Campaign with no agents

[kashutu]

Hi guys, here is my Goautodial specs

GoAutoDial CE 2.0 --- ISO
VERSION: 2.2.1-237
BUILD: 100510-2015

Guys i need some guidelines ASAP. I am stuck with one dilema which can use the expertise. Here is the senario.

I want to run the campaign with no agent logged in, so i created a remote agent with 50 lines and created the campaign with one number only. I want all those 50 lines dial the same number over and over again. This is not exactly a survey campaign. I just want dialer to dial the number and keep the call alive for as long as possible, on the other end there will be recording which will be played after the call gets connected. So, lets say that i am dialing the number from my landline and i hear the recording on the other end. After the recording has been done playing, i keep the call alive by not hanging up. I just want to achieve this from my vicidial. If someone can help me out i shall be very greatfull.

PLEASE DON'T IGNORE THIS THREAD IAM IN A DESPARATE HELP.

I have done some testing but i am not able to reach my goal. Dialer is dialing the number fine, but right after it dials, it hangsup immediatley. I think the call does get connected but then dialer try to tranfer it to the agent but no one is there take the call and then dialer hangsup the call, here is the CLI>

[Dec 11 13:16:18] -- Executing [00441698231076@default:1] AGI("Local/00441698231076@default-da7f,2", "agi://127.0.0.1:4577/call_log") in new stack
[Dec 11 13:16:18] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Dec 11 13:16:18] -- Executing [00441698231076@default:2] Dial("Local/00441698231076@default-da7f,2", "SIP/hussain/441698231076||Ttor") in new stack
[Dec 11 13:16:18] -- Called hussain/441698231076
[Dec 11 13:16:21] -- SIP/hussain-0000222c is making progress passing it to Local/00441698231076@default-da7f,2
[Dec 11 13:16:40] -- SIP/hussain-0000222c is circuit-busy
[Dec 11 13:16:40] == Everyone is busy/congested at this time (1:0/1/0)
[Dec 11 13:16:40] -- Executing [00441698231076@default:3] Hangup("Local/00441698231076@default-da7f,2", "") in new stack
[Dec 11 13:16:40] == Spawn extension (default, 00441698231076, 3) exited non-zero on 'Local/00441698231076@default-da7f,2'
[Dec 11 13:16:40] -- Executing [h@default:1] DeadAGI("Local/00441698231076@default-da7f,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----1-----CONGESTION----------") in new stack

Thank You all for looking and helping

[kashutu]

Anyone with any suggestion or even any thoughts that might help please.


Thank You

[sniper888]

Hi,

Try to check :
1.your carrier setting.
2.number you dial
3.Your allowed country to dial using your carrier.

[kashutu]

Thanx snipper. I checked the number it wasn't working. Now i have changed the number, and after activating remote agent, the phone is ringing at the other end but call is not getting connected, just to include this here that on the other end is the recording that gets played after the call in connected. You might think of it as an answering machine.

HERE IS THE CLI WITH REMOTE AGENT:

[Dec 13 02:30:04] -- Executing [00442868628897@default:1] AGI("Local/00442868628897@default-6921,2", "agi://127.0.0.1:4577/call_log") in new stack
[Dec 13 02:30:04] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Dec 13 02:30:04] -- Executing [00442868628897@default:2] Dial("Local/00442868628897@default-6921,2", "SIP/hussain/442868628897||Ttor") in new stack
[Dec 13 02:30:04] -- Called hussain/442868628897
[Dec 13 02:30:06] -- SIP/hussain-000003d7 is making progress passing it to Local/00442868628897@default-6921,2
[Dec 13 02:30:07] == Parsing '/etc/asterisk/manager.conf': [Dec 13 02:30:07] Found
[Dec 13 02:30:07] == Manager 'sendcron' logged on from 127.0.0.1
[Dec 13 02:30:07] == Manager 'sendcron' logged off from 127.0.0.1
[Dec 13 02:30:08] == Manager 'sendcron' logged off from 127.0.0.1
[Dec 13 02:30:09] -- SIP/hussain-000003d7 is ringing
[Dec 13 02:30:21] -- SIP/hussain-000003d7 is circuit-busy
[Dec 13 02:30:21] == Everyone is busy/congested at this time (1:0/1/0)
[Dec 13 02:30:21] -- Executing [00442868628897@default:3] Hangup("Local/00442868628897@default-6921,2", "") in new stack
[Dec 13 02:30:21] == Spawn extension (default, 00442868628897, 3) exited non-zero on 'Local/00442868628897@default-6921,2'
[Dec 13 02:30:21] -- Executing [h@default:1] DeadAGI("Local/00442868628897@default-6921,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----1-----CONGESTION----------") in new stack

