Inbound in Vicidial and some other questions

[pedro]

Hello,
first of all I'd like to thank the developers of Vicidial for they work, they've made a fine piece of software.

I'm trying to setup a blended callcenter with 35 seats, I've been trying out Vicidial and it has been doing very well so far, but I still don't get some things.

When I want to do inbound, where or how can I define what calls (from what number, I suppose) will be distributed to which in-group?

The inbound agents, when in blended mode, are able to call just the numbers contained in the CLOSER campaign, or is there some way to have them call some other campaigns?

Overall, is there a way to have an agent logged in more campaigns?

The Busy calls and NA calls etc... Are these lost or are they called again in some time?

Is there any way how to listen to the agents calls and maybe take over the call? And is there a way how to trigger call recording remotely?

[mflorell]

Hello,
first of all I'd like to thank the developers of Vicidial for they work, they've made a fine piece of software.

Thank you very much!

When I want to do inbound, where or how can I define what calls (from what number, I suppose) will be distributed to which in-group?

Here is a rough draft of one of the tutorials from the upcoming VICIDIAL Manager Manual:

B. Create an inbound-group, point an external number at it and take calls

This Tutorial requires either a little systems administration knowledge or some help from your Systems Administrator. Also required is a T1 PRI, SIP or IAX line with inbound and callerID delivery. You have just tested an outbound campaign and now you want to have your company's 800 number point to VICIDIAL so that you can have Agents answering inbound phone calls with callerID information.

1. Find out from your carrier(phone line provider) what they will be sending your calls to you as. Many carriers will only send you the last 4 digits of the number that has been called. You need this information to give your System Administrator so that they can setup your dialplan properly. For this tutorial we will use 800-227-7655 as the 800 number and our provider will send us all 10 digits for these calls.
2. To take inbound numbers in VICIDIAL you will need to first setup an inbound group(or in-group) in the VICIDIAL admin.php page.
3. Go to the IN-GROUPS section of the VICIDIAL admin website and click on the ADD NEW IN-GROUP.
4. For this tutorial we will use the following values for the fields on the NEW IN-GROUP form:
group ID: SALESLINE
group name: Primary Sales Line
group color: red
active: Y
web form: <we leave this blank> If you have a CRM, put that address here.
voicemail: <we leave this blank>
next agent call: oldest_call_finish
fronter display: Y
script: NONE
get call launch: NONE
5. We will click submit to create the in-group and then go to the CAMPAIGNS section and the ADD NEW CAMPAIGN link.
6. For this tutorial we will use the following for the fields on the NEW CAMPAIGN form:
campaign ID: CLOSER
campaign name: Closer and inbound campaign
active: Y
- we will leave park extension, park filename and web form blank for this tutorial -
allow closers: Y
hopper level: 5
auto dial level: 1
next agent call: oldest call finish
local call time: 24hours
- we will leave voicemail, script and get call launch blank for this tutorial -
7. Click submit and scroll down to the "Allowed Inbound Group" section and check the box for SALESLINE then click the SUBMIT button to commit the changes.
8. Now we need to setup the dialplan to send the calls coming into this number to go to VICIDIAL. The following lines need to be added to your dialplan(the extensions.conf file) by your System Administrator:
exten => 8002277655,1,Ringing ; call ringing
exten => 8002277655,2,Wait(1) ; Wait 1 second for CID delivery from PRI
exten => 8002277655,3,Answer ; Answer the line
exten => 8002277655,4,AGI(agi-VDADcloser_inboundCID.agi,SALESLINE-----8002277655-----Closer---------------999-----1)
exten => 8002277655,5,Hangup
Then you need to reload Asterisk for this to go into effect.
9. Now that you have your in-group setup you should have your agent log into the vicidial client app: http://server-address/agc/vicidial.php
10. On the vicidial.php login screen you will need to enter in the phone login and phone password for the phone that the agent will be using(you should have a list of the phone login and password for each of your stations from your system administrator) then click submit.
11. You will then need to enter in your username(7777) and password(1234) and select your campaign(CLOSER) then click submit.
12. You should now be logged in as agent 7777 and your phone should ring to place you into the VICIDIAL session
13. Because you logged into a CLOSER campaign, you should now see a green screen with the listing of all in-groups. Click on the SALESLINE group in the left side column and then click SUBMIT at the bottom of the screen.
14. Since the campaign is set to a dial level of one (1) you are in auto dial mode and will need to click the RESUME button to make yourself available to take inbound calls.
15. When a call comes into the 800 number, the customer's callerID and possibly name or location should appear in the vicidial agent screen fields as well as the status bar at the top changing to the color red(which we set in the in-group screen earlier).
16. When you are done with the call click on the HANGUP CUSTOMER button to get to the Disposition screen and then pick a disposition to terminate the call as.
17. After placing a live call you can now click on the LOGOUT link in the top right corner of your VICIDIAL client page to properly log out of the system. At this time your phone connection should be hungup.

The inbound agents, when in blended mode, are able to call just the numbers contained in the CLOSER campaign, or is there some way to have them call some other campaigns?

An agent in the CLOSER campaign can only receive outbound calls that are in the lists assigned to the CLOSER campaign.

Overall, is there a way to have an agent logged in more campaigns?

Currently there is no way to do this. This functionality greatly complicates the dialing engine because of multi-campaign Agents accounting for a fraction of an active agent in the dialing process.

The Busy calls and NA calls etc... Are these lost or are they called again in some time?

