vicidial.org

by **kaarrtti** » Tue May 31, 2011 9:27 pm

Goautodial 2.0
VERSION : 2.2.1-237
BUILD : 100510-201

Hi all experts,

i cant hear anything when dialing through agent screen. But fine through softphone. Is it codec problem, pls give solution.

Thanks in advance

Regds
kaarrtti

by **williamconley** » Thu Jun 02, 2011 12:09 am

Troubleshooting stage one: Show asterisk CLI from both types of calls.

Welcome to the party

by **fusionict** » Wed Jun 22, 2011 2:06 am

Hi William,

I've got the same error too..what seems to be the problem on this? I can't hear the phone ringing on the agent window. Then suddenly someone will answer the call.

Help would be much appreciated

by **cristian** » Wed Jun 22, 2011 3:38 am

We need to see what your dialer shows in the "asterisk -r" screen. Can ya'll post this for us?

Do you hear "you are the only person in this conference"?

by **fusionict** » Wed Jun 22, 2011 5:32 am

yup I do hear the "you are the only person in this conference" but after that when I hit the Dial Now it started dialing but I can't hear a thing. Then suddenly someone will answer the phone. Here is my asterisk CLI

--Created MeetMe conference 1023 for conference '8600051'
--<SIP/cc100-00000002> Playing 'conf-onlyperson' (language 'en')
==Manager 'sendcron' logged off from 127.0.0.1
==Parsing '/etc/asterisk/manager.conf'
==Manager 'sendcron' logged on from 127.0.0.1
--Executing [8600051@default:1] MeetMe("Local/8600051@default-94c6,2","*8600051:F") in new stack
>Channel Local/8600051@default-94c6,1 was answered.
--Executing [913145440590@default:1] AGI("Local/8600051@default-94c6,1","agi://127.0.0.1:4577/call_log") in new stack
--AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
--Executing [913145440590@default:2] Dial("Local/8600051@default-94c6,1","IAX2/fusionict/*13145440590::tTor") in new stack
--Called fusionict/*13145440590
--Call accepted by (my host IP) (format ulaw)
--Format for call is ulaw
--IAX2/fusionict-3152 is making progress passing it to Local/8600051@default-94c6,1
==Manager 'sendcron' logged off from 127.0.0.1
--IAX2/fusionict-3152 answered local/8600051@default-94c6,1

by **cristian** » Wed Jun 22, 2011 1:29 pm

So you are not hearing ring back (ring sounds) on manual dial then. Is that correct?

by **fusionict** » Thu Jun 23, 2011 12:48 am

Hi Cristian,

Yes, i cannot hear that the phone of the client is ringing.

by **sofcall** » Thu Jun 23, 2011 5:19 pm

GoAutoDial CE 2.0
VERSION: 2.2.1-237
BUILD: 100510-2015
from iso
single server.
no extra hardware.

same probleme here.don't hear the ringing tone till someone answers.
i do hear the message "you are the only person in this conference"

here is the cli.

[Jun 24 00:03:31] > Channel Local/8600051@default-c7f4,1 was answered.
[Jun 24 00:03:31] -- Executing [0471489449@default:1] AGI("Local/8600051@default-c7f4,1", "agi://127.0.0.1:4577/call_log") in new stack
[Jun 24 00:03:31] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Jun 24 00:03:31] -- Executing [0471489449@default:2] Dial("Local/8600051@default-c7f4,1", "SIP/siptrunk/0471489449||To") in new stack
[Jun 24 00:03:31] -- Called siptrunk/0471489449
[Jun 24 00:03:31] -- SIP/siptrunk-00000010 is ringing
[Jun 24 00:03:35] == Manager 'sendcron' logged off from 127.0.0.1
[Jun 24 00:03:48] -- SIP/siptrunk-00000010 answered Local/8600051@default-c7f4,1
[Jun 24 00:04:03] == Parsing '/etc/asterisk/manager.conf': [Jun 24 00:04:03] Found
[Jun 24 00:04:03] == Manager 'sendcron' logged on from 127.0.0.1
[Jun 24 00:04:03] == Parsing '/etc/asterisk/manager.conf': [Jun 24 00:04:03] Found
[Jun 24 00:04:03] == Manager 'sendcron' logged on from 127.0.0.1
[Jun 24 00:04:03] == Manager 'sendcron' logged off from 127.0.0.1
[Jun 24 00:04:06] == Parsing '/etc/asterisk/manager.conf': [Jun 24 00:04:06]

by **gardo** » Fri Jun 24, 2011 2:42 pm

When you're on auto-dial you're not suppose to hear the ringing part before the call gets answered. That defeats the purpose of a dialer. Answered calls are the ones that get passed on to the agent. Unless you're using analog lines for your trunks then you're not suppose to hear any ringing, busy, disconnected tones. These are automatically filtered out by the system.

