Transfer call from remote agent to DID

[mubeen]

Hi,

I have cluster installation with 4 servers, one each of Telephony, DB, archive and web

VERSION: 2.14-694a
BUILD: 181005-1738
ViciBox v.7.0.4-170113

I'm running press-1 campaign with remote agents. Normally call is transferred to an extension. Is there a way I can transfer the call to remote agent?

[williamconley]

Survey Method: AGENT_XFER

That's what it's for: Agents (remote or not) are supposed to be logged in to the campaign to activate dialing. When all agents are busy, dialing should stop.

[mubeen]

Actually what I'm looking for is a way to transfer calls from survey to a DID

[williamconley]

I bet you've just jumped into a semantic ditch here.

What do you mean by "DID"? You want to transfer the call to a Vicidial DID in the same server which is how vicidial gets Inbound calls?

Or you want to transfer these calls to another server as inbound, thus making it an outbound call for this server?

The BEST method to transfer calls to another server (like, if you're selling press 1 calls to another room for $10/pop) would be to put that external phone number in a remote agent, log the remote agent into the campaign, and set the "lines" to the number of agents they have in their call center (so you can route the overflow to a DIFFERENT client instead of having the calls go unanswered). Be sure to create a user for the remote agent whose number is nowhere near any other user since Vicidial will increment the user number for each successive Line (eg: if you put user 3001 in as the user for the remote agent with 10 lines, you'll end up with 3001-3010 in use, so be sure 3001-3010 are not already assigned to someone). Better yet, you can create 3001-3010 so you can see live calls in the real time screen.

IF, however, you want to send these calls to a DID on this server to transfer them to an agent (on this server), you can either route them directly to an ingroup instead (recommended) or use a call menu to route immediately to a DID with a timeout.

Probably be easier if we actually knew what your reason was.

[mubeen]

Trying to do the the 2nd option you mentioned

I'm running press-1 campaign, normally at dial level of 100 with 4-5 agents. For some reason all calls on which client presses a digit is not transferred to agents and is dropped.

I've seen others transferring the call from press-1 to some internal dummy DID using remote agent and then to ingroup and to agents.

[williamconley]

Your drop call seconds is probably set too high. If the client listened to an audio file, the time is already up before they press one.

[mubeen]

Well I can see the client press the digit in CLI, but call drops. My drop call time is 15 sec.

Anyway, How can I transfer it to a dummy DID or an ingroup?

[williamconley]

1) set it to a MUCH higher number. Who presses 1 within 15 seconds of answering?

2) TEST it yourself (put yourself in the list only, nobody else, with 1:1 dialing for test).

3) As stated previously, you can't transfer it to a DID, but you CAN transfer it to a call menu and then use the timeout feature of the call menu to transfer it anywhere you like (DID, ingroup, extension, whatever)

[mubeen]

Thank you William, have set it to 60 sec...time for some testing

Your help is always appreciated

[mubeen]

No, increasing drop call to 60 didn't resolved the issue.

If I create remote agent,

What will be "External extension"? As per manager manual, This is the number that you want the calls forwarded to. I want it forwarded to an Ingroup or call menu which I believe have to setup in campaign.

I have made 2 campaigns, one outbound for press-1 on which remote agent will be logged in and other inbound on which real agents will log in and take calls received via ingroup through call-menu.

Also the Server IP in remote agent will be of Telephony server, right?

[williamconley]

Also the Server IP in remote agent will be of Telephony server, right?

Um. NO.

I think you have probably skipped a Very Important Step.

If you start over and also start at page one of the Vicidial Manager's Manual, you'll likely end up with a fully functional server without having to reinvent any wheels or blow a month trying to make this work "the way you think it might at first glance, randomly, on each web page". The Manager's Manual exists for a reason. If you start at page one (and DO NOT skip anything), you'll have a much better experience. Seriously. Page one. No Skipping. Work your way through the entire thing for free. You'll come out the other side "ready to rock". But only if you avoid skipping pages you think you don't need.

If you intend to make money from this app for several years, investing a little time in the FREE Vicidial Manager's Manual now, up front, will be well worth your time invested.

When you hit a snag, provide the version/page/line of the manual and to state your problem tell us what happened (or didn't) and what you think should have happened (or not). You can also provide relevent settings from the broken item and even CLI output from a test call (beginning to end, one call: NOT 3000 lines of unrelated code and NOT just the end of the call).

I say all this because I get the general impression you are very close to asking how to hook your horse up to the "wagon" which is actually a car and doesn't need a horse. LOL

[geek_kaushal]

No, increasing drop call to 60 didn't resolved the issue.

If I create remote agent,

What will be "External extension"? As per manager manual, This is the number that you want the calls forwarded to. I want it forwarded to an Ingroup or call menu which I believe have to setup in campaign.

