not able to autodial the leads

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not able to autodial the leads

Postby ramindia » Sun Nov 19, 2006 10:31 am

Hi all


iam able install from scratch_install
successfully in my FC5 box

with ASterisk 1.2.12.1
astguiclient_2.0.1

iam able to create a extension, and campaign also

and logged in

using x-lite logged in using one of the extension 1003(example)

http://ipaddress/agc/vicidial.php

logged in using username, as soon as i logged in

i get a incoming call on X-lite, tne i answer the call
its says you are the first person on the conference.


then i have enable to checkbox of LEAD PREVIEW

then i have clicked DIAL NEXT NUMBER

and in the form iam able to see the information i loaded in the conference

then i have Clickets the DIAL LEAD

in my x-lite i get auto recorded message start playing ( you have successfully installed astrix gui .. blabla)
but i dont see the call establishing.

I see only waiting for ring ...

then i get popup message.. dialtimeout, contact your administrator.

then i have clicked HANGUP customer
and iam still able to hear pre-recording sounds.

even i have changed the dial prefix to empty and X also
but there is no use.
( iam able to dial the same number using manual dial)
could some one tell me what is the wrong iam doing here at this step

why iam not able to autodial the lead.

here is the my screen output


screen -r
There are several suitable screens on:
2656.asterisk (Dead ???)
2754.ASTVDauto (Dead ???)
2725.ASTVDadapt (Dead ???)
3252.ASTupdate (Dead ???)
2817.ASTsend (Detached)
2692.ASTVDadapt (Dead ???)
2904.ASTupdate (Dead ???)
2761.ASTupdate (Dead ???)
2761.ASTVDremote (Dead ???)
2881.ASTVDremote (Dead ???)
2903.ASTVDremote (Dead ???)
2949.ASTupdate (Dead ???)
2692.asterisk (Dead ???)
2776.asterisk (Dead ???)
2747.ASTVDadapt (Dead ???)
3018.ASTVDremote (Detached)
2729.ASTlisten (Dead ???)
2759.ASTVDremote (Dead ???)
2852.ASTlisten (Dead ???)
2763.ASTupdate (Dead ???)
2889.ASTsend (Dead ???)
2696.ASTVDauto (Dead ???)
2810.ASTlisten (Detached)
2686.asterisk (Dead ???)
2856.ASTVDauto (Dead ???)
2899.ASTVDauto (Dead ???)
2952.ASTsend (Dead ???)
2883.ASTlisten (Dead ???)
2697.ASTVDauto (Detached)
2987.ASTupdate (Dead ???)
2653.asterisk (Dead ???)
2914.ASTVDremote (Dead ???)
2883.ASTVDremote (Dead ???)
811..vici (Dead ???)
2941.ASTsend (Dead ???)
2745.ASTsend (Dead ???)
2702.ASTVDadapt (Detached)
2695.ASTVDadapt (Dead ???)
2714.ASTVDadapt (Dead ???)
3240.ASTVDadapt (Dead ???)
2900.ASTVDauto (Dead ???)
2789.ASTlisten (Dead ???)
2907.ASTVDremote (Dead ???)
2739.ASTlisten (Dead ???)
2888.ASTlisten (Dead ???)
2791.asterisk (Dead ???)
2720.asterisk (Dead ???)
2962.ASTVDremote (Dead ???)
3296.ASTlisten (Dead ???)
2900.ASTupdate (Dead ???)
11279.ASTVDadapt (Dead ???)
2755.ASTVDauto (Dead ???)
2682.asterisk (Dead ???)
2748.ASTsend (Dead ???)
2859.ASTsend (Dead ???)
