How to enable this feature?

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How to enable this feature?

Postby rootpass » Tue Apr 29, 2008 6:05 pm

Hello guys, good day!

How do I enable this feature? :
Ability to autodial campaigns to start with a simple IVR then direct to agent


thank you very much!
rootpass
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Postby mflorell » Tue Apr 29, 2008 6:07 pm

The SURVEY agi script starts with a single IVR question and then sends the call to an agent. You would need to customize the script for your uses.
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Postby rootpass » Tue Apr 29, 2008 8:24 pm

thank you Matt
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Postby rootpass » Tue Apr 29, 2008 8:42 pm

Hi Matt,


I've seen the agi-VDADtransferSURVEY.agi and am not sure what to edit.

Here's what I wanted to do.

1. Dialer calls plays message giving the callee and option to be transfered to an agent by pressing 0.
2. Dialer transfers calls to the agent.

I see on the agi-VDADtransferSURVEY.agi

if ($pin == "1") {$voter = 'DEMOCRAT';}
if ($pin == "2") {$voter = 'REPUBLICAN';}
if ($pin == "3") {$voter = 'INDEPENDANT';}
if ($pin == "0") {$voter = 'OPERATOR';}

$stmtA = "UPDATE vicidial_list set security_phrase='$voter' where lead_id = '$CIDlead_id';";
if ($AGILOG) {$agi_string = "|$stmtA|"; &agi_output;}
$affected_rows = $dbhA->do($stmtA);
if ($AGILOG) {$agi_string = "-- VDAD vicidial_list VOTER update: |$affected_rows|$CIDlead_id|$voter"; &agi_output;}

# Thank you for your response, we would now like to transfer you to a poll-taker for just a few more questions. Please stay on the line.

$AGI->stream_file('US_pol_survey_transfer');


Do i upload my .gsm file and change the US_pol_survey_transfer to the file name of my .gsm file?

just wanted to confirm.

thanks,
rootpass
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Postby mflorell » Tue Apr 29, 2008 11:33 pm

Yes, you should replace the $voter values with the choices that you want.

Also, you will need to replace the US_pol_survey_transfer with the recording that you want to be played.
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Postby rootpass » Wed Apr 30, 2008 4:08 pm

Hi Matt,

By setting the Campaign VDAD exten to 8366 (so i thought) will this SURVEY work as is?

Although I just changed the AGI streams to the sound files I wanted to play. Leaving the "press" options as-is even with the press 8 for DNC. It doesn't seem to work?

on my

For the primary message when the callee answers

$interrupt_digit = $AGI->stream_file('Live_Answer','123456789');

when transferring to an agent:


# Thank you for your response, we would now like to transfer you to a poll-taker for just a few more questions. Please stay on the line.

$AGI->stream_file('RAG_Transfer');

And when "8" is pressed for DNC

else ### user pressed 8 to not be contacted again
{

$AGI->stream_file('RAG_DNC');

$stmtA = "DELETE FROM vicidial_auto_calls where callerid='$callerid' order by call_time desc limit 1;";
$affected_rows = $dbhA->do($stmtA);
if ($AGILOG) {$agi_string = "-- VDAD vac record deleted: |$affected_rows|$VDADcampaign|"; &agi_output;}



It doesn't seem to work. No message played when I answer a test call.


Is there anything else I should set on the Campaign settings?