by **cristian** » Fri Jun 24, 2011 2:48 pm

This is a conversation about manual dial.

by **sofcall** » Tue Jul 05, 2011 4:54 pm

thanx a lot.
but what about this issu .

when i statue the call using manial methode the button DIAL NEXT get inactive and i had to desconnect and connect again to receive a call.

cli when making manuel dial methode:

Connected to Asterisk 1.4.39.1-vici RPM by demian@goautodial.com currently running on go (pid = 8610)
Verbosity is at least 3
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing [8600051@default:1] MeetMe("Local/8600051@default-e6d9,2", "8600051|F") in new stack
-- Executing [0386577318@default:1] AGI("Local/8600051@default-e6d9,1", "agi://127.0.0.1:4577/call_log") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing [0386577318@default:2] Dial("Local/8600051@default-e6d9,1", "SIP/siptrunk/0386577318||To") in new stack
-- Called siptrunk/0386577318
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
-- SIP/siptrunk-0000002a is ringing
== 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
== Manager 'sendcron' logged off from 127.0.0.1
-- SIP/siptrunk-0000002a answered Local/8600051@default-e6d9,1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing [h@default:1] DeadAGI("Local/8600051@default-e6d9,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----37-----5") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... --37-----5 completed, returning 0
== Spawn extension (default, 0386577318, 2) exited non-zero on 'Local/8600051@default-e6d9,1'
== Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-e6d9,2'
-- Executing [h@default:1] DeadAGI("Local/8600051@default-e6d9,2", "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
***********************

by **cristian** » Tue Jul 05, 2011 4:59 pm

You should open a new thread.

by **cristian** » Tue Jul 05, 2011 5:34 pm

On this topic;

We resolved this issue by using inband ringing. If you want to try our solution PM me.

by **williamconley** » Tue Jul 05, 2011 7:33 pm

sofcall wrote:thanx a lot.
but what about this issu .

when i statue the call using manial methode the button DIAL NEXT get inactive and i had to desconnect and connect again to receive a call.

cli when making manuel dial methode:

you certainly should open a fresh thread with a fresh problem (and an appropriate subject line as well.

Can you verify this problem does or does not exist with the other manual dial methods on your installation?

Have you modified your installation at all?

Can you ALSO try this with the "standard" vicidial agent screen, which I believe Gardo has included in a different folder (agc2? something like that ...). This will confirm whether it is a configuration issue or a flaw in the "pretty" agent screen. If it is, it would be helpful for you to document it so Gardo can fix it for his next release.

by **dailafing** » Thu Nov 07, 2013 2:22 pm

I'm sorry, I cannot see how this issue has been resolved?

I'm in the same boat as OP; if I'm in manual dial (dialing single lines in order), or even if I hit the manual dial button while on auto dial, I don't hear any ringing at all. The only sound I hear is when an answer phone comes in or if the client picks up.

Also, an additional symptom is that once the customer picks up, the agent screen still says "waiting for ring"?
Eventually it says call failed.

I'm in the bus ATM so can't link cli or screen shot, but I'm hoping someone will explain how this thread is considered answered? And what the solution was...

Thanks

by **luckypig** » Fri Aug 22, 2014 4:36 am

I got the same error.

I can't hear ringback tone when I execute manual dial from agent web. My vicidial connect via SIP TRunk (ISP) Following:
1./ Press ring from web agent to call Phone, Phone answer, I hear "You are only the persion.."
2/ Press manual dial from web agent, fill phone number, press dial now. Although my mobile ringing, agent can't hear ringback tone.
3/ When I use phone to call mobile (not through web agent), This call is ok (hear ringback)

Can you help me

Thanks in advance

vicidial.org

Cant hear when dialing through agent screen.

Cant hear when dialing through agent screen.

Re: Cant hear when dialing through agent screen.

Re: Cant hear when dialing through agent screen.

Who is online