manual call

[Yamazaki]

im using goautodial 3.0 ce system, wanna know how to disable manual calls directly from softphone to obligate agent to call from web interface, and why when i call from my agent interface i cant hear ringing sound? Im still waiting the call till i receive the answer without ringing. Thanks in advance

[williamconley]

Yamazaki: I know you can hear me, as you have responded to my posts on your threads. But I'll try this again and see if this little extra note wakes you up just a bit:

1) Welcome to the Party!

2) As you are obviously new here, I have some suggestions to help us all help you:

When you post, please post your entire configuration including (but not limited to) your installation method and vicidial version with build.

This IS a requirement for posting along with reading the stickies (at the top of each forum) and the manager's manual (available on EFLO.net, both free and paid versions)

You should also post: Asterisk version, telephony hardware (model number is helpful here), cluster information if you have one, and whether any other software is installed in the box. If your installation method is "from scratch" you must post your operating system and should also post the .iso version from which you installed your original operating system. If your installation is "Hosted" list the site name of the host.

If this is a "Cloud" or "Virtual" server, please note the technology involved along with the version of that techology (ie: VMware Server Version 2.0.2). If it is not, merely stating the Motherboard model # and CPU would be helpful.

Similar to This:

Vicibox X.X from .iso | Vicidial X.X.X-XXX Build XXXXXX-XXXX | Asterisk X.X.X | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation | Intel DG35EC | Core2Quad Q6600

3) Note that above it states that at an Absolute Minimum you should include your Vicidial Version with Build. You think you could arrange that? It may be hidden in the GoAutoDial stuff ... if so let me know and I'm sure we can help you find it. But please post it on all future support requests.

4) This is actually a Vicidial question, not so much a GoAutoDial question. Unless Gardo has a cool wizard that will accomplish the task in question (possible, I've not heard of it ...), this question would be best asked in the Support board instead of the Goautodial board. Goautodial is merely the Installer for Vicidial, it's not actually the Dialer software. It does have a lot of pretty wizards now, though, so it's difficult to get past them to the Vicidial screens.

5) Once you dig down to the Vicidial Screens there are several methods to disallow agent direct dialing from their phones. One is to use a Dial Prefix which can be a single digit or multiple digits. This allows the agents to dial from their soft phones ONLY if they know the dial prefix. The dial prefix must be added to the Campaign Dial Prefix field and must match the number you add to all "exten" entries in your carrier Dialplan Entry. And then it must be removed before sending to the carrier. For instance, a dial prefix of "4" would be put in the Campaign's Dial Prefix field and then all extensions would change by adding the 4:

if your dialplan exten was:

Code: Select all

exten=>_1NXXNXXXXXX,

ALL lines with the old exten entry would merely have the 4 added:

Code: Select all

exten=>_41NXXNXXXXXX,

Then to remove the 4 before sending to the carrier, the "dial" line would require changing the "EXTEN" to "EXTEN:1" to remove that first digit. If it was already "EXTEN:1", it would change to "EXTEN:2". Example with "EXTEN:1":

Code: Select all

exten=>_41NXXNXXXXXX,n,Dial(${DIAL4TRUNK}/${EXTEN:1},,To)

6) But since annoying agents eventually find out (or just try every number from 1-1000 until they get lucky), you can also alter the agent PHONE (under admin->phones) to exist in an otherwise Nonexistent context. Normally both contexts are "default". Try changing either or both to "jailed" or "nocalls" and see what happens. Do be sure to test that the agent in question can still Log In and get calls as an agent after your change in addition to the lack of outbound calling. And remember: Changes take 60 seconds to register, so have patience before you test!

Happy Hunting!

[Yamazaki]

thanks williamconley u gave me a good detailled help.


this is my conf

1/ gotoautodial 3.0 CE
2/ Asterisk 1.4.39.1-vici.go
3/ i686 running Linux , single server
4/installation from scratch


[voip1]
srvlookup=yes
disallow=all
allow=g729
type=friend
authusername=XXXX72
username=XXXX72
secret=XXX18
host=sip.test.com
fromdomain=sip.test.com
authuser=XXXX72
fromuser=XXXX72
dtmfmode=rfc2833
qualify=1000