2860.ASTsend (Dead ???)
2874.ASTupdate (Dead ???)
2796.ASTsend (Dead ???)
2728.ASTupdate (Dead ???)
2910.ASTupdate (Dead ???)
2681.asterisk (Dead ???)
2738.ASTlisten (Dead ???)
2680.asterisk (Dead ???)
2905.ASTVDremote (Dead ???)
30880..vici (Dead ???)
2718.ASTVDauto (Dead ???)
2895.ASTVDadapt (Dead ???)
2769.asterisk (Detached)
2858.ASTsend (Dead ???)
30730.ASTlisten (Dead ???)
2966.ASTupdate (Dead ???)
2894.ASTVDadapt (Dead ???)
2675.ASTVDauto (Dead ???)
11294.ASTupdate (Dead ???)
2840.ASTsend (Dead ???)
11290.ASTVDremote (Dead ???)
2892.ASTVDadapt (Dead ???)
2866.ASTVDauto (Dead ???)
2684.asterisk (Dead ???)
2886.ASTVDadapt (Dead ???)
2879.ASTupdate (Dead ???)
2875.ASTlisten (Dead ???)
2947.ASTVDremote (Dead ???)
2868.ASTVDauto (Dead ???)
2899.ASTupdate (Dead ???)
581.ASTlisten (Dead ???)
2883.ASTsend (Dead ???)
2700.ASTVDauto (Dead ???)
2888.ASTVDauto (Dead ???)
11386.ASTsend (Dead ???)
2749.ASTVDadapt (Dead ???)
2880.ASTlisten (Dead ???)
3306.ASTsend (Dead ???)
2751.ASTVDadapt (Dead ???)
2876.ASTVDremote (Dead ???)
2896.ASTVDauto (Dead ???)
2679.ASTVDadapt (Dead ???)
2946.ASTlisten (Dead ???)
2685.asterisk (Dead ???)
2869.ASTVDadapt (Dead ???)
2902.ASTupdate (Dead ???)
2885.ASTsend (Dead ???)
2688.asterisk (Dead ???)
2908.ASTVDremote (Dead ???)
2833.ASTlisten (Dead ???)
2766.ASTVDremote (Dead ???)
3022.ASTupdate (Detached)
2742.ASTlisten (Dead ???)
2890.ASTVDauto (Dead ???)
2873.ASTupdate (Dead ???)
2887.ASTsend (Dead ???)
2983.ASTVDremote (Dead ???)
2896.ASTupdate (Dead ???)
2793.asterisk (Dead ???)
2757.ASTupdate (Dead ???)
2877.ASTupdate (Dead ???)
2897.ASTupdate (Dead ???)
2744.ASTsend (Dead ???)
2848.ASTlisten (Dead ???)
2879.ASTlisten (Dead ???)
2747.asterisk (Dead ???)
3016.ASTVDremote (Dead ???)
2891.ASTVDremote (Dead ???)
2750.ASTlisten (Dead ???)
2978.ASTVDauto (Dead ???)
2873.ASTVDremote (Dead ???)
2893.ASTVDadapt (Dead ???)
2937.ASTsend (Dead ???)
2865.ASTVDauto (Dead ???)
2904.ASTVDremote (Dead ???)
2894.ASTsend (Dead ???)
2957.ASTVDauto (Dead ???)
1502.ASTlisten (Dead ???)
2973.ASTsend (Dead ???)
2877.ASTlisten (Dead ???)
2904.ASTVDadapt (Dead ???)
2863.ASTVDadapt (Dead ???)
2967.ASTlisten (Dead ???)
2943.ASTVDremote (Dead ???)
2755.ASTlisten (Dead ???)
2881.ASTsend (Dead ???)
11284.ASTVDauto (Dead ???)
3236.ASTVDauto (Dead ???)
11337.ASTlisten (Dead ???)
3247.ASTVDremote (Dead ???)
2651.asterisk (Dead ???)
2752.ASTVDauto (Dead ???)
2877.ASTVDremote (Dead ???)
Remove dead screens with 'screen -wipe'.
Type "screen [-d] -r [pid.]tty.host" to resume one of them.



help will be appriciated

Ram
ramindia
 
Posts: 688
Joined: Wed Oct 11, 2006 4:06 am
Location: India

Postby Op3r » Sun Nov 19, 2006 10:46 am

type screen -wipe to remove the dead screen

then try again
Get paid for US outbound Toll Free calls. PM me.
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Postby ramindia » Sun Nov 19, 2006 12:48 pm