I have made 2 campaigns, one outbound for press-1 on which remote agent will be logged in and other inbound on which real agents will log in and take calls received via ingroup through call-menu.

Also the Server IP in remote agent will be of Telephony server, right?

Okay, lets pinpoint your situation here. Let me give you some options. lemme know what you intend to do.
1) You want to send the call to an external number (actual phone number) when customer presses 1. That external number could be a cell number or a DID pointed to the same server (or a different server) going in an in-group for agents to answer.
2) You want to create a dummy DID right in the same server, and point it to an in-group to connect to persons who press 1.
3) You want to send the call to a different server using asterisk to asterisk call transfer, and allow the agents to connect to press 1 optins in an in-group in a different server.

You can use any number in external extension as long as the survey method is not AGENT_XFER

[mubeen]

@ William,

Actually I was running above 600 concurrent calls on a cluster with single telephony server, system load was around 50, I thought it could handle that number of calls but forgot that Asterisk has its limits.
I understand the call will be thrown to call menu / ingroup through the campaign and remote agent is just another agent but I fail to understand the concept of external extension completely

And yes, I was skipping pages and reading only what was needed. Will start from page 1

@geek_kushal

Trying to do option 2

[williamconley]

@ William,

Actually I was running above 600 concurrent calls on a cluster with single telephony server, system load was around 50, I thought it could handle that number of calls but forgot that Asterisk has its limits.
I understand the call will be thrown to call menu / ingroup through the campaign and remote agent is just another agent but I fail to understand the concept of external extension completely

And yes, I was skipping pages and reading only what was needed. Will start from page 1

@geek_kushal

Trying to do option 2

It's not an asterisk limit, actually. It's a perl limit. Vicidial uses hi-res timing in perl modules which require (essentially) every tick of the system clock. In a virtualized system OR in an overloaded system, some of those ticks can get missed and at that point just about anything can happen. So if your system load was 50 and you don't have 50 cores, your system was overloaded and ticks were being missed.

When you perform your initial setup to get everythign working, test it with a single call to get it all properly configured (eg: Your cell phone and YOU on an agent phone, so you can hear and observe the entire call to be sure everything works without accepting the word of someone else).

After that works, be sure you have a test call to yourself mixed in every few hundred calls so you can be sure the outbound audio works AND pressing 1 actually transfers you to an agent AND you get two-way clear audio on that agent call. Then as you raise and lower your channel usage you can see where the overload occurs and drop it down a bit under that. Continue to test for a couple days to be comfortable that you set the limit in the right place.

Happy Hunting!

[mubeen]

Thank you William for detailed reply

Have installed another Telephony server, added it to cluster from web and checked in reports (Its showing 2nd telephony server), Allowed in IPTables. I'm not sure how to confirm whether it has been properly integrated or not. I did not restarted any old server after installing new telephony server

When I created a new phone on from web using new telephony server IP, Its not saving in the sip-vicidial.conf file.

[mubeen]

Okay, resolved the issue. Had to update the IP

I have all my servers on cloud, do I have to load balance by configuring and registering phones in different Telephony servers of is there any other way as well?

[geek_kaushal]

I just tested it on a high capacity server built for P1. I put this in extensions.conf

exten => _2213,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _2213,2,AGI(agi-set_variables.agi,)
exten => _2213,3,Set(CALLERID(all)=${phone_number})
exten => _2213,n,AGI(agi-DID_route.agi)
exten => _2213,n,Hangup()

and added 2213 as DID and pointed to the ingroup where closers are logged in.

So, when 1 is pressed, it takes the customer's phone as caller id and dials 2213, which sends the call in the ingroup.

Survey Method: EXTENSION
Survey Survey Xfer Extension: 2213

I dont know if thats perfect, but it works wonderful..

[williamconley]

Okay, resolved the issue. Had to update the IP

I have all my servers on cloud, do I have to load balance by configuring and registering phones in different Telephony servers of is there any other way as well?

load balance is a vague term. do you refer to web server load? mysql server load? dialer load?

web server load has no "built in to vicidial" feature.

mysql server load is managed by adding a replication/report server for all non-dialer functions (reporting, mostly).

dialer load is balanced via phone aliases. In admin->phones there is a top menu with aliases. this is, however, a complex task unless you just use the vicidial manager's manual to guide you. the concept is that each agent registers their soft phone to ALL dialers and the vicidial system determines which to actually use when they log in. as agents log in/out all day long, the system balances automatically.