In the most recent code snapshot there is a function to automatically re-call leads at a specified interval without resetting the list and without defining them to be called. We only recommend this for Busys since the customer never knows that you attempted a call when it returns busy, not true of NAs. NAs can be dialed again by resetting the list and selecting NA as a dial status for the campaign.

Is there any way how to listen to the agents calls and maybe take over the call? And is there a way how to trigger call recording remotely?

You can listen in quietly on a call by dialing 6 + session ID(68600051) and you can enter the conversation directly by dialing the sessionID(8600051). As for triggering recording remotely, the record_conf_1_hour.php script can do this(currently it will only record the current call not for 1 hour).

[pedro]

Hello,
first of all I'd like to thank the developers of Vicidial for they work, they've made a fine piece of software.

Thank you very much!

You're very welcome, we're the ones who are thankful.

When I want to do inbound, where or how can I define what calls (from what number, I suppose) will be distributed to which in-group?

Here is a rough draft of one of the tutorials from the upcoming VICIDIAL Manager Manual:

Thanks, we will try it. I believe we should be able to create a simple IVR through these, right?

Overall, is there a way to have an agent logged in more campaigns?

Currently there is no way to do this. This functionality greatly complicates the dialing engine because of multi-campaign Agents accounting for a fraction of an active agent in the dialing process.

Ok, we'll learn to live with that. Multiple lead lists can partially replace this feature for us.

I've got another question, according to the Vicidial page this feature should be available, unfortunately I cannot find a way how to enable it. Can you please point me in the right direction?

Ability to autodial campaigns to start with a simple IVR then direct to agent

[mflorell]

Take a look at these two scripts, they both do some IVR functionality before sending a call onto a VICIDIAL Agent:
agi-VDADtransferSURVEY.agi
agi-VDADautoREMINDERxfer.agi

[christivi]

hi to all
im facing an issue. in fact im trying to create an inbound campaign. i followed the manager manual. i created the campaign as demanded. when i click on detail view to change 'allow inbound and blended' from N to Y nothing happens. it doesn't change and 'allowed inbound group' is no where to be seen.
my dial method is RATIO as recommended.
my config:

Campaign ID: TEST_IN
Campaign Name: Closer and inbound campaign
Active: yes
Allow Closers: Y
Hopper level: 5
Auto Dial level: 1
Next Agent Call: oldest call finish
Local Call Time: 24hours

i noticed that when i make change in the detail view even if vicidial confirm that the campaign have been modified nothing changes.

[williamconley]

always post your installation method and your vicidial version with build as an absolute minimum (this is a requirement for getting the free help )

if this is a fresh installation and you have never had the machine "live", you may want to check your installation log in /var/log (depending on your installation version) and/or REINSTALL while watching for errors.

[christivi]

thanks for replying
im using ViciBox Redux v.3.0.1 which i installed following the walk through installation.
my installation is a fresh new installation.
so what you suggest is reinstalling vicidial ? at which stage of the whole process ? or you suggest i reinstall everything ?

[williamconley]

always post your installation method and your vicidial version with build as an absolute minimum (this is a requirement for getting the free help )

Start over. Watch for errors. Do it a couple times (Free, right?). Perhaps consider using a different version, just in case (if you used SVN and OS Update, try 2.2.1 or don't update the OS, in case there's a little bug in there somewhere messing with you!). IF you actually find a little bug (if it resolutely fails to allow modifications to the database, but only if you update the OS, for instance), please post a note on the Issue Tracker

[ktsis]

I'm using GoAutoDial CE 2.1, Asterisk 1.4.39.1-vici and a Digium TDM410P with 4 FXO ports without EC hardware.
Each port is supposed to both receive and make calls via PSTN.

GoAutoDial and the FXO card install successfull. I am able to login as an agent and my softphone rings
when the agent logs in.

I followed the step you stated above to configure the FXO card to take inbound call. but each time call
from mobile phone i get this error log:

[Apr 26 09:44:34] -- Starting simple switch on 'DAHDI/1-1'
[Apr 26 09:44:37] NOTICE[4721]: chan_dahdi.c:8053 ss_thread: Got event 18 (Ring Begin)...
[Apr 26 09:44:38] NOTICE[4721]: chan_dahdi.c:8053 ss_thread: Got event 2 (Ring/Answered)...
[Apr 26 09:44:38] == Starting DAHDI/1-1 at from-pstn,s,1 failed so falling back to exten 's'
[Apr 26 09:44:38] == Starting DAHDI/1-1 at from-pstn,s,1 still failed so falling back to context 'default'
[Apr 26 09:44:38] -- Sent into invalid extension 's' in context 'default' on DAHDI/1-1
[Apr 26 09:44:38] -- Executing [i@default:1] Playback("DAHDI/1-1", "invalid") in new stack
[Apr 26 09:44:38] -- <DAHDI/1-1> Playing 'invalid' (language 'en')
[Apr 26 09:44:40] == Refreshing DNS lookups.
[Apr 26 09:44:42] == Auto fallthrough, channel 'DAHDI/1-1' status is 'UNKNOWN'
[Apr 26 09:44:42] -- Executing [h@default:1] DeadAGI("DAHDI/1-1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Apr 26 09:44:42] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Apr 26 09:44:42] -- Hungup 'DAHDI/1-1'
[Apr 26 09:45:02] == Parsing '/etc/asterisk/manager.conf': [Apr 26 09:45:02]

[striker]

you are using from-pstn as inbound context
option 1
change from-pstn to trunkinbound in your chan_dahdi.conf or dahdi-channels.conf
and in extensions.conf add the below dialplan under trunkinbound