thank you again sir!
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Postby mflorell » Wed Apr 30, 2008 4:56 pm

can you post some agiout.2008-04-XX logfile output for this script?
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Postby rootpass » Wed Apr 30, 2008 9:53 pm

hmmm, the latest agiout is agiout.2008-04-07 under folder /var/log/astguiclient

[root@server1 astguiclient]# tail -f agiout.2008-04-07
2008-04-07 13:00:59|VD_amd_post.agi| -- rdnis = unknown
2008-04-07 13:00:59|VD_amd_post.agi| -- request = VD_amd_post.agi
2008-04-07 13:00:59|VD_amd_post.agi| -- type = SIP
2008-04-07 13:00:59|VD_amd_post.agi| -- uniqueid = 1207585709.466
2008-04-07 13:00:59|VD_amd_post.agi|AGI Variables: |1207585709.466|SIP/accela1-09a2ef50|8320|SIP|V0407122829000004294|V0407122829000004294|3|
2008-04-07 13:00:59|VD_amd_post.agi|+++++ VD amd START: |4294|2008-04-07 13:00:59||
2008-04-07 13:00:59|VD_amd_post.agi||UPDATE vicidial_log set status='AL' where lead_id='4294' order by call_date desc limit 1;|
2008-04-07 13:00:59|VD_amd_post.agi|-- VDAD vicidial_log AM update: |1|1207585709.466|
2008-04-07 13:00:59|VD_amd_post.agi||UPDATE vicidial_list set status='AL' where lead_id='4294';|
2008-04-07 13:00:59|VD_amd_post.agi|-- VDAD vicidial_list AM update: |1|1207585709.466|





latest file i can find is vdautodial.2008-04-30


I just started testing SURVEY 2 days ago.


thank you again sir!
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Postby rootpass » Wed Apr 30, 2008 9:59 pm

sorry latest agiout is agiout.2008-04-15

2008-04-15 9:12:03|VD_amd_post.agi| -- rdnis = unknown
2008-04-15 9:12:03|VD_amd_post.agi| -- request = VD_amd_post.agi
2008-04-15 9:12:03|VD_amd_post.agi| -- type = SIP
2008-04-15 9:12:03|VD_amd_post.agi| -- uniqueid = 1208265010.3286
2008-04-15 9:12:03|VD_amd_post.agi|AGI Variables: |1208265010.3286|SIP/accela1-09aadb10|8320|SIP|V0415091010000010648|V0415091010000010648|3|
2008-04-15 9:12:03|VD_amd_post.agi|+++++ VD amd START: |10648|2008-04-15 9:12:03||
2008-04-15 9:12:03|VD_amd_post.agi||UPDATE vicidial_log set status='AL' where lead_id='10648' order by call_date desc limit 1;|
2008-04-15 9:12:03|VD_amd_post.agi|-- VDAD vicidial_log AM update: |1|1208265010.3286|
2008-04-15 9:12:03|VD_amd_post.agi||UPDATE vicidial_list set status='AL' where lead_id='10648';|
2008-04-15 9:12:03|VD_amd_post.agi|-- VDAD vicidial_list AM update: |1|1208265010.3286|


thank you!
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Postby mflorell » Thu May 01, 2008 1:27 am

Do you have the AGI Output for your server set to BOTH?

I need to see the SURVEY AGI output.
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Postby rootpass » Fri May 02, 2008 11:27 am

Hi Matt,

Am not sure if this is it.

[root@server1 astguiclient]# tail -f agiout.2008-05-02
2008-05-02 12:11:04|agi-VDADtransferSURVEY.agi||SELECT count(*) FROM vicidial_live_agents where callerid='V0502120946000015594';|
2008-05-02 12:11:04|agi-VDADtransferSURVEY.agi||SELECT count(*) FROM vicidial_auto_calls where callerid='V0502120946000015594' and status IN('LIVE','XFER');|
2008-05-02 12:11:04|agi-VDADtransferSURVEY.agi||UPDATE vicidial_list set status='PU' where lead_id = '15594';|
2008-05-02 12:11:04|agi-VDADtransferSURVEY.agi|-- VDAD vicidial_list update: |1|15594|XFER
2008-05-02 12:10:37|agi-VDADtransferSURVEY.agi|interrupt_digit |0|
2008-05-02 12:10:37|agi-VDADtransferSURVEY.agi|+++++ VDAD STATUS : pressed |1209744570.6|SIP/accela2-09f34a78|8366|SIP|V0502120930000015592|15592|2008-05-02 12:10:37
2008-05-02 12:10:48|agi-VDADtransferSURVEY.agi|interrupt_digit |0|
2008-05-02 12:10:48|agi-VDADtransferSURVEY.agi|+++++ VDAD STATUS : pressed |1209744571.9|SIP/accela2-09f4eb40|8366|SIP|V0502120930000015591|15591|2008-05-02 12:10:48
2008-05-02 12:11:04|agi-VDADtransferSURVEY.agi|interrupt_digit |0|
2008-05-02 12:11:04|agi-VDADtransferSURVEY.agi|+++++ VDAD STATUS : pressed |1209744586.18|SIP/accela2-09f5c290|8366|SIP|V0502120946000015594|15594|2008-05-02 12:11:04


