Call barging/listen

[maykelsoft]

hi all,

i've noticed that while doing barging/listening, the agent can hear a tone... how can i make it silent?

Vicidialnow 1.3
VERSION: 2.2.0-236
BUILD: 100413-2328

Thanks!

[mflorell]

Listening should be silent if your system is setup properly.

[maykelsoft]

hi,

is there anything i need to adjust on extensions.conf to achieve the silent?

thanks!

[ronator]

you mean the pling sound. dont know right now how to modify the web page links to do so but you can do it with a phone that is registered at your vicidial server (since in the section ADMIN->PHONE the phones you want to listen FROM has "Admin Monitor" set to 1 ) silent monitorig is done through extension 68600xxx or 78600xxx where

8600xxx is just the session id of the agent you wanna listen to which is connected to the vicidial-conference room of the agent, e.g. 8600051.

If you dial this string on your "admin phone", you enter the specified conf room, signalling to the agent and you can talk (be heard, also by customers!) If you add a 7 to that string, the agent won't here that you entered (no pling), but he/she can hear any sounds that come in from your phone. This is useful, if you want to join the dialog but you have to consider legality (do NOT record and inform your agents, that silent monitoring is done for quality assurance) <- *just a hint*

If u add a 6 to that string, there is no pling and you are muted, so the agent won't ever here, that someone entered. So, what I do is dialing 68600051 to listen uncovered and unheard what the agent is saying to the customer. I don't even have these listen/barge links in my interface anymore o_O Thank you for focusing me on that fact ;D

Be careful when playing around in extensions.conf with these standard extensions !!! make a copy !!!

hope I could help
best wishes

[mflorell]

If you are on 2.0.5 or higher you should be dialing 0 + session ID to blind monitor, not 6.

[maykelsoft]

hi,

will i dial 0 + session id on softphone?

thanks!

[ronator]

yes, that's right, pardon me, dial 0 + session-id ... the old system used 6 + session-id or 7 + session-id. I also got that update from a fresh ViciDial Manager Manual. But with 6 + session-id will also work, but after the call is finished from either the caller or the callee, you will be kicked ... and sometimes it's really interessting what agents say, AFTER they hung up ;-)

a good hint might be, to have a look in the extensions.conf (and maybe extensions-vicidial.conf) to see what extension are defined there. Furthermore, if something doesn't work as you'd expect, open the asterisk CLI with "asterisk -rvvvvvvvvvvvvv" (veeeeery verbose) and watch out for any error message while repeating the failing task ... that often helps, because you should get information, what's going wrong (sometimes an ext is not defined, then it goes to invalid ext (i) or timeout). Keep in mind that it is always helpful to have a console open !

best wishes
Ron Salvatore

[maykelsoft]

Hi ronator,

thanks for the relpy, barging/listening is working, all i need to do is to disable the tone while connecting. the agent hear a tone when someone barge. i just want to eliminate the tone and make it silent.

thanks!

yes, that's right, pardon me, dial 0 + session-id ... the old system used 6 + session-id or 7 + session-id. I also got that update from a fresh ViciDial Manager Manual. But with 6 + session-id will also work, but after the call is finished from either the caller or the callee, you will be kicked ... and sometimes it's really interessting what agents say, AFTER they hung up

a good hint might be, to have a look in the extensions.conf (and maybe extensions-vicidial.conf) to see what extension are defined there. Furthermore, if something doesn't work as you'd expect, open the asterisk CLI with "asterisk -rvvvvvvvvvvvvv" (veeeeery verbose) and watch out for any error message while repeating the failing task ... that often helps, because you should get information, what's going wrong (sometimes an ext is not defined, then it goes to invalid ext (i) or timeout). Keep in mind that it is always helpful to have a console open !

best wishes
Ron Salvatore

[roundqube]

I'm connected to a phone (using hard sip phone) that is also connected as an agent. On my extension I have admin monitor set to 1. I type in 08600053 and click send dtmf but I cannot get any monitoring on an agent (log below):

I cannot click the Listen link via the website because I'm not using a sip soft phone. This link will only work via soft phone?