[trunkinbound]
exten => s,1,AGI(agi-DID_route.agi)

option2
create a new context from-pstn in extensions.conf and add the below dialplan
[from-pstn]
exten => s,1,AGI(agi-DID_route.agi)
exten => h,1,DeadAGI(agi://127.0.0.1:4577/call_log--HVcauses ... EBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME})

[ktsis]

Thanks alot for the reply. I followed the second option. When i call the line connected to the first port on the FXO card, i got the the following log:


[Apr 27 22:26:39] == Refreshing DNS lookups.
[Apr 27 22:26:46] -- Starting simple switch on 'DAHDI/1-1'
[Apr 27 22:26:49] NOTICE[9139]: chan_dahdi.c:8053 ss_thread: Got event 18 (Ring Begin)...
[Apr 27 22:26:50] ERROR[9139]: callerid.c:564 callerid_feed: No start bit found in fsk data.
[Apr 27 22:26:50] WARNING[9139]: chan_dahdi.c:8100 ss_thread: Failed to decode CallerID on channel 'DAHDI/1-1'
[Apr 27 22:26:50] WARNING[9139]: chan_dahdi.c:8205 ss_thread: CallerID returned with error on channel 'DAHDI/1-1'
[Apr 27 22:26:50] -- Executing [s@from-pstn:1] AGI("DAHDI/1-1", "agi-DID_route.agi") in new stack
[Apr 27 22:26:50] -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-DID_route.agi
[Apr 27 22:26:50] ERROR[9139]: utils.c:967 ast_carefulwrite: write() returned error: Broken pipe
[Apr 27 22:26:50] -- AGI Script agi-DID_route.agi completed, returning 0
[Apr 27 22:26:50] -- Executing [9998811112@default:1] Wait("DAHDI/1-1", "2") in new stack
[Apr 27 22:26:52] -- Executing [9998811112@default:2] Answer("DAHDI/1-1", "") in new stack
[Apr 27 22:26:52] -- Executing [9998811112@default:3] Playback("DAHDI/1-1", "ss-noservice") in new stack
[Apr 27 22:26:52] -- <DAHDI/1-1> Playing 'ss-noservice' (language 'en')
[Apr 27 22:26:52] WARNING[9139]: file.c:1297 waitstream_core: Unexpected control subclass '2'
[Apr 27 22:26:55] == Refreshing DNS lookups.
[Apr 27 22:26:57] -- Executing [9998811112@default:4] Playback("DAHDI/1-1", "vm-goodbye") in new stack
[Apr 27 22:26:57] -- <DAHDI/1-1> Playing 'vm-goodbye' (language 'en')
[Apr 27 22:26:58] -- Executing [9998811112@default:5] Hangup("DAHDI/1-1", "") in new stack
[Apr 27 22:26:58] == Spawn extension (default, 9998811112, 5) exited non-zero on 'DAHDI/1-1'
[Apr 27 22:26:58] -- Executing [h@default:1] DeadAGI("DAHDI/1-1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
[Apr 27 22:26:58] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Apr 27 22:26:58] -- Hungup 'DAHDI/1-1'

I believe i am closing to getting. Please how to i fix the error in the above log.

Thanks

[williamconley]

your system is sending the calls to the "default" DID. modify the route that the default DID takes to an ingroup.

[ktsis]

Please how to i do that?

[williamconley]

under "Inbound" (Left menu) there is an option called "Show DIDs". There you will find "Default DID".

Change the default DID to point somewhere other than the extension it is presently pointed to (which is a recording that says this number is not in service, which is working perfectly, by the way, according to your previous post).

If you don't know how ... That makes you a Newbie! Here are your Newbie Suggestions! (Pay particular attention to the one about the Vicidial Manager's Manual! Read it from start to finish, do not skip any of the lessons, do them all in order from the beginning!)

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

Happy Hunting!

[ktsis]

Thanks.

I have gone through Manager Manual(Page 22). here is what i did:
I login into vicidial admin, click on show DIDs, and edited the default.

1. I changed extension field from 9998811112 to 08039093941 and left every other setting as default.

2. I open up my extension.conf and added the following lines under default context

exten => 08039093941,1,Ringing
exten => 08039093941,2,Wait(1)
exten => 08039093941,3,AGI(agi://127.0.0.1:4577/call_log--fullCID--${EXTEN}-----${CALLERID}-----${CALLERIDNUM}-----${CALLERIDNAME})
exten => 08039093941,4,Answer
exten => 08039093941,5,Dial(sip/8001,30,To)
exten => 08039093941,6,Voicemail,u2000

3. Reload asterisk

4. Loggin as an agent with phone no 8001, my softphone rings and click on resume botton

4. I call 08039093941 from mobile phone.

My softphone rings but the agent web page does not show any call coming in.