By the way, it's now working where it plays the default file. But I already changed the US_pol_survey_hello to Live_Answer which is my Live_Answer.gsm file uploaded on the /sounds directory

# Hello, we are conducting a simple survery today, if you are a voter and registered as a Democrat, please press 1. If you are registered as a Republican, please press 2. If you are registered as an Independant or other party please press 3. If you do not want to be contacted again, please press 8.

$interrupt_digit = $AGI->stream_file('Live_Answer','123456789');


FYI: my setup has 1 db/Web server and 2 asterisk servers.

I edited the agi-VDADtransferSURVEY.agi file on asterisk server 1 and 2. Am I editing the wrong agi file? Should it be the one on the dB/Web server?

thanks again Matt, bare with my dumbness.
rootpass
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Postby rootpass » Fri May 02, 2008 12:09 pm

stupid me, really dumb

sounds should be uploaded at /var/lib/asterisk/sounds of both asterisk servers.


Thank you sir! I learned.

thanks,
rootpass
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Postby rootpass » Tue May 06, 2008 3:38 am

another question,

is it possible to automatically transfer a live call to an agent after the messages has been played if the callee hasn't pressed any button?


thank you again sirs!
rootpass
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Postby mflorell » Tue May 06, 2008 4:21 am

The program could be changed to do that, currently it does not by default.
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Postby rootpass » Tue May 06, 2008 5:58 am

ok, thank you :)
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Problem .....

Postby speed » Wed Apr 15, 2009 6:40 pm

Hello @ all,

i will do the same as rootpass.

i make an new extension 8370

this ist the my modify agi script.

<-- REMOVED -->

and he do only ring / then i answear / a few seconds und hangup.


here the cli

Executing AGI("Local/6004369981178753@default-0865,1", "agi-VDADtransferSURVEY_myphone.agi|8370") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDADtransferSURVEY_myphone.agi
-- AGI Script agi-VDADtransferSURVEY_myphone.agi completed, returning 0
-- Executing AGI("Local/6004369981178753@default-0865,1", "agi-VDADtransferSURVEY_myphone.agi|8370") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDADtransferSURVEY_myphone.agi
-- AGI Script agi-VDADtransferSURVEY_myphone.agi completed, returning 0
-- Executing AGI("Local/6004369981178753@default-0865,1", "agi-VDADtransferSURVEY_myphone.agi|8370") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDADtransferSURVEY_myphone.agi
-- AGI Script agi-VDADtransferSURVEY_myphone.agi completed, returning 0
-- Executing Hangup("Local/6004369981178753@default-0865,1", "") in new stack
== Spawn extension (default, 8370, 7) exited non-zero on 'Local/6004369981178753@default-0865,1'


thx for help
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Postby mflorell » Wed Apr 15, 2009 6:59 pm

Please don't post large files here, if you need to post a large script like this either post a DIFF against the standard file or post the file to the issue tracker and link to it here.

Are you using the 'o' Dial flag?

Are you using sip-silence Playback on this extension?