-- Executing [8500998@default_master:3] AGI("Local/8500998@default_fl-pg01-835f,2", "agi-dtmf.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-dtmf.agi
-- Playing 'silence' (escape_digits=) (sample_offset 0)
-- Playing '0' (escape_digits=) (sample_offset 0)
-- Playing 'silence' (escape_digits=) (sample_offset 0)
-- Playing '8' (escape_digits=) (sample_offset 0)
-- Playing 'silence' (escape_digits=) (sample_offset 0)
-- Playing '6' (escape_digits=) (sample_offset 0)
-- Playing 'silence' (escape_digits=) (sample_offset 0)
-- Playing '0' (escape_digits=) (sample_offset 0)
-- Playing 'silence' (escape_digits=) (sample_offset 0)
-- Playing '0' (escape_digits=) (sample_offset 0)
-- Playing 'silence' (escape_digits=) (sample_offset 0)
-- Playing '0' (escape_digits=) (sample_offset 0)
-- Playing 'silence' (escape_digits=) (sample_offset 0)
-- Playing '5' (escape_digits=) (sample_offset 0)
-- Playing 'silence' (escape_digits=) (sample_offset 0)
-- Playing '3' (escape_digits=) (sample_offset 0)
-- Playing 'silence' (escape_digits=) (sample_offset 0)
-- Playing 'silence' (escape_digits=) (sample_offset 0)
-- Playing 'silence' (escape_digits=) (sample_offset 0)
-- Playing 'silence' (escape_digits=) (sample_offset 0)
== Manager 'sendcron' logged off from 127.0.0.1
-- Playing 'silence' (escape_digits=) (sample_offset 0)
-- Playing 'silence' (escape_digits=) (sample_offset 0)
-- Playing 'silence' (escape_digits=) (sample_offset 0)
-- Playing 'silence' (escape_digits=) (sample_offset 0)
-- Playing 'silence' (escape_digits=) (sample_offset 0)
-- AGI Script agi-dtmf.agi completed, returning 0
-- Executing [8500998@default_master:4] Hangup("Local/8500998@default_fl-pg01-835f,2", "") in new stack
== Spawn extension (default_master, 8500998, 4) exited non-zero on 'Local/8500998@default_fl-pg01-835f,2'
-- Executing [h@default_master:1] DeadAGI("Local/8500998@default_fl-pg01-835f,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
== Spawn extension (default_master, 78600052, 1) exited non-zero on 'Local/8500998@default_fl-pg01-835f,1'
-- Executing [h@default_master:1] DeadAGI("Local/8500998@default_fl-pg01-835f,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0

[williamconley]

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

Similar to This:
Vicibox X.X from .iso | Vicidial X.X.X Build XXXX | Asterisk X.X.X | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation

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)
_______

have you read the manual?

have you tried DIALING 08600053 on the sip phone? (which cannot be busy when the call is made, since you are MAKING a call ...)

[roundqube]

VERSION: 2.0.5-173
BUILD: 90320-0424
Installed via SCRATCH INSTALL method.
Asterisk version 1.4.22.1
Single Server
Sangoma UT50 (USB Synch Tool)

I have read the manual and when I dial 08600053 for example, the SIP phone is tied up in a session with Vicidial so I cannot dial (no DTMF is sent). So I tried to send DTMF via the web client and that produced the logs I posted earlier.

[williamconley]

you can only use the phone for one thing at a time. you cannot use it in a conference AND to monitor another conference at the same time (unless it has two lines in it).

[roundqube]

So I can only monitor from a phone directly attached to the Vicidial / Asterisk SIP?

Because right now we're using corporate hardphones (Polycom's) registered to other Asterisk servers (exch21 for example - extension 21123). I have this phone entry in the Vicidial phones list but I'm not directly attached to Vicidial.

