vicidial.org

Hello;

GoAutoDial CE 2.1, 2.6.18-238.9.1.el5.goPAE (SMP), Intel(R) Core(TM) i7-3770 CPU @ 3.40GHz, Single Machine.

Is it condition to be able to get calls at the agent Phone is to accept the call when the agent login and to stay connected? But this is really bothering to keep the Phone connected and if in mistake the button pressed to hangup, then the agent will not calls. Any other solution? Is there a setting to overcome this (so the agent cal get calls even if the line is not connected)?

Also, when the agent login and he got the call from the server to accept it, the caller id that appears is very complicated (related maybe to the conference id or something like this), how can we change it and use a friendly numbers to appear at the Phone LCD?

Regards
Bilal

you have not listed your vicidial version and build. for shame.

you can check out the "on-hook agent" settings if you like. they also have preferences for the callerid. depending on your version of Vicidial (which you have not listed). you should know that your version of GoAutodial is useful, but goautodial is just the installer for Vicidial. Vicidial can also be installed with Vicibox (from Vicibox.com) or from source code. But the software you are asking questions about is Vicidial, not goautodial. The Vicidial Manager's Manual may be useful to you (available from EFLO.net ... run by The Vicidial Group ...).

VERSION: 2.4-309a, BUILD: 110430-1642, GoAutoDial CE 2.1, 2.6.18-238.9.1.el5.goPAE (SMP), Intel(R) Core(TM) i7-3770 CPU @ 3.40GHz, Single Machine.

You mean by the "on-hook agent": that is mentioned in the Admin --> Phone ? There, I found only on-hook agent (Y/N). So, if I need to receive calls without keeping the line connected, then I have to select "Y"? As now it is "N".
About the CallerID, this I did not get where are you talking about it? Is it also at the Admin --> Phone?

on-hook phones do not ring when the agent logs in. instead, they ring when a call is sent to the agent. the callerid of that call is controlled by the Ingroup setting for on-hook callerid and can be set to be the customer's callerid.

if the agent does not answer, the call will ring to the next agent instead. while the client listens to hold music if you have that configured. after each prospect, the agent will hang up again and wait for the phone to ring. but the agent must still control the call in the agent screen (including dispositioning each call after hangup) or they will never receive another call.

if you change on-hook to "Y", the agent phone will not ring until the agent gets a call.

But my question was about something else.
I need the agent to receive calls without staying the line connected. In other words, when agent login, he get a calls from the server and the agent should accept it to be part of the conference, right? And if we look to the phine, we will find the button line at the phone is online (commented), in this case, the agent is getting calls, right?
What if the agent disconnected the button line, he will get any more calls? Or the agent should stay in the conference?
I am talking about the incoming calls and not the outbound calls.

bghayad wrote:I need the agent to receive calls without staying the line connected. In other words, when agent login, he get a calls from the server and the agent should accept it to be part of the conference, right?

williamconley wrote:on-hook phones do not ring when the agent logs in. instead, they ring when a call is sent to the agent.

they do not stay on the line connected. they only connect when a call actually comes to them which will cause their phone to ring.

I selected on-hook to be "Y" and yes now the agent is login without ring on the Phone. The call is coming to the Phone but when I answer it from the Phone, I am hearing the message that this is a not valid extension ! Also, I see in the Phone display "RANGEAGENT ....", i do not know what they mean by this Range.

Is it required to configured the allowed number to get from? Is it because I selected on-hook to be "Y", why?
When on-hook is "N", so I have to accept the call when I login .. I am getting the calls and I am able to talk .. But, I do not need this scenario.

Appreciate the kindly help to know why I am hearing the message that the extension is not valid .. and I do not see the call is coming to the agent screen.

Regards
Bilal

show asterisk command line output from a single test call where "this is not a valid extension" is played. we need to see what extension it tried to dial to find out what happened. not 3000 lines of unrelated code or other calls. just a single test call for this purpose.

Code: Select all

asterisk -R

will provide the asterisk command line.