Hi

yes i have cleared screen -wipe

and rebooted the server

and try to dial , again i get the same voice

"you have succcessfully installed asterisk voice recording message"

screen -r
There are several suitable screens on:
2670.asterisk (Detached)
2733.ASTlisten (Detached)
2950.ASTsend (Detached)
2703.ASTVDauto (Detached)
2959.ASTVDremote (Detached)
2714.ASTupdate (Detached)
2699.ASTVDadapt (Detached)
Type "screen [-d] -r [pid.]tty.host" to resume one of them.


any help will be apprciated

Ram
ramindia
 
Posts: 688
Joined: Wed Oct 11, 2006 4:06 am
Location: India

Postby mflorell » Mon Nov 20, 2006 12:36 am

Please post Asterisk CLI output of when your try to place the first call.
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Postby ramindia » Mon Nov 20, 2006 1:11 am

Hi

when i click dial next number on /agc/vicidial.php

i get pre-recorded message on (x-lite)

and asterisk console i get the following message ( i have not enabled sip debug, its just console message, do i need to send debug also, just let me know)


== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing MeetMe("Local/8600051@default-d1dd,2", "8600051") in new stack
> Channel Local/8600051@default-d1dd,1 was answered.
== Starting Local/8600051@default-d1dd,1 at default,9117322180369,1 failed so falling back to exten 's'
-- Executing Wait("Local/8600051@default-d1dd,1", "1") in new stack
-- Executing Answer("Local/8600051@default-d1dd,1", "") in new stack
-- Executing Set("Local/8600051@default-d1dd,1", "TIMEOUT(digit)=5") in new stack
-- Digit timeout set to 5
-- Executing Set("Local/8600051@default-d1dd,1", "TIMEOUT(response)=10") in new stack
-- Response timeout set to 10
-- Executing BackGround("Local/8600051@default-d1dd,1", "demo-congrats") in new stack
-- Playing 'demo-congrats' (language 'en')
== Manager 'sendcron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Spawn extension (default, s, 5) exited non-zero on 'Local/8600051@default-d1dd,1'
== Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-d1dd,2'
== Manager 'sendcron' logged off from 127.0.0.1
== 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
== Manager 'sendcron' logged off from 127.0.0.1


Ram
ramindia
 
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Location: India

Postby abhasbajpai » Mon Nov 20, 2006 2:22 am

what happens when you dial manuly i mean straight from x-lite bypassing vicidial
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Postby ramindia » Mon Nov 20, 2006 3:20 am

Hi

when iam using with x-lite

call go perfect

there is no problem

Ram
ramindia
 
Posts: 688
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Location: India

Postby mflorell » Mon Nov 20, 2006 7:51 am

You do not have any entries in extensions.conf that will match 9117322180369

exten => 91XXXXXXXXXX,1,...
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Postby ramindia » Mon Nov 20, 2006 8:15 am

Hi


thanks for the reply

i have modified the extension.conf

now iam able to make calls out going.

what is the modification required for inbound support center

take example, if I have DID register with provider

when i call come in to DID, it should check the agents logged for that support group and transfer the call to them, if he is not answering , transfer the next available executive, and it so on, after 3 transfer, should go to voice mail.

how can i achive this ?

any help will be appriciated.

( iam also just analysing conf_examples, i see some examples of extension.conf, just looking around and see if i can get proper guidence from group also)

Ram
ramindia
 
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Location: India

Postby mflorell » Mon Nov 20, 2006 11:18 am

Have you read the manager manual?

From the example:
; inbound VICIDIAL call with CID delivery through T1 PRI
exten => 1234,1,Ringing
exten => 1234,2,Wait(1)
exten => 1234,3,Answer
exten => 1234,4,AGI(agi-VDADcloser_inboundCID.agi,CL_GALLERIA-----7275555134-----Closer-----park----------999-----1)
exten => 1234,5,Hangup


For more details on the parameters for agi-VDADcloser_inboundCID.agi look at the code of the script.
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Postby ramindia » Tue Nov 21, 2006 9:22 am

Hi Matt

thanks for suggestion

the example i see from T1/ PRI

but iam integrating with SIP provider.

i have few questions here is.