[voip2]
srvlookup=yes
disallow=all
allow=g729
type=friend
authusername=XXXX70
username=XXXX70
secret=XXX140
host=sip.test2.com
fromdomain=sip.test2.com
authuser=XXXX70
fromuser=XXXX70
dtmfmode=rfc2833
qualify=1000

#phones#

[8001]
username=8001
secret=goautodial
accountcode=8001
callerid="8001" <0000000000>
mailbox=8001
type=friend
host=dynamic
canreinvite=no
context=default
disallow=all
allow=g729
deny=0.0.0.0/0.0.0.0
permit=10.0.0.0/0.0.0.0

#dial plan#


; voip1
exten => _[89]ZXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _[89]ZXXXXXXX,2,Set(CALLFILENAME=${CALLERID}-${EXTEN}-${STRFTIME(${EPOCH},,$
exten => _[89]ZXXXXXXX,3,Monitor(wav,${CALLFILENAME},m)
exten => _[89]ZXXXXXXX,4,Dial(SIP/${EXTEN}@voip1,29,tToR)
exten => _[89]ZXXXXXXX,5,Hangup()


;voip2
exten => _[6]XXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _[6]XXXXXXXX,2,Set(CALLFILENAME=${CALLERID}-${EXTEN}-${STRFTIME(${EPOCH},,%$
exten => _[6]XXXXXXXX,3,Monitor(wav,${CALLFILENAME},m)
exten => _[6]XXXXXXXX,4,Dial(SIP/${EXTEN}@voip2,29,tToR)
exten => _[6]XXXXXXXX,5,Hangup()


so what i understand from u is to change

exten => _[6]XXXXXXXX,4,Dial(SIP/${EXTEN}@voip2,29,tToR)

To

exten => _33[6]XXXXXXXX,4,Dial(SIP/${EXTEN:1}@voip2,29,tToR)
33 as prefix, and why in interface web i cant hear ringing ??

[williamconley]

When you post, please post your entire configuration including (but not limited to) your installation method and vicidial version with build.

This IS a requirement for posting along with reading the stickies (at the top of each forum) and the manager's manual (available on EFLO.net, both free and paid versions)

You should also post: Asterisk version, telephony hardware (model number is helpful here), cluster information if you have one, and whether any other software is installed in the box. If your installation method is "from scratch" you must post your operating system and should also post the .iso version from which you installed your original operating system. If your installation is "Hosted" list the site name of the host.

If this is a "Cloud" or "Virtual" server, please note the technology involved along with the version of that techology (ie: VMware Server Version 2.0.2). If it is not, merely stating the Motherboard model # and CPU would be helpful.

Similar to This:

Vicibox X.X from .iso | Vicidial X.X.X-XXX Build XXXXXX-XXXX | Asterisk X.X.X | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation | Intel DG35EC | Core2Quad Q6600

[Yamazaki]

1/ gotoautodial 3.0 CE
2/ Asterisk 1.4.39.1-vici.go
3/ i686 running Linux , single server
4/installation from scratch

[williamconley]

1) it's either goautodial or from scratch. it can't be both.

2) vicidial version with build single most important piece of information, and often hidden by the goautodial wizards and screens (i often ponder as to whether that is an accident or if gardo is trying to hide the "Vicidial" facts, but mostly:) Since vicidial is the Dialer software and all the configuration done by the Goautodial wizards are there to configure Vicidial ... knowing the vicidial version and build is very important. When you dig down to the Vicidial core system (past all the wizards gardo has placed in the way) the vicidial version with build will be in the bottom left corner of the administration screens. Please remember to include it on all future posts. Makes it MUCH easier to assist you.

[Yamazaki]

sorry now i understand
this is what i found


VERSION: 2.4-309a
BUILD: 110430-1642

[Yamazaki]

is this enought ??

[williamconley]

When a manual dial call is initiated, the sound for ringing is expected to be passed back from the telco receiving the call (pre-answer sound). If they do not provide such a sound, you can attempt the "BriStuff" asterisk module with the "R" switch which (sometimes) provides the ringing sound when there is not one. If that fails due to technical reasons (usually compatibility), your only option is to use the loopback (dial 88) dialplan method. This generates a second call which allows asterisk to provide the ringing for the first call. But now you have two calls which increases your load. This method also allows the modification of the Caller ID Name field (handy in canada).

http://www.eflo.net/VICIDIALforum/viewtopic.php?p=59078

viewtopic.php?t=16625