[williamconley]

as long as you can dial that number on the vicidial server, you're good. you do not have to be doing this directly from a phone on the system. you CAN do it from a phone THROUGH another server, as long as you route the call properly.

if you call through another asterisk server and set THAT asterisk server to dial THROUGH vicidial if 08600XXX is dialed, it can still land in the right spot.

[roundqube]

What would the exten line look like for that?

exten => 08600XXX ... ?

[roundqube]

Also, does this only work with SIP? Alot of my call centers are on IAX2 registered extensions.

[williamconley]

that's almost funny.

iax = inter-asterisk-exchange

this is what iax is designed for so YES, it will most certainly work with IAX (even in some instances where it would be difficult or impossible for SIP)

also, you will note that vicidial uses iax for call transfers between servers in cluster mode.

it will be set up as a standard trunk between asterisk servers. there is a lot of documentation for how to call one asterisk server from another.

[roundqube]

I'm testing with one of my servers that I'm registered to and calling the Vicidial server. For now I've manually entered the conference room I want to monitor for testing.

My registration server:
exten => 08600052,1,Dial(SIP/08600052@dlr11fp01fl)

Vicidial/Dialer:
exten => _08600052,1,Dial(${TRUNKblind}/6${EXTEN:1},55,To)
exten => _08600XXX,1,Dial(${TRUNKblind}/6${EXTEN:1},55,To)

Error Log:
[Jul 29 14:21:50] NOTICE[2849]: chan_sip.c:14847 handle_request_invite: Call from 'reg21pg01fl' to extension '08600052' rejected because extension not found.

Yes, I've reloaded the dialplan on both servers. I see the call go from my server to the dialer and thats where the error is produced.

[williamconley]

on the inbound "account", the "context=default" must be there so it can "find" the extension in question.

[roundqube]

Added the extension to the context and now I get this:

-- Executing [08600051@default:1] Dial("SIP/reg21pg01fl-b7a404d8", "IAX2/ASTblind:test@127.0.0.1:41569/68600051|55|To") in new stack
-- Called ASTblind:test@127.0.0.1:41569/68600051
[Jul 30 11:26:29] NOTICE[2859]: chan_iax2.c:9317 socket_process: Host 127.0.0.1 failed to authenticate as ASTblind
[Jul 30 11:26:31] WARNING[2852]: chan_iax2.c:9079 socket_process: Call rejected by 127.0.0.1: No authority found
-- Hungup 'IAX2/127.0.0.1:41569-3041'
== Everyone is busy/congested at this time (1:0/0/1)
== Auto fallthrough, channel 'SIP/reg21pg01fl-b7a404d8' status is 'CHANUNAVAIL'

[roundqube]

I'm seeing no iax2 peers:

*CLI> iax2 show peers
Name/Username Host Mask Port Status
0 iax2 peers [0 online, 0 offline, 0 unmonitored]

[roundqube]

Nevermind, I'm getting closer. Disregard my last two posts.

[williamconley]

hey, works in progress ... the next guy will see all this and work his way through as well ... keep it going ...

[roundqube]

I must have overwritten my iax.conf because we were doing some customizations (recall that we are using Vicidial call centers in VMWare) with a hardware Asterisk server and hosting DB's within their own VM's so we can move them around from one server to another.

Anyhow, I updated the iax.conf and now show iax2 peers shows the following:
*CLI> iax2 show peers
Name/Username Host Mask Port Status
ASTblind 127.0.0.1 (D) 255.255.255.255 52085 OK (1 ms)
ASTloop 127.0.0.1 (D) 255.255.255.255 57192 OK (1 ms)
2 iax2 peers [2 online, 0 offline, 0 unmonitored]

Then I hard coded my extensions.conf for now with:
exten => 68600052,1,Dial(${TRUNKblind}/6${EXTEN:1},55,To)