Code: Select all

screen -r asterisk

will also provide asterisk command line with extra perl debugging information, but this method is dangerous. you must exit it with [ctrl-A][ctrl-D] to avoid crashing asterisk. if you enter [ctrl-C] while in this mode, you shut off asterisk and have to reboot. LOL

Hello;

This is the CLI logs:

[Apr 4 09:06:56] -- Executing [22281258@trunkinbound:1] AGI("SIP/kems-00000071", "agi-DID_route.agi") in new stack
[Apr 4 09:06:56] -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-DID_route.agi
[Apr 4 09:06:56] -- AGI Script Executing Application: (Monitor) Options: (wav|/var/spool/asterisk/monitor/MIX/20130404090656_22281258_50911347)
[Apr 4 09:06:56] ERROR[10719]: utils.c:967 ast_carefulwrite: write() returned error: Broken pipe
[Apr 4 09:06:56] -- AGI Script agi-DID_route.agi completed, returning 0
[Apr 4 09:06:56] -- Executing [99909*5***DID@default:1] Answer("SIP/kems-00000071", "") in new stack
[Apr 4 09:06:56] -- Executing [99909*5***DID@default:2] AGI("SIP/kems-00000071", "agi-VDAD_ALL_inbound.agi") in new stack
[Apr 4 09:06:56] -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_inbound.agi
[Apr 4 09:06:56] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Apr 4 09:06:56] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Apr 4 09:06:56] -- AGI Script Executing Application: (StopMonitor) Options: (wav|/var/spool/asterisk/monitor/MIX/20130404090656_22281258_50911347)
[Apr 4 09:06:56] == Parsing '/etc/asterisk/manager.conf': [Apr 4 09:06:56] Found
[Apr 4 09:06:56] == Manager 'sendcron' logged on from 127.0.0.1
[Apr 4 09:06:56] -- Executing [010*149*001*130*501@default:1] Goto("Local/010*149*001*130*501@default-b24d,2", "default|501|1") in new stack
[Apr 4 09:06:56] -- Goto (default,501,1)
[Apr 4 09:06:56] -- Executing [501@default:1] Dial("Local/010*149*001*130*501@default-b24d,2", "SIP/501|60|") in new stack
[Apr 4 09:06:56] -- Called 501
[Apr 4 09:06:56] -- SIP/501-00000072 is ringing
[Apr 4 09:06:58] WARNING[10719]: res_musiconhold.c:692 get_mohbyname: Music on Hold class 'default' not found
[Apr 4 09:06:58] WARNING[10719]: res_musiconhold.c:692 get_mohbyname: Music on Hold class 'default' not found
[Apr 4 09:07:01] == Parsing '/etc/asterisk/manager.conf': [Apr 4 09:07:01] Found
[Apr 4 09:07:01] == Manager 'sendcron' logged on from 127.0.0.1
[Apr 4 09:07:01] == Parsing '/etc/asterisk/manager.conf': [Apr 4 09:07:01] Found
[Apr 4 09:07:01] == Manager 'sendcron' logged on from 127.0.0.1
[Apr 4 09:07:02] == Manager 'sendcron' logged off from 127.0.0.1
[Apr 4 09:07:02] -- SIP/501-00000072 answered Local/010*149*001*130*501@default-b24d,2
[Apr 4 09:07:02] > Channel Local/010*149*001*130*501@default-b24d,1 was answered.
[Apr 4 09:07:02] == Starting Local/010*149*001*130*501@default-b24d,1 at default,8331*39*Y4040906560000000017*agent001*501,1 failed so falling back to exten 's'
[Apr 4 09:07:02] == Starting Local/010*149*001*130*501@default-b24d,1 at default,s,1 still failed so falling back to context 'default'
[Apr 4 09:07:02] -- Sent into invalid extension 's' in context 'default' on Local/010*149*001*130*501@default-b24d,1
[Apr 4 09:07:02] -- Executing [i@default:1] Playback("Local/010*149*001*130*501@default-b24d,1", "invalid") in new stack
[Apr 4 09:07:02] -- <Local/010*149*001*130*501@default-b24d,1> Playing 'invalid' (language 'en')
[Apr 4 09:07:02] WARNING[10792]: file.c:1297 waitstream_core: Unexpected control subclass '-1'
[Apr 4 09:07:02] -- Executing [h@default:1] DeadAGI("Local/010*149*001*130*501@default-b24d,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----6-----0") in new stack
[Apr 4 09:07:02] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---6-----0 completed, returning 0
[Apr 4 09:07:02] == Spawn extension (default, 501, 1) exited non-zero on 'Local/010*149*001*130*501@default-b24d,2'
[Apr 4 09:07:02] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Apr 4 09:07:02] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Apr 4 09:07:02] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Apr 4 09:07:02] -- Playing 'generic_hold' (escape_digits=) (sample_offset 0)
[Apr 4 09:07:04] == Manager 'sendcron' logged off from 127.0.0.1
[Apr 4 09:07:04] == Manager 'sendcron' logged off from 127.0.0.1
[Apr 4 09:07:05] -- Executing [h@default:1] DeadAGI("SIP/kems-00000071", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Apr 4 09:07:05] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Apr 4 09:07:06] == Spawn extension (default, i, 1) exited non-zero on 'SIP/501-00000072'
[Apr 4 09:07:06] -- Executing [h@default:1] DeadAGI("SIP/501-00000072", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Apr 4 09:07:06] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Apr 4 09:07:07] == Parsing '/etc/asterisk/manager.conf': [Apr 4 09:07:07] Found
[Apr 4 09:07:07] == Manager 'sendcron' logged on from 127.0.0.1
[Apr 4 09:07:07] == Manager 'sendcron' logged off from 127.0.0.1
go*CLI>

