Ringing sound on manual dial

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Ringing sound on manual dial

Postby vctor » Tue Jul 03, 2007 11:07 am

I would like to have this feature. I am willing to help code it. I was thinking of checking on javascript if the status on the manager is still RINGING, if it is it should have maybe a Ringing sound or just a text that says the phone is ringing.

What do yuo think guys?
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Postby aster1 » Wed Jul 04, 2007 7:35 am

Do you basically want to hear ringing in manual dial ?? There's a easy way to do it i think . You can run Answer() first in dialplan before dialing destination then i think you will be able to hear a ringtone . Not sure if it would work properly . Doesn't agent hear ringing by default in manual dial ? ( I never used manual dial mode so not sure :D )
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Postby ramindia » Wed Jul 04, 2007 7:52 am

Hi Aster1

is this happends in manual dial ?

agent can hear ringing sound when the call happening ?

i have not heard anything like that, i only hear the direct voice
from customer, or Answering Machine.

is this happends in autodial ?

agent can hear ringing sound ?

clarify if any one can able to achieve this ?

ram
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Postby aster1 » Wed Jul 04, 2007 9:58 am

In auto-dial you never hear ringing sound . I think its same for manual dial i have not used manual dial so not sure . I had a carrier who used to send answer signal and then ringing started ( this is very bad for a carrier ) but due to this asterisk used to think call was picked up and used to transfer it to agent via agi script . Due to this agent was able to hear ringing sound but worst part was that all calls were marked as answered .. drop percentage went to huge numbers .. AMD always failed so i suggest this would be worst for auto-dial anyway what good does ringing sound do to a agent ?? For manual dial its a different story since theres no AMD or much of agi transfers involved but u will still have to bear ringing calls which no one will pickup .
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Postby mflorell » Thu Jul 05, 2007 10:48 am

You should hear a ringing sound on Manual Dial calls.

If you do not then it is probably a carrier issue.
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Postby tbenson » Thu Jul 12, 2007 12:58 pm

Ring is deffinately part of Manual Dial. Auto seperates you from wasting time that way, but manual just drops the call into the conference room when you click dial. Our small callcenters were using it to start as they were too used to hearing the ring and being involved in the hangups.
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my experience

Postby devafree » Thu Jul 12, 2007 2:17 pm

With all the ISPs in India, who also provide the minutes thru various 3rd party carriers, we hear the ringing tone for manual calls.
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Postby gardo » Fri Jul 13, 2007 2:45 am

What version of astguiclient do you hear the ringing tones? I'm still using 2.0.2 and I don't hear the ringing tones when doing manual calls through vicidial.
http://goautodial.com
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Postby vctor » Sat Jul 14, 2007 9:23 am

mflorell wrote:You should hear a ringing sound on Manual Dial calls.

If you do not then it is probably a carrier issue.


Matt,

If I use a softphone and do a manual call I hear a RINGBACK, why is it that I can not hear it when doing it on vici? Is it a telco issue?
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Postby ramindia » Sat Jul 14, 2007 10:22 am

Hi

I have tried more than 6 SIP carriers

I have not heard from single provider ring sounds when iam using
Vicidial in manual mode

but when i call from Xlite i can hear the ring sound

iam using 2.0.3

ram
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Postby devafree » Sat Jul 14, 2007 2:49 pm

hello

I am using 2.0.3b5 , asterisk 1.4 . I hear the ring on Manual mode on vicidial.

devafree
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Postby ramindia » Sat Jul 14, 2007 3:16 pm

Hi

can you post extension.conf

are you using by chance " answer" in dial plan ?

ram
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Postby prolixus » Mon Dec 10, 2007 11:29 pm

yo guys, regarding this, i tried calling our carrier/provider, everything is normal with their side, what do you think is the challenge why there is no ring or delays in rings?

for example:

i tried dialing a number, the call is incall but no ring at all, then my agents would try to hangup put in dispo then ready for next call, a sudden ring would be heard which was the previous call made. i have noticed that the conference is the one responsible for ending the call, as a solution we tried hanging up those calls in the conference, but still, on the next fresh call, the ring still no there but another scenario would rise which is somehow anooying because as the number is incall with no rings and almost a dial timeout, a person from the other line would speak up. this is a bit nice shall i say, but i dont think it is normal to wait for such a second of time without ringing anythin at all.

we are using manual dial method.

thanks!

prolixus
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Any Feedback on this POST

Postby mar_itsup » Tue Jan 06, 2009 6:58 pm

Hi Guys,

Any update on this post? I'm also looking forward for the best solutions you can provide. No Ringback status on VICI but in eyebeam its ok we hear ringback tone.


Thanks

Mar
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Postby mflorell » Tue Jan 06, 2009 7:18 pm

This doesn't seem to happen on all systems, and on systems where it does happen we have used the IAX loop trunk to answer the call and dial out over the outbound trunk before to get the ringing sound.

