Manual Dial Problem

[gmcust3]

Here is the way I follow to dial Manually :

STEPS to Dial Manually :

Agent Login to the Dialer.
Click on Manual Link
Enter 10 digit Phone Number.
Click on DIAL NOW.
Agent will hear a Beep sound and then a Silence.
When Customer Picks up , Agent can see the LIVE CALL GREEN on their CRM.
Agent speaks to the Customer and then Hang up and Dispose the Call.


When we dial Manually, from agent portal, there is NO GURANTEE that call will connect. Sometimes, it goes into BLANK with NO response as if there is dead air.

Am I making any Mistake OR its a Problem with the Vici ???

[williamconley]

sounds normal (if you are merely using vicidial as a phone system)

you should load the leads into the list for that campaign and hit "dial next number" instead. then the agent would no longer have to dial the numbers manually. no paper. no data entry at all.

next you can graduate to having vicidial "auto-dial" and change the way you do business by having your agents on another live call within a minute each time without dialing a number. (or as little as 10 seconds if configured properly with multiple agents on one campaign)

[gmcust3]

Agents get every day HOT HOT leads where they just call by clicking on Manual Dial Link and make call.

I asked the system admin to load leads but he said Agents are happy with copy paste.

But they are Tired of call going Blank and they want to hear the RING RING RING sound when ts actually ringing.

Can we make it happen by changing the dial plan or anything ?

[williamconley]

dunno,

what is your dial plan entry for your carrier?

search google for "asterisk cmd dial"

PS: (it's only three lines, so help me if you post your entire extensions.conf file i will mess up your karma for two years! LOL)

[gmcust3]

[S202_9]
disallow=all
allow=g729
allow=g711
allow=ulaw
type=friend
username=username
fromuser=username
secret=password
host=IP
dtmfmode=rfc2833


Global String :
---------------------------
SIP9 = SIP/S202_9

Dialplan Entry:
------------------------
exten => _9X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _9X.,2,Dial(${SIP9}/${EXTEN:2},,tTor)
exten => _9X.,3,Hangup

[williamconley]

you may want to investigate how this little bit works from extensions.conf:

Code: Select all

; This is a loopback dialaround to allow for hearing of ringing for 3way calls
exten => _881NXXNXXXXXX,1,Answer
exten => _881NXXNXXXXXX,2,Dial(${TRUNKloop}/9${EXTEN:2},,To)
exten => _881NXXNXXXXXX,3,Hangup


[gmcust3]

Should I change it to :

exten => _9X.,1,Answer
exten => _9X.,2,Dial(${SIP9}/${EXTEN:2},,tTor)
exten => _9X.,3,Hangup

tTor should be to To ??

I tried it and it shows

Waiting for Ring... 27 seconds.....

[williamconley]

sorry, no time for spoon feeding today. you'll have to look up the dial cmd as i suggested earlier and research the loopback trunk to find out how and when it is used to generate a ring tone.

i would have to ask you, if there is a special trunk to generate a ring tone, do you think they would have created such a thing if all they really had to do is "take out the 'r'"? i think there's a fundamental understanding missing in your suggestion.

you must TRY your solution, then try to find out how the sample loopback is used.

experiment!