[elismerkaj]

hi there

my server is
ViciBox v.8.0.1

VERSION: 2.14-858a
BUILD: 220513-0819
© 2022 ViciDial Group


i have configured the survey with call menu and than call menu to extension 772 and added the dialplane this one
exten => _772,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _772,2,Dial(SIP/101&SIP/110&SIP/111&SIP/cc117&SIP/cc118&SIP/cc119&SIP/cc120&SIP/cc121,,tTo)
exten => _772,3,Hangup

everthing works fine but when it rings to the sip it shows eve the client number and i dont want that
i want to hide the number(at least 4, 5 digits of the number)
is there any solution

I just tested it on a high capacity server built for P1. I put this in extensions.conf

exten => _2213,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _2213,2,AGI(agi-set_variables.agi,)
exten => _2213,3,Set(CALLERID(all)=${phone_number})
exten => _2213,n,AGI(agi-DID_route.agi)
exten => _2213,n,Hangup()

and added 2213 as DID and pointed to the ingroup where closers are logged in.

So, when 1 is pressed, it takes the customer's phone as caller id and dials 2213, which sends the call in the ingroup.

Survey Method: EXTENSION
Survey Survey Xfer Extension: 2213

I dont know if thats perfect, but it works wonderful..

[elismerkaj]

hi thre resovled this problem by selecting EXtension on the survye instead of callmenu
so when the client pres 1 it rings to the extesniion 772 exten => _772,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _772,2,Dial(SIP/101&SIP/110&SIP/111&SIP/cc117&SIP/cc118&SIP/cc119&SIP/cc120&SIP/cc121,,tTo)
exten => _772,3,Hangup


and rings to all the sip that i have inserted over there with caller id 0000000

now i have one issue how to extract the reports ???
does anyone have any idea?

hi there

my server is
ViciBox v.8.0.1

VERSION: 2.14-858a
BUILD: 220513-0819
© 2022 ViciDial Group


i have configured the survey with call menu and than call menu to extension 772 and added the dialplane this one
exten => _772,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _772,2,Dial(SIP/101&SIP/110&SIP/111&SIP/cc117&SIP/cc118&SIP/cc119&SIP/cc120&SIP/cc121,,tTo)
exten => _772,3,Hangup

everthing works fine but when it rings to the sip it shows eve the client number and i dont want that
i want to hide the number(at least 4, 5 digits of the number)
is there any solution

I just tested it on a high capacity server built for P1. I put this in extensions.conf

exten => _2213,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _2213,2,AGI(agi-set_variables.agi,)
exten => _2213,3,Set(CALLERID(all)=${phone_number})
exten => _2213,n,AGI(agi-DID_route.agi)
exten => _2213,n,Hangup()

and added 2213 as DID and pointed to the ingroup where closers are logged in.

So, when 1 is pressed, it takes the customer's phone as caller id and dials 2213, which sends the call in the ingroup.

Survey Method: EXTENSION
Survey Survey Xfer Extension: 2213

I dont know if thats perfect, but it works wonderful..

[williamconley]

i have configured the survey with call menu and than call menu to extension 772 and added the dialplane this one
exten => _772,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _772,2,Dial(SIP/101&SIP/110&SIP/111&SIP/cc117&SIP/cc118&SIP/cc119&SIP/cc120&SIP/cc121,,tTo)
exten => _772,3,Hangup

You should have Remote Agents in the campaign. Read the Vicidial Manager's Manual. Remote agents can have phones. Rather than manually adding sip phones to an extension, you change each remote agent (one phone per agent) to ACTIVE or INACTIVE based on whether you want that agent to get calls. There's even a "remote agent interface", so you can have them activate their own phones, if you like.

Better yet ... Logged In agents with Firefox running the agent interface, for full Vicidial functionality.

Have you ... read the Vicidial Manager's Manual?

when it rings to the sip it shows eve the client number and i dont want that
i want to hide the number(at least 4, 5 digits of the number)

This would be a setting in the InGroup: if you were using remote agents you could control the number presented to the agents with OnHook settings. Of course, OnHook would require you to route the calls through an InGroup.

All of which is covered in the Manual for proper setup to avoid having to re-invent Vicidial using your likely previous knowledge of Asterisk. Like an Amish using a horse to pull their "station wagon" because ... it's a wagon, right? lol

now i have one issue how to extract the reports ???

There's an entire menu for Reports. Since you have not stated that specific information is missing from one of them or being sought by you, all I can do is refer you to the reports.

I will say that if you want reports based on who took calls, the Vicidial system has no idea since you are generating calls directly to sip extensions outside Vicidial. Refer to the above remarks about "remote" or "logged in" Vicidial agents to get reports.

Additionally: Remote agents would show as "in call" and you could avoid generating new calls for agents already in calls, thus Vicidial could stop dialing entirely when all your agents are busy. Avoiding dropped calls.