HERE IS THE CLI WITH MANUAL DIALING WHICH IS WORKING FINE:

[Dec 13 02:34:09] -- Executing [00442868628897@default:1] AGI("Console/dsp", "agi://127.0.0.1:4577/call_log") in new stack
[Dec 13 02:34:09] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Dec 13 02:34:09] -- Executing [00442868628897@default:2] Dial("Console/dsp", "SIP/hussain/442868628897||Ttor") in new stack
[Dec 13 02:34:09] -- Called hussain/442868628897
[Dec 13 02:34:11] -- SIP/hussain-000003dc is making progress passing it to Console/dsp
[Dec 13 02:34:40] -- SIP/hussain-000003dc answered Console/dsp
[Dec 13 02:34:40] << Console call has been answered >>
[Dec 13 02:34:40] WARNING[15307]: chan_oss.c:686 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory
[Dec 13 02:34:41] WARNING[15307]: chan_oss.c:686 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory
[Dec 13 02:34:42] WARNING[15307]: chan_oss.c:686 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory
[Dec 13 02:34:43] WARNING[15307]: chan_oss.c:686 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory
[Dec 13 02:34:44] WARNING[15307]: chan_oss.c:686 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory
[Dec 13 02:34:45] WARNING[15307]: chan_oss.c:686 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory
[Dec 13 02:34:46] WARNING[15307]: chan_oss.c:686 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory
[Dec 13 02:34:47] WARNING[15307]: chan_oss.c:686 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory
[Dec 13 02:34:48] WARNING[15307]: chan_oss.c:686 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory
[Dec 13 02:34:49] WARNING[15307]: chan_oss.c:686 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory

When i dial manually, the does get connected and it stays alive untill i hang it up manually. However, with Remote Agent, the call doesn't get connected even though it does ring. Can someone please let me know how can i get it to work with REMOTE AGENT.

Thanks

[williamconley]

Code: Select all

AGI("Console/dsp", "agi://1

This is an example of a call made at the console. Please post one made from an agent ...

[kashutu]

Code: Select all

AGI("Console/dsp", "agi://1

This is an example of a call made at the console. Please post one made from an agent ...

here is the CLI with agent logged in:

[Dec 23 13:45:47] -- Executing [00441416431271@default:1] AGI("SIP/cc101-00000022", "agi://127.0.0.1:4577/call_log") in new stack
[Dec 23 13:45:47] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Dec 23 13:45:47] -- Executing [00441416431271@default:2] Dial("SIP/cc101-00000022", "SIP/hussain/441416431271||Ttor") in new stack
[Dec 23 13:45:47] -- Called hussain/441416431271
[Dec 23 13:45:53] -- SIP/hussain-00000023 is ringing
[Dec 23 13:45:53] -- SIP/hussain-00000023 answered SIP/cc101-00000022
[Dec 23 13:45:53] WARNING[2132]: res_musiconhold.c:692 get_mohbyname: Music on Hold class 'default' not found
[Dec 23 13:45:53] WARNING[2132]: res_musiconhold.c:692 get_mohbyname: Music on Hold class 'default' not found
[Dec 23 13:45:53] WARNING[2132]: res_musiconhold.c:692 get_mohbyname: Music on Hold class 'default' not found
[Dec 23 13:45:56] WARNING[995]: res_musiconhold.c:692 get_mohbyname: Music on Hold class 'default' not found
[Dec 23 13:45:56] WARNING[995]: res_musiconhold.c:692 get_mohbyname: Music on Hold class 'default' not found
[Dec 23 13:45:56] WARNING[995]: res_musiconhold.c:692 get_mohbyname: Music on Hold class 'default' not found
[Dec 23 13:45:56] WARNING[2056]: res_musiconhold.c:692 get_mohbyname: Music on Hold class 'default' not found
[Dec 23 13:45:56] WARNING[2056]: res_musiconhold.c:692 get_mohbyname: Music on Hold class 'default' not found
[Dec 23 13:45:56] WARNING[2056]: res_musiconhold.c:692 get_mohbyname: Music on Hold class 'default' not found

[williamconley]

ok, so you said with remote it rings but does not answer ... this shows a call ringing and then answering. It also shows that your MOH is not configured properly, but that's not related.

So ... is there a question in here ...?

