vicidial.org

i know that this is an asterisk question, but how can i use multiple carriers using only 1 prefix on the dialpan? that will choose the carrier simultaneously or randomly.

lets say i have carrier1, carrier2 and carrier3, is it possible to use them together using 1 prefix?

is this a possible for 2 carriers? :
exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91NXXNXXXXXX,n,Dial(${TRUNK1}/${EXTEN:1},,To)
exten => _91NXXNXXXXXX,n,Dial(${TRUNK1}/${EXTEN:1},,To)
exten => _91NXXNXXXXXX,n,Hangup

anyone please?

Hi,

You can do this,

exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91NXXNXXXXXX,n,Dial(sip/carrier1/${EXTEN:1},,To)
exten => _91NXXNXXXXXX,n,Dial(sip/carrier2/${EXTEN:1},,To)
exten => _91NXXNXXXXXX,n,Hangup

@ Op3r

I have the same desire to have 2 carriers running at the same time on the same campaign., So I do not understand your solution (not because it is unclear - I am just a novice)

So I have 2 carriers in Vicidial. Carrier A and B. Do I place the dial plan you suggested in both carrier settings? How exactly do I set this up?

this dialplan works like failover , if the first one fails to dial a number , the number will be dialled again in the second carrier.


@noworldorder
you need to put this dialpaln in any one of the carrier's dialplan entery.(not in both)


If you want to randomly use all the carries , below dialplan will be the one which i am using in my Freepbx


exten => _9044X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _9044X.,n,Set(Trunk=${RAND(1|3)})
exten => _9044X.,n,GoToIf($[${Trunk} = 1]?trunkA)
exten => _9044X.,n,GoToIf($[${Trunk} = 2]?trunkB)
exten => _9044X.,n,GoToIf($[${Trunk} = 3]?trunkC)
exten => _9044X.,n(trunkA),Dial(${TRUNKA}/${EXTEN:4},,tToR)
exten => _9044X.,n(trunkB),Dial(${TRUNKB}/${EXTEN:4},,tToR)
exten => _9044X.,n(trunkC),Dial(${TRUNKC}/${EXTEN:4},,tToR)
exten => _9044X.,n,Hangup

hope it will be helpfull.........

Thanks Striker! I need a bit more clarity (super novice, remember?)

Are you saying that all I do is change the dialplan of one of the two carriers and have them both active? Is that all?

If you look at my carrier setting below, can you tell me what I would change to have both of them calling on the same campaign?:

***********FIRST CARRIER *************
rapidvox

register => yyy:xxx@sip.rapidvox.com:5060

[rapidvox]
type=friend
dtmfmode=rfc2833
host=sip.rapidvox.com
context=outbound
username=yyy
fromuser=yyy
trustrpid=yes
sendrpid=yes
secret=xxx
disallow=all
allow=ulaw
nat=yes
qualify=1000

TrunkVox=SIP/rapidvox
exten => _X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _X.,n,Dial(${TrunkVox}/${EXTEN},10000,To)
exten => _X.,n,Hangup

***********SECOND CARRIER *************
switch2voip


register => yyy:xxx@66.33.147.150/5208913412

[switch2voip]
type=peer
username=yyy
host=66.33.147.150
fromuser=yyy
canreinvite=no
secret=xxx
bindport=5060
bindaddr=xx.xx.xxx.xx
context=default
disallow=all
allow=g729
dtmfmode=rfc2833
qualify=1000

switch2voip= SIP/switch2voip

exten => _X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _X.,n,Dial(${switch2voip}/${EXTEN},10000,To)
exten => _X.,n,Hangup

in rapidvox

put

exten => _7X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _7X.,n,Dial(${TrunkVox}/${EXTEN:1},10000,To)
exten => _7X.,n,Hangup

in switch2voip
exten => _8X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _8X.,n,Dial(${switch2voip}/${EXTEN:1},10000,To)
exten => _8X.,n,Hangup

and in same switch2voip
exten => _9X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _9X.,n,Dial(${switch2voip}/${EXTEN:1},10000,To)
exten => _9X.,n,Dial(${TrunkVox}/${EXTEN:1},10000,To)
exten => _9X.,n,Hangup

or
exten => _6X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _6X.,n,Set(Trunk=${RAND(1|2)})
exten => _6X.,n,GoToIf($[${Trunk} = 1]?trunkA)
exten => _6X.,n,GoToIf($[${Trunk} = 2]?trunkB)
exten => _6X.,n(trunkA),Dial(${TrunkVox}/${EXTEN:1},,tToR)
exten => _6X.,n(trunkB),Dial(${switch2voip}/${EXTEN:1},,tToR)
exten => _6X.,n,Hangup



then in campaing set the PREFIX as per ur requirement
7 to dial via rapidvox
8 to dial via switch2voip
9 to dial via the failover dialplan
6 to dial via the random dialplan

Thanks Striker (and everyone on this forum that helps)

IT WORKS!

one last thing... is ther a way to make it cycle through the given carriers sequentially rather than randomly.

Presumably this would be changed:

exten => _6X.,n,Set(Trunk=${RAND(1|2)})

thanks

please can someone help me on this. companion for the 7, 8, or 6 I must write'' 3-Way Call Dial Prefix:'' or'' Dial Prefix:'' or'' Dial Prefix:''
thank you

you can modify the system to "group" the carrier calls and count the calls in each group .. and use the next one as each fills up or use the one with the least active call. All sorts of things are possible.

you can also make the decision based on the time ... ie: 0-20 sec carrier 9, 21-40 sec carrier 8 ...