Here is the current log:

[Apr 28 01:52:09] == Refreshing DNS lookups.
[Apr 28 01:52:13] -- Starting simple switch on 'DAHDI/1-1'
[Apr 28 01:52:15] NOTICE[20972]: chan_dahdi.c:8053 ss_thread: Got event 18 (Ring Begin)...
[Apr 28 01:52:17] NOTICE[20972]: chan_dahdi.c:8053 ss_thread: Got event 2 (Ring/Answered)...
[Apr 28 01:52:17] -- Executing [s@from-pstn:1] AGI("DAHDI/1-1", "agi-DID_route.agi") in new stack
[Apr 28 01:52:17] -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-DID_route.agi
[Apr 28 01:52:17] ERROR[20972]: utils.c:967 ast_carefulwrite: write() returned error: Broken pipe
[Apr 28 01:52:17] -- AGI Script agi-DID_route.agi completed, returning 0
[Apr 28 01:52:17] -- Executing [08039093941@default:1] Ringing("DAHDI/1-1", "") in new stack
[Apr 28 01:52:17] -- Executing [08039093941@default:2] Wait("DAHDI/1-1", "1") in new stack
[Apr 28 01:52:18] -- Executing [08039093941@default:3] AGI("DAHDI/1-1", "agi://127.0.0.1:4577/call_log--fullCID--08039093941---------------") in new stack
[Apr 28 01:52:18] -- AGI Script agi://127.0.0.1:4577/call_log--fullCID- ... ---------- completed, returning 0
[Apr 28 01:52:18] -- Executing [08039093941@default:4] Answer("DAHDI/1-1", "") in new stack
[Apr 28 01:52:18] -- Executing [08039093941@default:5] Dial("DAHDI/1-1", "sip/8001|30|To") in new stack
[Apr 28 01:52:18] -- Called 8001
[Apr 28 01:52:18] -- SIP/8001-00000004 is ringing
[Apr 28 01:52:43] -- Got SIP response 603 "Decline" back from 192.168.1.200
[Apr 28 01:52:43] -- SIP/8001-00000004 is busy
[Apr 28 01:52:43] == Everyone is busy/congested at this time (1:1/0/0)
[Apr 28 01:52:43] -- Executing [08039093941@default:6] VoiceMail("DAHDI/1-1", "u2000") in new stack
[Apr 28 01:52:43] WARNING[20972]: app_voicemail.c:4203 leave_voicemail: No entry in voicemail config file for '2000'
[Apr 28 01:52:43] == Auto fallthrough, channel 'DAHDI/1-1' status is 'BUSY'
[Apr 28 01:52:53] -- Executing [h@default:1] DeadAGI("DAHDI/1-1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----21-----BUSY----------") in new stack
[Apr 28 01:52:53] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Apr 28 01:52:53] -- Hungup 'DAHDI/1-1'

How do i make the call to show on agent web page?

[striker]

you didnt follow the admin manual properly

totally doing in wrong manner.

1. create a context and dialplan as below

[from-pstn]
exten => s,1,AGI(agi-DID_route.agi)
exten => h,1,DeadAGI(agi://127.0.0.1:4577/call_log--HVcauses ... EBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME})

2. go to Inbound-create a new ingroup for eg: FXOinbound
3. go to inbound-DID-click default and modify as below
--> DID route = Ingroup
--> ingroupid = select FXOinbound from dropdown.
--> submit
4. go to campaign--modify-detailview screen
Allow Inbound and Blended:Y
dialmethod = ratio or inboundman
submit
again go to last line of the same camapaign
Allowed Inbound Groups: select the check box of agetndirect adn FXOinbound
submit

5. go to user -- select the agent
at agent interface option select = Agent Choose Ingroups=1

6. now login as agent and while logging in it will promot to select ingroups , select agentdirect and FXO inbound .
now press resume to take calls

** as you are using FXO line so you wont receive the Inbound DID no, so calls from all the 4ports will land to default DID

[ktsis]

Thanks Striker. I followed you steps. When i call with a mobile line, i get a voice message that say i should holdon and i will be transfered to next available agent. The call as not appeared on the agent interface. I also notice that when i drop the call on the mobile phone i use to make the call, the call does not stop.

Here is the log:

== Manager 'sendcron' logged off from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing [8600051@default:1] MeetMe("SIP/8001-00000000", "8600051|F") in new stack
== Parsing '/etc/asterisk/meetme.conf': Found
== Parsing '/etc/asterisk/meetme-vicidial.conf': Found
-- Created MeetMe conference 1023 for conference '8600051'
-- <SIP/8001-00000000> Playing 'conf-onlyperson' (language 'en')
== Manager 'sendcron' logged off from 127.0.0.1
-- Starting simple switch on 'DAHDI/1-1'
[Apr 28 04:53:49] NOTICE[31110]: chan_dahdi.c:8053 ss_thread: Got event 18 (Ring Begin)...
[Apr 28 04:53:51] NOTICE[31110]: chan_dahdi.c:8053 ss_thread: Got event 2 (Ring/Answered)...
-- Executing [s@from-pstn:1] AGI("DAHDI/1-1", "agi-DID_route.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-DID_route.agi
[Apr 28 04:53:51] ERROR[31110]: utils.c:967 ast_carefulwrite: write() returned error: Broken pipe
-- AGI Script agi-DID_route.agi completed, returning 0
-- Executing [99909*1***DID@default:1] Answer("DAHDI/1-1", "") in new stack
-- Executing [99909*1***DID@default:2] AGI("DAHDI/1-1", "agi-VDAD_ALL_inbound.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_inbound.agi
-- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
-- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Apr 28 04:53:53] WARNING[31110]: res_musiconhold.c:692 get_mohbyname: Music on Hold class 'default' not found
[Apr 28 04:53:53] WARNING[31110]: res_musiconhold.c:692 get_mohbyname: Music on Hold class 'default' not found
-- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
-- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
-- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
-- Playing 'generic_hold' (escape_digits=) (sample_offset 0)


So eager to get it working!

Thanks

[williamconley]

show your DID and Ingroup settings (not all of them, just those that are pertinent ... if you leave any out we'll help you along).