Regards
Bilal

the call is generated to your agent, and when the agent answers the system must be 'notified' that the agent has accepted the call to route the agent and prospect to the same room.

apparently this notification is handled by a second leg of the call to the agent passing through a special extension to initiate the process, and this specially coded call is failing:

Code: Select all

Starting Local/010*149*001*130*501@default-b24d,1 at default,8331*39*Y4040906560000000017*agent001*501,1 failed so falling back to exten 's'

apparently your system is having a tough time routing the call to that server/meetme room after the agent answers his leg of the call.

have you altered the IP address of your server after installation? of course, it may also be a bug in your version of Vicidial (it's kinda old ...).

Yes of course I changed the IP address after installation .. multiple times.
I did not understand your sentence that my system is having a tough time routing ...

Well, What is the solution? If it is bug, how I can fix it? I do not need to do re-installation please !

Regards
Bilal

run this script one time for each and every "prior" ip address the system had:

Code: Select all

/usr/share/astguiclient/ADMIN_update_server_ip.pl

each of the OLD ips will be used, but the NEW ip will always be the present ip. does not hurt to run this script as often as you want to, as long as the NEW ip is correct.

Do you mean if I changed the IP address 5 times, I have to apply the script 5 times?
Also, you told me that my build is kinda old, so if I downloaded a goautodial now from the website, I will have newer build? How I can upgrade to this newer build?

When I was changing the IP address, always I was applying the command: update_server_ip, so what is the difference?

Although, I did not test yet, but meanwhile I applied it maybe 10 times .. and the below is the result (Appreciate if you tell me if I am in the right direction):

[root@go astguiclient]# perl ADMIN_update_server_ip.pl
Previous astGUIclient configuration file found at: /etc/astguiclient.conf

Would you like to use interactive mode (y/n): [y]

STARTING SERVER IP ADDRESS CHANGE FOR VICIDIAL...