Still can't get it to listen. Get the following error:
-- Executing [08600051@default:1] Dial("SIP/reg21pg01fl-b7a06700", "IAX2/ASTblind:test@127.0.0.1:41569/68600051|55|To") in new stack
-- Called ASTblind:test@127.0.0.1:41569/68600051
[Jul 30 12:03:18] NOTICE[2855]: chan_iax2.c:9329 socket_process: Rejected connect attempt from 127.0.0.1, request '68600051@default' does not exist
[Jul 30 12:03:18] WARNING[2856]: chan_iax2.c:9079 socket_process: Call rejected by 127.0.0.1: No such context/extension
-- Hungup 'IAX2/127.0.0.1:41569-1895'
== Everyone is busy/congested at this time (1:0/0/1)
== Auto fallthrough, channel 'SIP/reg21pg01fl-b7a06700' status is 'CHANUNAVAIL'

Assuming that I need to just have the _X86000XX in a default context? How do I tell it to use another context? Because about our VM's, we have several call centers so we changed the default context to [default_fl-xx01] where xx is their call center code, then within that context, we set a few variables like conf file etc.. and point to [default_master]. There is no longer [default] context.

[default_fl-xx01]
exten => _X.,1,Set(__vPHONECENTER=fl-xx01)
exten => _X.,n,Set(__vFASTAGI_IP=10.1.1.1)
exten => _X.,n,Set(__vCONF_FILE=/etc/astguiclient_fl-xx01.conf)
exten => _X.,n,Goto(default_master,${EXTEN},1)

[williamconley]

why are you calling the same machine? (127.0.0.1)

i thought you were trying to call a different machine with that conference in it

[roundqube]

In this case, I have one call center on the actual 127.0.0.1 (sorry for the confusion). This call center was on the dialer long before we came up with the VMWare idea.

And this call center's vicidial conferences are numbered 8600050-8600100, so I am just trying to listen in on 8600051 or 8600052.

[default_fl-pg01]
exten => _X.,1,Set(__vPHONECENTER=fl-pg01)
exten => _X.,n,Set(__vFASTAGI_IP=127.0.0.1)
exten => _X.,n,Set(__vCONF_FILE=/etc/astguiclient.conf)
exten => _X.,n,Goto(default_master,${EXTEN},1)

[williamconley]

if you're already on the machine, you don't need anything special to dial it. just dial it from a phone that can dial in the default context on that box from a phone registered to that box.

[roundqube]

Problem is that I am not registered on that box. It's our dialer but the phones are registered to other Asterisk servers.

So I am dialing 08600052 from my phone registered to Server A which is doing an Asterisk-Asterisk call to Server B. But it's going into the default context (of which we do not have because of our customizations).

[williamconley]

according to what you were posting before ... your phone was registering to box a, then box a was calling ... itself. not box b. that's what 127.0.0.1 is: "self"

[roundqube]

I'm on a phone on box A and box A has an extension in the dial plan so when I call 08600052, it does a sip call via Dial(SIP/08600052@boxB) so the call reaches box B.

At this point, it hits the dialplan on Box B, and thats when I see that error of loopback because essential BoxB sees the call and transfers to the astblind on loopback right?

I'm abviously confused here on what to do. Need some assistance. Tried searching these forums and google all day.

[williamconley]

!

Sorry, I never use the 0, i always use "6" to enter silently which does not use the loopback. so i didn't recognize it.

try 6 or try to figure out why your loopback is not working.

begin with just dialing the conference room (without the 0 or the 6) to get it working. it will make a tone when you enter, but you'll at least be "working" and then you can work out the details afterwords.

[roundqube]

Got that fixed and now we are able to listen in. Haven't tested barging (thats next) but here's my other issue and it relates to the Vicidial web interface.

When I try to go to that "special" campaign summary screen for listen/barge options via the VDAD campaign summery under Reports, it shows me no campaigns although we have a ton active.

Any ideas?

[williamconley]

maybe you should open a new thread for that and list all the related information.