[ktsis]

DID:
DID Extension: default
Active: Y
DID Route: AGENT
Extension:08039093941
Extension Context:default
Phone Extension:8001
Server IP:192.168.1.2
User Route Settings In-Group: FXOinbound-fxoinbound
In-Group ID: FXOinbound-fxoinbound

In-Group:
Group ID: FXOinbound
Group Name:fxoinbound
Group Colour: Yellow
Active:Y
Netx Agent Call:oldest_call_finsh

Thanks

[williamconley]

set the DID route to ingroup.

then (farther down in the did settings) be sure to specify WHICH ingroup.

and use the vicidial manager's manual to configure the DID and Ingroup properly. Step by step. That's what the Vicidial Manager's Manual is for. Really.

[ktsis]

BIG THANKS to striker and williamconley for you assistance. I can now receive call via the agent interface....Very Cool Stuff.

But i want to clarify it i get the right concept.

When call hits the fxo card, the call is sent to DID created. and DID routes the call to the in-group if it is chosen as DID ROUTE. The Campaign(inboundfxo) that is attched to the in-group can access the call. Also agents assigned to the campaign(inboundfxo) and the in-group can access the call.

I just need a general overview of how DID, in-group, campaign, agents, etc work together if one is using fxo card, sip trunk, or voip gateway devices(GOIP).

Thanks alot.

[williamconley]

you'll need to look at assigning a telephone number (DID) to each port of the FXO card. At present, your system is just sending them to the "catch all" of the vicidial DID system which then routes the call (ie: the "default" DID). Setting a specific 10 digit DID for each port of the FXO card will be managed in the configuration files for the hardware in question. It is also a good idea to change the context on the inbound calls to "trunkinbound" instead of "from-pstn" for a more standard configuration.

call route:

1) convince the call to "ring" somewhere in the asterisk server. This can be done in one of three ways ... if the call is SIP, your provider can be informed of the IP of your server directly (often with a web interface) or you can "register" which stores your IP in a dynamic table. But for PSTN/T1/E1 hardware, the call actually is sent directly by the phone company through the line to the hardware. In any of these cases the first requirement is met: the call arrives at your Vicidial server. Now you must convince VICIDIAL to manage the call.

2) Vicidial managing the call is activated by sending the call to "trunkinbound". This context is stored in the "extensions.conf" file and its only purpose is to pass the call to the DID agi script. This script represents "vicidial taking over the call". So now you need vicidial to "manage" the call.

3) Vicidial begins to manage the call in the "DIDs" menu (available via "Inbound -> Show DIDs"). At this point, control of the call is managed by the definition of the DID and the path is determined by how you define that DID. From there, the call can go anywhere within Vicidial based on the definition of the DID.

[ktsis]

Hmmm .

I understand better. but you didn't mention anything about campaign and In-group. And how they relate to DID.

Thanks

[williamconley]

Ingroup is a route you can send a DID to (among many others, but the system is primarily designed around this concept). Campaigns are what agents log in to, and after they log in to a campaign they can also accept calls from Ingroups. But that's what the manual is for.

[ktsis]

What your are doing for me is giving me an overview of the whole process. It makes me appreciate the manual better and a guide when troubleshooting.

I am working on the outbound calls now. just like you did for me incase of the inbound, can you give me an overview of the outbound call flow with respect to using PSTN and SIP Providers.

I appreciate. Thanks

[williamconley]

Depends on your usage. For most people, the agent logs in (each agent, however many there are) and each campaign will "calculate" how many agents are available for calls right now but not presently In One. Then the campaign will generate outbound calls without actually assigning them to an agent ... just the correct number of calls ringing based on the number of agents and the campaign settings.

Any time an agent pauses or logs out or accepts an inbound call or is given a call from the outbound campaign ... that agent will then no longer be "READY", this will affect the number of outbound calls being dialed (which would then be "ringing" calls).

Any time a call is terminated without being answered, Vicidial will mark the call (based on the result) and generate another call to replace it.

Any time a call is answered, it will be routed to an agent. Which agent is determined by the campaign settings.

Upon arrival at the agent, there will be a "bing" telling the agent a call has arrived and then the data will populate the agent screen. Remember that this is a computer, so this all happens within a second or two from the answer.

[ktsis]

Now i am working on the Outbound. I added the following lines to my extension.conf in the default context

exten => _9XXXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _9XXXXXXXXXXX,2,Dial(${TRUNK}/${EXTEN:1},,tTor)
exten => _9XXXXXXXXXXX,3,Hangup

Save and reload asterisk

when i dial a number like 080xxxxxxxx or 070xxxxxxxx or any number from the agent interface, it says this is not a vaild extension.

I check my log and this is what i have:

[May 1 06:22:25] == Parsing '/etc/asterisk/manager.conf': [May 1 06:22:25] Found
[May 1 06:22:25] == Manager 'sendcron' logged on from 127.0.0.1
[May 1 06:22:31] > Channel SIP/8001-00000001 was answered.
[May 1 06:22:31] -- Executing [8600051@default:1] MeetMe("SIP/8001-00000001", "8600051|F") in new stack
[May 1 06:22:31] == Parsing '/etc/asterisk/meetme.conf': [May 1 06:22:31] Found
[May 1 06:22:31] == Parsing '/etc/asterisk/meetme-vicidial.conf': [May 1 06:22:31] Found
[May 1 06:22:31] -- Created MeetMe conference 1023 for conference '8600051'
[May 1 06:22:31] -- <SIP/8001-00000001> Playing 'conf-onlyperson' (language 'en')
[May 1 06:22:33] == Manager 'sendcron' logged off from 127.0.0.1
[May 1 06:22:49] == Parsing '/etc/asterisk/manager.conf': [May 1 06:22:49] Found
[May 1 06:22:49] == Manager 'sendcron' logged on from 127.0.0.1
[May 1 06:22:49] -- Executing [8600051@default:1] MeetMe("Local/8600051@default-b10d,2", "8600051|F") in new stack
[May 1 06:22:49] > Channel Local/8600051@default-b10d,1 was answered.
[May 1 06:22:49] == Starting Local/8600051@default-b10d,1 at default,08060700544,1 failed so falling back to exten 's'
[May 1 06:22:49] == Starting Local/8600051@default-b10d,1 at default,s,1 still failed so falling back to context 'default'
[May 1 06:22:49] -- Sent into invalid extension 's' in context 'default' on Local/8600051@default-b10d,1
[May 1 06:22:49] -- Executing [i@default:1] Playback("Local/8600051@default-b10d,1", "invalid") in new stack
[May 1 06:22:49] -- <Local/8600051@default-b10d,1> Playing 'invalid' (language 'en')
[May 1 06:22:49] WARNING[13110]: file.c:1297 waitstream_core: Unexpected control subclass '-1'
[May 1 06:22:49] WARNING[13110]: file.c:1297 waitstream_core: Unexpected control subclass '-1'
[May 1 06:22:51] == Manager 'sendcron' logged off from 127.0.0.1
[May 1 06:22:53] == Auto fallthrough, channel 'Local/8600051@default-b10d,1' status is 'UNKNOWN'
[May 1 06:22:53] -- Executing [h@default:1] DeadAGI("Local/8600051@default-b10d,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[May 1 06:22:53] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[May 1 06:22:53] == Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-b10d,2'
[May 1 06:22:53] -- Executing [h@default:1] DeadAGI("Local/8600051@default-b10d,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[May 1 06:22:53] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[May 1 06:23:01] == Parsing '/etc/asterisk/manager.conf': [May 1 06:23:01] Found
[May 1 06:23:01] == Manager 'sendcron' logged on from 127.0.0.1

I appreciate all your help.

Thanks

[williamconley]

undo any changes to extension.conf that you may have made.

Add a Carrier (SIP or IAX) with that exten configuration in the Dialplan area and Vicidial will add the lines to an included file of extensions.conf in the correct area.

If you still experience an issue, consider:

XXXXXXXXX. instead of XXXXXXXXXXX

Note two less "X" and an added "."

This will result in any number at least that many digits matching. No more "invalid extension" messages.

[ktsis]

From vicidial admin
admin-->carriers-->add a new carrier

carrier id: local_call
carrier name: local call
protocol: sip
dialplan:
[default]
exten => _XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _XXXXXXXXXX,2,Dial(${TRUNK}/${EXTEN:1},,tTor)
exten => _XXXXXXXXXX,3,Hangup

also tried this dialplan
[default]
exten => _XXXXXXXXX.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _XXXXXXXXX.,2,Dial(${TRUNK}/${EXTEN:1},,tTor)
exten => _XXXXXXXXX.,3,Hangup

active: y

i submit the page and reload asterisk

When i dial, i still get the voice message "this is not a vaild extension"

Here is my log:

-- Executing [8600051@default:1] MeetMe("Local/8600051@default-cbc7,2", "8600051|F") in new stack
== Starting Local/8600051@default-cbc7,1 at default,08060700544,1 failed so falling back to exten 's'
== Starting Local/8600051@default-cbc7,1 at default,s,1 still failed so falling back to context 'default'
-- Sent into invalid extension 's' in context 'default' on Local/8600051@default-cbc7,1
-- Executing [i@default:1] Playback("Local/8600051@default-cbc7,1", "invalid") in new stack
-- <Local/8600051@default-cbc7,1> Playing 'invalid' (language 'en')
[May 1 07:22:02] WARNING[20209]: file.c:1297 waitstream_core: Unexpected control subclass '-1'
[May 1 07:22:02] WARNING[20209]: file.c:1297 waitstream_core: Unexpected control subclass '-1'
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1


Thanks

[striker]

go to asterisk cli and type this

dialplan show 08060700544@default


-- did you created new default context in extensions.conf
or entered those dialplans under the existing default context

[striker]

i think you are adding the line [default] in diaplan entery

just add the dialplan "no need to add the line [default]"
exten => _XXXXXXXXX.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _XXXXXXXXX.,2,Dial(${TRUNK}/${EXTEN:1},,tTor)
exten => _XXXXXXXXX.,3,Hangup

[ktsis]

Thanks alot.

Now when i call from agent interface, i get a voice message that "the number i dial does not exist". I also notice that on the gsm terminal connected to the fxo card, the led shows the number i am calling but it removes the first digit. if i dial 08011111111, i see 8011111111. I dont know what i am missing out.

here is the dial plan i use:
exten => _XXXXXXXXX.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _XXXXXXXXX.,2,Dial(${TRUNK}/${EXTEN:1},,tTor)
exten => _XXXXXXXXX.,3,Hangup

Thanks.