[kashutu]

Sorry William when you said to get you the CLI with an agent logged in, i actually had a live agent logged in and the CLI was with agent cc101 taking live call.

Here is the CLI with the remote agent logged in.

[Dec 23 15:48:38] -- Executing [8368@default:2] AGI("SIP/hussain-0000001e", "agi://127.0.0.1:4577/call_log") in new stack
[Dec 23 15:48:38] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Dec 23 15:48:38] -- Executing [8368@default:3] AGI("SIP/hussain-0000001e", "agi-VDAD_ALL_outbound.agi|NORMAL-----LB") in new stack
[Dec 23 15:48:38] -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi
[Dec 23 15:48:39] -- AGI Script Executing Application: (Monitor) Options: (wav|/var/spool/asterisk/monitor/MIX/20111223-154839_1416431271)
[Dec 23 15:48:39] ERROR[12925]: utils.c:967 ast_carefulwrite: write() returned error: Broken pipe
[Dec 23 15:48:39] ERROR[12925]: utils.c:967 ast_carefulwrite: write() returned error: Broken pipe
[Dec 23 15:48:39] ERROR[12925]: utils.c:967 ast_carefulwrite: write() returned error: Broken pipe
[Dec 23 15:48:39] ERROR[12925]: utils.c:967 ast_carefulwrite: write() returned error: Broken pipe
[Dec 23 15:48:39] -- AGI Script agi-VDAD_ALL_outbound.agi completed, returning 0
[Dec 23 15:48:39] -- Executing [192*168*001*009*8366@default:1] Goto("SIP/hussain-0000001e", "default|8366|1") in new stack
[Dec 23 15:48:39] -- Goto (default,8366,1)
[Dec 23 15:48:39] -- Executing [8366@default:1] Playback("SIP/hussain-0000001e", "sip-silence") in new stack
[Dec 23 15:48:39] -- <SIP/hussain-0000001e> Playing 'sip-silence' (language 'en')
[Dec 23 15:48:39] -- Executing [8366@default:2] AGI("SIP/hussain-0000001e", "agi://127.0.0.1:4577/call_log") in new stack
[Dec 23 15:48:39] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Dec 23 15:48:39] -- Executing [8366@default:3] AGI("SIP/hussain-0000001e", "agi-VDAD_ALL_outbound.agi|SURVEYCAMP-----LB") in new stack
[Dec 23 15:48:39] -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi
[Dec 23 15:48:39] -- AGI Script agi-VDAD_ALL_outbound.agi completed, returning 0
[Dec 23 15:48:39] -- Executing [8366@default:4] AGI("SIP/hussain-0000001e", "agi-VDAD_ALL_outbound.agi|SURVEYCAMP-----LB") in new stack
[Dec 23 15:48:39] -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi
[Dec 23 15:48:39] -- AGI Script agi-VDAD_ALL_outbound.agi completed, returning 0
[Dec 23 15:48:39] -- Executing [8366@default:5] Hangup("SIP/hussain-0000001e", "") in new stack
[Dec 23 15:48:39] == Spawn extension (default, 8366, 5) exited non-zero on 'SIP/hussain-0000001e'
[Dec 23 15:48:39] -- Executing [h@default:1] DeadAGI("SIP/hussain-0000001e", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
[Dec 23 15:48:39] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Dec 23 15:48:39] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---5-----0 completed, returning 0
[Dec 23 15:48:39] == Spawn extension (default, 00441416431271, 2) exited non-zero on 'Local/00441416431271@default-0e25,2'
[Dec 23 15:48:40] == Manager 'sendcron' logged off from 127.0.0.1

My goal is to have a remote agent logged in and the call keeps alive for as long as possible. On the other end of the call, there is a recording. I just need the call to be alive once it is connected. Can you please help me with this scenario.

THanks

[williamconley]

ok, you're losing me ...

your first example shows a direct call from an agent to a prospect:

Code: Select all

Executing [00441416431271@default:2] Dial("SIP/cc101-00000022", "SIP/hussain/441416431271||Ttor")

which shows a call directly from 101 to a regular phone. Not a logged in agent call at all.

Now your new one shows an interesting mix of 8368 (the standard autodial extension) and 8366 (survey campaign, which is NOT a standard autodial extension) both in the same CLI. How are they related?

I am completely confused at this point.

Please describe the process you are attempting to achieve in detail, and please also verify that you HAVE gone through the Vicidial Manager's Manual and performed all the cool exercises to get your system online STANDARD before trying to do anything ... odd