1 .what happends when the caller come with caller ID

will i get a gui forms to fill the information about the client information.
related to campaign.

like how we fill the campain form, closed or busy or not intrested something like.

2. what if the caller ID is Unknown, what happends to the call ?

is the call terminated ? or tranfer to the agent in the campaign ?


3. Lot people in the group i see they are using timer using digium cards

if i dont have digium cards installed in my Server what happends ?

ztdummy is recomended ?

4. for timer purpose i can use any dummy clone card of xp100 ?
or i should use only digium card only.

5. what is the recomendation for recording , IDE( 8000rpm SATA 2)
or SCSI 15000 RPM, what raid level ?

6. is this possible inbound and outbound campaign in one server.
( with the config of Xeon 3.0Ghz Dual, 4GB RAM, Gigabit ethernet hdd "
for recording suggest by the group IDE or SCSI")

or any other suggestion will be good.

Ram
ramindia
 
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Location: India

Postby mflorell » Tue Nov 21, 2006 11:37 am

SIP provider should be the same as PRI in terms of what they send, but you do need to know what digits they send you so you can enter it in the extensions.conf properly.

To lookup existing phone numbers in vicidial_list you would need to use the agi-VDADcloser_inboundCIDlookup script.agi

If a call comes in with no callerID it will still be sent onto an agent.

Digium/Sangoma cards are recommended, but an X100P should work just as well for a Zaptel timer. You can also use LInux 2.6.11 or higher with ztdummy for fairly good quality timer.

For a lot of recording you should use SCSI drives, I recommend RAID 10, but that is not required. You could also go with Western Digital Raptor drives, but they are not as fast as SCSI.

Yes you can do inbound and outbound on the same server, or even spread it out load-balanced across multiple servers if you like.
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Postby ramindia » Tue Nov 21, 2006 12:54 pm

Hi

you were mentioning with the kernel 2.6.X , ztdummy will do job

i have following version

Linux vici.mypc.com 2.6.18-1.2200.fc5 #1 SMP Sat Oct 14 16:59:56 EDT 2006 x86_64 x86_64 x86_64 GNU/Linux

if the above said verison should not to install xp100 card i belive ? for timer

what version is recomended zaptel driver. ( iam using zaptel-1.2.9.1)


ram
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Postby mflorell » Tue Nov 21, 2006 1:40 pm

For Linux 2.6.X kernel timing you need to set the frequency to 1000. I have only ever done this while building a new kernel, not sure how it would work if using a pre-compiled kernel. That's more of an issue for the Digium forums or the asterisk-users mailing list. You need at least a 2.6.11 kernel, so 2.6.18 is just fine.

Zaptel 1.2.9.1 should be fine.

An X100P at a minimum is always recommended if this will be a high volume production server for VICIDIAL.
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Postby ramindia » Sat Nov 25, 2006 5:19 am

Hi


i have checked

the installation version already with 1000Hz

so i belive i no need to change,

and i have low voice on other side

how can i increase the pitch of the voice

any suggestions

Ram
ramindia
 
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Postby mflorell » Sat Nov 25, 2006 8:10 am

Is the voice quiet even on directly dialed calls not done through VICIDIAL?
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Postby ramindia » Sat Nov 25, 2006 10:54 am

Hi

when dial from x-lite directly the voice is good quality
and iam able to hear very good voice

when i use dialer vici, then i have this problem

any suggestion

Ram
ramindia
 
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Postby mflorell » Mon Nov 27, 2006 10:47 am

Try an X100P card
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Postby ramindia » Mon Nov 27, 2006 11:14 am

Hi

thanks, i have ordered one

may next week i will get , let me install and come back to you
onthis issue later

ram
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