[ktsis]

Thanks all for your supports.

please i noticed something, when call comes into agent web interface it does not display the mobile number calling. but it shows this in the status:

STATUS: Incoming: ()- Group- fxoinbound Fronter: - default Session ID: 8600051


There is a view of my log

[May 12 00:00:18] NOTICE[7488] chan_dahdi.c: Got event 2 (Ring/Answered)...
[May 12 00:00:18] VERBOSE[7488] logger.c: [May 12 00:00:18] -- Executing [s@from-pstn:1] AGI("DAHDI/1-1", "agi-DID_route.agi") in new stack
[May 12 00:00:18] VERBOSE[7488] logger.c: [May 12 00:00:18] -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-DID_route.agi
[May 12 00:00:18] VERBOSE[7488] logger.c: [May 12 00:00:18] -- AGI Script agi-DID_route.agi completed, returning 0
[May 12 00:00:18] VERBOSE[7488] logger.c: [May 12 00:00:18] -- Executing [99909*1***DID@default:1] Answer("DAHDI/1-1", "") in new stack
[May 12 00:00:18] DEBUG[7488] chan_dahdi.c: Took DAHDI/1-1 off hook
[May 12 00:00:18] DEBUG[7488] chan_dahdi.c: No echo training requested
[May 12 00:00:18] DEBUG[7488] chan_dahdi.c: Requested indication -1 on channel DAHDI/1-1
[May 12 00:00:18] VERBOSE[7488] logger.c: [May 12 00:00:18] -- Executing [99909*1***DID@default:2] AGI("DAHDI/1-1", "agi-VDAD_ALL_inbound.agi") in new stack
[May 12 00:00:18] VERBOSE[7488] logger.c: [May 12 00:00:18] -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_inbound.agi
[May 12 00:00:19] VERBOSE[7488] logger.c: [May 12 00:00:19] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[May 12 00:00:19] VERBOSE[7488] logger.c: [May 12 00:00:19] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[May 12 00:00:19] VERBOSE[7516] logger.c: [May 12 00:00:19] == Parsing '/etc/asterisk/manager.conf': [May 12 00:00:19] VERBOSE[7516] logger.c: [May 12 00:00:19] Found
[May 12 00:00:19] VERBOSE[7516] logger.c: [May 12 00:00:19] == Manager 'sendcron' logged on from 127.0.0.1
[May 12 00:00:19] VERBOSE[7517] logger.c: [May 12 00:00:19] -- Executing [192*168*100*230*78600051@default:1] Goto("Local/192*168*100*230*78600051@default-d5aa,2", "default|78600051|1") in new stack
[May 12 00:00:19] VERBOSE[7517] logger.c: [May 12 00:00:19] -- Goto (default,78600051,1)
[May 12 00:00:19] VERBOSE[7517] logger.c: [May 12 00:00:19] -- Executing [78600051@default:1] MeetMe("Local/192*168*100*230*78600051@default-d5aa,2", "8600051|Fq") in new stack
[May 12 00:00:19] VERBOSE[7516] logger.c: [May 12 00:00:19] > Channel Local/192*168*100*230*78600051@default-d5aa,1 was answered.
[May 12 00:00:19] VERBOSE[7518] logger.c: [May 12 00:00:19] -- Executing [83047777777777@vicidial-auto:1] Answer("Local/192*168*100*230*78600051@default-d5aa,1", "") in new stack
[May 12 00:00:19] VERBOSE[7518] logger.c: [May 12 00:00:19] -- Executing [83047777777777@vicidial-auto:2] Playback("Local/192*168*100*230*78600051@default-d5aa,1", "ding") in new stack
[May 12 00:00:19] VERBOSE[7518] logger.c: [May 12 00:00:19] -- <Local/192*168*100*230*78600051@default-d5aa,1> Playing 'ding' (language 'en')
[May 12 00:00:19] WARNING[7518] file.c: Unexpected control subclass '-1'
[May 12 00:00:19] WARNING[7518] file.c: Unexpected control subclass '-1'
[May 12 00:00:19] DEBUG[7488] chan_dahdi.c: Ignore switch to REVERSED Polarity on channel 1, state 6
[May 12 00:00:19] DEBUG[7488] chan_dahdi.c: Ignoring Polarity switch to IDLE on channel 1, state 6
[May 12 00:00:19] DEBUG[7488] chan_dahdi.c: Polarity Reversal event occured - DEBUG 2: channel 1, state 6, pol= 0, aonp= 0, honp= 0, pdelay= 600, tv= -1763347985
[May 12 00:00:19] VERBOSE[7518] logger.c: [May 12 00:00:19] -- Executing [83047777777777@vicidial-auto:3] Hangup("Local/192*168*100*230*78600051@default-d5aa,1", "") in new stack
[May 12 00:00:19] VERBOSE[7518] logger.c: [May 12 00:00:19] == Spawn extension (vicidial-auto, 83047777777777, 3) exited non-zero on 'Local/192*168*100*230*78600051@default-d5aa,1'
[May 12 00:00:19] VERBOSE[7518] logger.c: [May 12 00:00:19] -- Executing [h@vicidial-auto:1] DeadAGI("Local/192*168*100*230*78600051@default-d5aa,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
[May 12 00:00:19] VERBOSE[7518] logger.c: [May 12 00:00:19] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[May 12 00:00:19] VERBOSE[7517] logger.c: [May 12 00:00:19] == Spawn extension (default, 78600051, 1) exited non-zero on 'Local/192*168*100*230*78600051@default-d5aa,2'
[May 12 00:00:19] VERBOSE[7517] logger.c: [May 12 00:00:19] -- Executing [h@default:1] DeadAGI("Local/192*168*100*230*78600051@default-d5aa,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[May 12 00:00:19] VERBOSE[7517] logger.c: [May 12 00:00:19] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[May 12 00:00:20] VERBOSE[7488] logger.c: [May 12 00:00:20] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[May 12 00:00:20] VERBOSE[7488] logger.c: [May 12 00:00:20] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[May 12 00:00:20] VERBOSE[7488] logger.c: [May 12 00:00:20] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[May 12 00:00:20] VERBOSE[7488] logger.c: [May 12 00:00:20] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[May 12 00:00:20] VERBOSE[7488] logger.c: [May 12 00:00:20] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[May 12 00:00:20] VERBOSE[7488] logger.c: [May 12 00:00:20] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[May 12 00:00:20] ERROR[7488] utils.c: write() returned error: Broken pipe
[May 12 00:00:20] VERBOSE[7488] logger.c: [May 12 00:00:20] -- AGI Script agi-VDAD_ALL_inbound.agi completed, returning 0
[May 12 00:00:20] VERBOSE[7488] logger.c: [May 12 00:00:20] -- Executing [192*168*100*230*8600051@default:1] Goto("DAHDI/1-1", "default|8600051|1") in new stack
[May 12 00:00:20] VERBOSE[7488] logger.c: [May 12 00:00:20] -- Goto (default,8600051,1)
[May 12 00:00:20] VERBOSE[7488] logger.c: [May 12 00:00:20] -- Executing [8600051@default:1] MeetMe("DAHDI/1-1", "8600051|F") in new stack
[May 12 00:00:20] DEBUG[7488] chan_dahdi.c: Requested indication -1 on channel DAHDI/1-1
[May 12 00:00:21] VERBOSE[7516] logger.c: [May 12 00:00:21] == Manager 'sendcron' logged off from 127.0.0.1
[May 12 00:00:32] DEBUG[7488] chan_dahdi.c: Ignore switch to REVERSED Polarity on channel 1, state 6
[May 12 00:00:32] DEBUG[7488] chan_dahdi.c: Ignoring Polarity switch to IDLE on channel 1, state 6
[May 12 00:00:32] DEBUG[7488] chan_dahdi.c: Polarity Reversal event occured - DEBUG 2: channel 1, state 6, pol= 0, aonp= 0, honp= 0, pdelay= 600, tv= -1763335379
[May 12 00:00:45] VERBOSE[7622] logger.c: [May 12 00:00:45] == Parsing '/etc/asterisk/manager.conf': [May 12 00:00:45] VERBOSE[7622] logger.c: [May 12 00:00:45] Found
[May 12 00:00:45] VERBOSE[7622] logger.c: [May 12 00:00:45] == Manager 'sendcron' logged on from 127.0.0.1
[May 12 00:00:45] VERBOSE[7488] logger.c: [May 12 00:00:45] == Spawn extension (default, 8600051, 1) exited non-zero on 'DAHDI/1-1'
[May 12 00:00:45] VERBOSE[7488] logger.c: [May 12 00:00:45] -- Executing [h@default:1] DeadAGI("DAHDI/1-1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[May 12 00:00:45] VERBOSE[7488] logger.c: [May 12 00:00:45] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[May 12 00:00:45] VERBOSE[7488] logger.c: [May 12 00:00:45] -- Hungup 'DAHDI/1-1'


