ast_request_and_dial: Unable to request channel Local/4309360@default == Parsing '/etc/asterisk/manager.conf': Found == Manager 'sendcron' logged on from 127.0.0.1 Sep 17 15:50:04 NOTICE[5173]: chan_local.c:526 local_alloc: No such extension/context 7164309360@default creating local channel Sep 17 15:50:04 NOTICE[5173]: channel.c:2514 __ast_request_and_dial: Unable to request channel Local/7164309360@default == Parsing '/etc/asterisk/manager.conf': Found == Manager 'sendcron' logged on from 127.0.0.1 Sep 17 15:50:04 NOTICE[5177]: chan_local.c:526 local_alloc: No such extension/context 7164309360@default creating local channel Sep 17 15:50:04 NOTICE[5177]: channel.c:2514 __ast_request_and_dial: Unable to request channel Local/7164309360@default == Parsing '/etc/asterisk/manager.conf': Found == Manager 'sendcron' logged on from 127.0.0.1 Sep 17 15:50:04 NOTICE[5179]: chan_local.c:526 local_alloc: No such extension/context 4309360@default creating local channel Sep 17 15:50:04 NOTICE[5179]: channel.c:2514 __ast_request_and_dial: Unable to request channel Local/4309360@default == Parsing '/etc/asterisk/manager.conf': Found == Manager 'sendcron' logged on from 127.0.0.1 Sep 17 15:50:04 NOTICE[5181]: chan_local.c:526 local_alloc: No such extension/context 7164309360@default creating local channel Sep 17 15:50:04 NOTICE[5181]: channel.c:2514 __ast_request_and_dial: Unable to request channel Local/7164309360@default == Parsing '/etc/asterisk/manager.conf': Found == Manager 'sendcron' logged on from 127.0.0.1 Sep 17 15:50:04 NOTICE[5183]: chan_local.c:526 local_alloc: No such extension/context 4309360@default creating local channel Sep 17 15:50:04 NOTICE[5183]: channel.c:2514 __ast_request_and_dial: Unable to request channel Local/4309360@default

by **williamconley** » Mon Sep 28, 2009 6:47 pm

interesting, you've shown that it didn't work, with indication of why.

what it is saying is:

Code: Select all: No such extension/context 7164309360@default creating local channel

Which means that there is no match or pattern match for 7164309360 in the default context. Which causes the dial to fail.

So: In your default context, is there a _7XXXXXXXXX or similar pattern?

This information would be in extensions.conf or another file included in extensions.conf, and would be in the context "default" or in another context included in "default".

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Asterisk Dials But vicidial wont..

Asterisk Dials But vicidial wont..

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