Thank you to answer me,
(I'm a beginner) if you can explain to me what I must change in the campaign and in the carrier:
Carrier1:
Account Entry: [SID121]
disallow=all
allow=g729
allow=ulaw
allow=G711a
allow=G711u
type=friend
username=XXXXXXXXXXXXXX
secret=XXXXXXXXXXXXX
host=XXXXXXXXXXXXXXXXX
dtmfmode=rfc2833
context=trunkinbound
insecure=invite,port

Dialplan Entry: exten=>_X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten=>_X.,2,Dial(SIP/SID121/${EXTEN},,tTor)
exten=>_X.,3,Hangup
Carrier2:
Account Entry: [SID104]
disallow=all
allow=g729
allow=ulaw
allow=G711a
allow=G711u
type=friend
username=XXXXXXXXXXXXXX
secret=XXXXXXXXXXXXX
host=XXXXXXXXXXXXXXXXX
dtmfmode=rfc2833
context=trunkinbound
insecure=invite,port

Dialplan Entry: exten=>_X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten=>_X.,2,Dial(SIP/SID104/${EXTEN},,tTor)
exten=>_X.,3,Hangup

Please, I want campaign1 uses Carrier1, and campaign2 uses Carrier2

when I replace the:
Dialplan Entry: exten=>_X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten=>_X.,2,Dial(SIP/SID104/${EXTEN},,tTor)
exten=>_X.,3,Hangup
with:
Dialplan Entry: exten=>_8X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten=>_8X.,2,Dial(SIP/SID104/${EXTEN},,tTor)
exten=>_8X.,3,Hangup
I get the message this extension is not available

When you add that "8" at the front of the dial pattern, you are using it to "select" this particular carrier. But you do not want to send that 8 to the carrier as part of the phone number, so it must be removed before sending the phone number.

This is done with ${EXTEN:1} instead of ${EXTEN}

The :1 will remove the first digit from the dial string.

Thank you it works.

Do you know how I can limit the duration of an outbond call? I want my agents to speak no more than 3 minutes.

Have a look here:

http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial

L(x[:y][:z]): Limit the call to 'x' ms, warning when 'y' ms are left, repeated every 'z' ms) Only 'x' is required, 'y' and 'z' are optional. Numbers must be integers- beware of AGI scripts that may return long integers in scientific notation (esp PHP 5.2.5&6) The following special variables are optional for limit calls: (pasted from app_dial.c)

First of all thank you very much.

But as a beginner I don't understand where I have to make those changes.

At the same time I'am looking at the link that you gave me.

Thanks

The second line of the Admin->Carrier entry for each carrier has a dialplan. The dialplan contains a "dial" command. This dial command has options ... such as a limiter for the length of each call.

Happy Hunting

please
it works. the duration of the call is 60 S and 30 S trigger Answering Machine Message.
but, in this manual, if you can guide me to do the same thing automatically.
thank you

Interesting question. Define "automatically". Once you modify the carrier dialplan, it's been modified.

I meant to say predictive calls, error came from the carrier. but I ais error messages. broken pipe. I just want the one calification automatically to troubleshoot .. VICIDIAL is what allows automatic calification?
thank you

When requesting support for an "error" it is customary to post the actual error seen as well as where you see it.

The broken pipe error is not actually an error. It is merely noting that there was nothing written in the log file.

LOL: I have NO idea what "one calification automatically to troubleshoot" means. I'd love to help, but that one baffles me.

Thank you for your efforts,
I wish that the statues be automated.
when the call is finished, the agent goes to another without choosing the status.

Disable Dispo is an option in later versions of Vicidial. It's a bad thing, as you are losing data, but it is available. You can also use Remote Agents with AutoAnswer on their phones, but once again, you lose a lot of the power of vicidial (data is power!). LOL

hello.
Im new in vicidial and i need to add a second carrier in the system. can some help please.
Hera is all the information received by provider>

Number: 123456789
Authentication Username: xxxxxxxx
Authentication Password: yyyyyyyy
Domain: telephony.aaa.
Protocol: UDP
Port: 5060

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) Adding a second carrier is much like adding the first carrier, assuming you used the manual and configured your first carrier as "9". In which case your next carrier would logically be "8" (and you would set the dial prefix for the campaign using the new carrier to "8" and you're all good).

You DID use the manual to set up your first carrier? Right?

Vicibox redux 4.0 from .iso | VERSION: 2.8-415a BUILD: 131007-1234 | Asterisk 1.4| Single Server QuadCore server with one procesor.

RukuBaruku wrote:Vicibox redux 4.0 from .iso | VERSION: 2.8-415a BUILD: 131007-1234 | Asterisk 1.4| Single Server QuadCore server with one procesor.

Vicibox has several "4.0.X" releases .. which one? Asterisk has MANY 1.4 releases .. which one?

3) Adding a second carrier is much like adding the first carrier, assuming you used the manual and configured your first carrier as "9". In which case your next carrier would logically be "8" (and you would set the dial prefix for the campaign using the new carrier to "8" and you're all good).

You DID use the manual to set up your first carrier? Right?