What do i need to do to make it display the incoming mobile number

Thanks

[striker]

since you are using FXO connection there is no gurantee that you will get the Caller id for the incomming calls.

check with the provider what singnalling they use and change the same in chan_dahdi.conf

cidsignalling=v23
; Type of caller ID signalling in use
; bell = bell202 as used in US (default)
; v23 = v23 as used in the UK
; v23_jp = v23 as used in Japan
; dtmf = DTMF as used in Denmark, Sweden and Netherlands
; smdi = Use SMDI for caller ID. Requires SMDI to be enabled (usesmdi).

cidstart=polarity4
; What signals the start of caller ID
; ring = a ring signals the start (default)
; polarity = polarity reversal signals the start
; polarity_IN = polarity reversal signals the start, for India,
; for dtmf dialtone detection; using DTMF.
;

[ktsis]

I am trying something new with same vicidial above. I got a four channel GSM Gateway (GoIP) from a friend. Please i need a guide on now to configure it to make it work with vicidial. Appeciate all your support and teaching. Thanks

[ktsis]

here is what i have tried.

I connect the goip4 gateway to computer running vicidial on same subnet.

GOIP4 ip address = 192.168.100.251/24
Vicidial Computer= 192.168.100.252/24


I logged into the Goip4 gateway and configure it as follows:

Call Settings
config mode: Trunk Gateway Mode
Sip Trunk Gateway 1 : 192.168.100.252 (ip address of vicidial computer)
Phone Number: 7711
Authentication ID: 7711


Call Divert
I enable Forward to VOIP
Forward Number(PSTN to VOIP):8001

Inside my vicidial computer i did the following.

i added the following to sip.conf file:
[7711]
calledid="Outside Line" <200>
canreinvite=no
context=from-gsm1
dtmfmode=rfc2833
host=192.168.100.251
port=5066



i also add the following to extensions.conf
[from-gsm1]
exten => 8001,1,Answer
exten => 8001,2,AGI(agi-AGENT_dial_in.agi)
exten => 8001,3,Hangup

exten => _X.,1,AGI(agi-DID_route.agi)
exten => h,1,DeadAGI(agi://127.0.0.1:4577/call_log--HVcauses ... EBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME})

When i call a gsm line inserted into the GOIP4 gateway, it rings on the softphone but does not show in the agent interface.

Please i do i make the call to go to the agent web interface.

kindly assist.

Thanks