Why have you not upgraded to 2.0.5?
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trubbles

Postby speed » Thu Apr 16, 2009 10:43 am

Hello

The script works but only it calls and then it hangup after answear.
How can i set the flags ?
How can i uppdate my vicidialnow ?

thx Speed


-- Executing AGI("SIP/SIPtrunk2-09a9f508", "agi-VDADtransferSURVEY_myphone.a gi|8370") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDADtransferSURVEY_myph one.agi
-- AGI Script agi-VDADtransferSURVEY_myphone.agi completed, returning 0
-- Executing AGI("SIP/SIPtrunk2-09a9f508", "agi-VDADtransferSURVEY_myphone.a gi|8370") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDADtransferSURVEY_myph one.agi
-- AGI Script agi-VDADtransferSURVEY_myphone.agi completed, returning 0
-- Executing AGI("SIP/SIPtrunk2-09a9f508", "agi-VDADtransferSURVEY_myphone.a gi|8370") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDADtransferSURVEY_myph one.agi
-- AGI Script agi-VDADtransferSURVEY_myphone.agi completed, returning 0
-- Executing Hangup("SIP/SIPtrunk2-09a9f508", "") in new stack
== Spawn extension (default, 8370, 7) exited non-zero on 'SIP/SIPtrunk2-09a9f5 08'
-- Executing DeadAGI("SIP/SIPtrunk2-09a9f508", "agi://127.0.0.1:4577/call_lo g--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... EBUG-----1 6--------------- completed, returning 0
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Postby mflorell » Thu Apr 16, 2009 10:53 am

Please post what you have for exten 8370 in your extensions.conf
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8370

Postby speed » Thu Apr 16, 2009 12:20 pm

hello matt,
here the extension


; VICIDIAL_auto_dialer transfer script AMD with Load Balanced:
exten => 8370,1,Playback(sip-silence)
exten => 8370,n,AGI(agi://127.0.0.1:4577/call_log)
exten => 8370,n,AMD(2000|2000|1000|5000|120|50|4|256)
exten => 8370,n,AGI(VD_amd.agi,${EXTEN})
exten => 8370,n,AGI(agi-VDADtransferSURVEY_myphone.agi,${EXTEN})
exten => 8370,n,AGI(agi-VDADtransferSURVEY_myphone.agi,${EXTEN})
exten => 8370,n,AGI(agi-VDADtransferSURVEY_myphone.agi,${EXTEN})
exten => 8370,n,Hangup

thx speed
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Postby mflorell » Thu Apr 16, 2009 6:35 pm

What codec are you using?
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codecs

Postby speed » Fri Apr 17, 2009 11:12 am

from siptrunk..


allow=ulaw
allow=alaw
allow=gsm
type=friend

one of these .. i dont know ..

here the changes in script
;::::::::::::::::::::::::::::::::::::::::::::::::::::::::::::
$script = 'agi-VDADtransferSURVEY_myphone.agi';

:::::::::::::::::::::::::::::::::::::::::::::::::::::::::

if ($pin !~ /^1|^0/) {exit;}
::::::::::::::::::::::::::::::::::::::
if ($pin =~ /^1|^0/)
{
$voter='';

if ($pin == "1") {$voter = 'Agent';}
if ($pin == "0") {$voter = 'Hangup';}

::::::::::::::::::::::::::::::::::::::

$AGI->stream_file('danke_interesse');

:::::::::::::::::::::::::::::::::::::::...

$interrupt_digit = $AGI->stream_file('hello_intro_my_phone','123456789');

:::::::::::::::::::::::::::::::::::::::::..
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??

Postby speed » Mon Apr 20, 2009 3:49 am

??

Can anyone check this last thread please ?


thx ,, Speed
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Postby mflorell » Mon Apr 20, 2009 5:16 am

That did not tell me what codec is actually being used for the call, you should see the codec being defined in the Asterisk CLI when the call is sent to the carrier.
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