Old server IP address or press enter for default: [10.149.1.130]

server IP address or press enter for default: [10.149.1.130]

old server_ip: 10.149.1.130
new server_ip: 10.149.1.130

Are these settings correct?(y/n): [y]
Writing change to astguiclient.conf file: /etc/astguiclient.conf

STARTING DATABASE TABLES UPDATES PHASE...
Updating servers table...
|1|UPDATE servers SET server_ip='10.149.1.130' where server_ip='10.149.1.130';|
Updating phones table...
|40|UPDATE phones SET server_ip='10.149.1.130' where server_ip='10.149.1.130';|
Updating inbound_numbers table...
|0E0|UPDATE inbound_numbers SET server_ip='10.149.1.130' where server_ip='10.149.1.130';|
Updating server_updater table...
|1|UPDATE server_updater SET server_ip='10.149.1.130' where server_ip='10.149.1.130';|
Updating conferences table...
|49|UPDATE conferences SET server_ip='10.149.1.130' where server_ip='10.149.1.130';|
Updating vicidial_conferences table...
|249|UPDATE vicidial_conferences SET server_ip='10.149.1.130' where server_ip='10.149.1.130';|
Updating vicidial_stations table...
|0E0|UPDATE vicidial_stations SET server_ip='10.149.1.130' where server_ip='10.149.1.130';|
Updating vicidial_remote_agents table...
|1|UPDATE vicidial_remote_agents SET server_ip='10.149.1.130' where server_ip='10.149.1.130';|
Updating phone_favorites table...
|0E0|UPDATE phone_favorites SET server_ip='10.149.1.130' where server_ip='10.149.1.130';|
Updating vicidial_server_trunks table...
|0E0|UPDATE vicidial_server_trunks SET server_ip='10.149.1.130' where server_ip='10.149.1.130';|
Updating vicidial_server_carriers table...
|5|UPDATE vicidial_server_carriers SET server_ip='10.149.1.130' where server_ip='10.149.1.130';|
Updating vicidial_inbound_dids table...
|2|UPDATE vicidial_inbound_dids SET server_ip='10.149.1.130' where server_ip='10.149.1.130';|
Updating vicidial_process_triggers table...
|0E0|UPDATE vicidial_process_triggers SET server_ip='10.149.1.130' where server_ip='10.149.1.130';|
Setting servers to rebuild conf files...
|1|UPDATE servers SET rebuild_conf_files='Y' where generate_vicidial_conf='Y' and active_asterisk_server='Y';|

SERVER IP ADDRESS CHANGE FOR VICIDIAL FINISHED!


- process runtime (1 sec) (0.0166666666666667 minutes)
[root@go astguiclient]#
[root@go astguiclient]#

Regards
Bilal

Dear William;

I did a test .. The same result ..
What I see in the LCD is the following (when the call is ringing):

RINGAGENT001...

And after hearing the message: I am sorry that is not a valid extension please try again, after little the Phone after is ringing again ... At the agent screen, I see that Calls in Queue is 1 (until the caller disconnect) .. But of course, I do not see at the agent screen any caller information like the caller_id.

But when I selected On-Hook Agent "NO", so the agent has to accept the call when he login .. then things are working fine .. but the problem that the agent should stay connected And I do not need this.

Is it required to have two extensions assigned to the two buttons at the phones, so maybe the transfer can happen?

By looking to this error:

Starting Local/010*149*001*130*501@default-b24d,1 at default,8331*39*Y4040906560000000017*agent001*501,1 failed so falling back to exten 's'

The question is: What is the 8331* and what the system is trying to do by this? Also, what is meant by Local?

Regards
Bilal

Resolved.
From this link: http://www.vicidial.org/VICIDIALforum/v ... hp?t=23296

Specifically this:

I was able to resolved this by adding the following entries on the extensions.conf inside the default context.

; these are used for the ring_all function in VICIDIAL
exten => _8331*.,1,Playback(sip-silence)
exten => _8331*.,n,AGI(agi-VDAD_RINGALL.agi,${EXTEN})
exten => _8331*.,n,AGI(agi-VDAD_RINGALL.agi,${EXTEN})
exten => _8331*.,n,AGI(agi-VDAD_RINGALL.agi,${EXTEN})
exten => _8331*.,n,Hangup

Regards
Bilal

Here's a silly question ... why were those lines Not There Already? How did they go missing?

Good postback, by the way. Specific solution to a posted error message.