What version of Asterisk are you using?
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No Ringback Problem on VICI ( PROVIDER PROBLEM)

Postby mar_itsup » Tue Jan 06, 2009 10:23 pm

Hi Matt,

Asterisk 1.2.24 built on a x86_64 running on Debian Linux, we already found the solutions. It's the provider, they did some adjustments then afterwards we already received ringback.

Thanks Matt for the reply.

We confirmed that's its PROVIDER PROBLEM.

Thanks

Mar
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Postby benomg » Sun Apr 11, 2010 11:39 pm

I'm having trouble getting a "ringback" on manual calls with both my dev and live environments and with 2 separate carriers. I'm also having trouble convincing one carrier that there is possibly something they can help me with (Mytel btw) by way of a trunk configuration on their end. Is there a hint that I can give them as to what might be disabling ringback for the vicidial dialing system (not sure of the term here)?

Like others here, I get a ringback when dialing straight from the sip phone but not when on a agent interface "conference" (again, trouble with the term) which doesn't help the case of the problem lying with the carrier.
VERSION: 2.2.1-237 BUILD: 100510-2015 on Asterisk 1.2.30.4 via VICIBox 1.0.0-rc2 + upgrade.
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Postby gardo » Mon Apr 12, 2010 10:43 am

What kind of trunk are you using?
http://goautodial.com
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Postby benomg » Mon Apr 12, 2010 6:31 pm

It's an IAX2 trunk.

The dial plan is such.

exten => _X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _X.,2,Dial(IAX2/testtrunk/${EXTEN},,tTor)
exten => _X.,3,Hangup
VERSION: 2.2.1-237 BUILD: 100510-2015 on Asterisk 1.2.30.4 via VICIBox 1.0.0-rc2 + upgrade.
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Postby gmcust3 » Tue May 25, 2010 5:06 pm

GoAutoDial CE
VERSION: 2.4-309a
BUILD: 110430-1642
No other software installed on the box.
I've read the manager manual.
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Postby phil_discount » Mon Nov 08, 2010 3:47 pm

Hello,

the problem is still alive, i've got the same problem.
since saturday i use IAX2 to dial out and now the agents hearing the ringtone anymore.

Code: Select all
exten => _99999.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _99999.,2,Dial(IAX2/WAHLSERVER:test@ka-vici-dialer2/${EXTEN},35,tTo)
exten => _99999.,3,Hangup


is there a way to solve it?

regards
philip
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Postby gmcust3 » Mon Nov 08, 2010 10:08 pm

Instead of tTo , Try tToR.
GoAutoDial CE
VERSION: 2.4-309a
BUILD: 110430-1642
No other software installed on the box.
I've read the manager manual.
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Postby phil_discount » Tue Nov 09, 2010 1:35 am

i tried it but i it wont work. i'm using asterisk 1.2
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Postby mflorell » Tue Nov 09, 2010 6:45 am

try upgrading to Asterisk 1.4.21.2
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Postby phil_discount » Tue Nov 09, 2010 7:24 am

ok will be the next step.

in relation to performance: what would be better - asterisk 1.2 or 1.4?
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Postby mflorell » Tue Nov 09, 2010 7:26 am

1.4.21.2
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Postby phil_discount » Sat Nov 13, 2010 12:00 pm

i upgraded to vicibox redux 3.0.5 with asterisk 1.4.27
i tried also with "tTor"
but i get no ringing tone during manual call or callback

in CLI i find "is ringing"
perhaps the "call accepted" some lines before is the problem?

Code: Select all
[Nov 13 17:54:39]     -- Executing [8600051@default:1] MeetMe("Local/8600051@default-d1c8,2", "8600051|F") in new stack
[Nov 13 17:54:39]        > Channel Local/8600051@default-d1c8,1 was answered.
[Nov 13 17:54:39]     -- Executing [999990123456789@default:1] AGI("Local/8600051@default-d1c8,1", "agi://127.0.0.1:4577/call_log") in new stack
[Nov 13 17:54:40]     -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Nov 13 17:54:40]     -- Executing [999990123456789@default:2] Dial("Local/8600051@default-d1c8,1", "IAX2/WAHLSERVER:test@192.168.203.33/999990123456789|35|tTo") in new stack
[Nov 13 17:54:40]     -- Called WAHLSERVER:test@192.168.203.33/999990123456789
[Nov 13 17:54:40]     -- Call accepted by 192.168.203.33 (format ulaw)
[Nov 13 17:54:40]     -- Format for call is ulaw
[Nov 13 17:54:43]   == Manager 'sendcron' logged off from 127.0.0.1
[Nov 13 17:54:46]     -- IAX2/WAHLSERVER-9775 is ringing
[Nov 13 17:54:49]     -- IAX2/WAHLSERVER-9775 stopped sounds
[Nov 13 17:54:49]     -- IAX2/WAHLSERVER-9775 answered Local/8600051@default-d1c8,1
[Nov 13 17:54:53]   == Parsing '/etc/asterisk/manager.conf': [Nov 13 17:54:53] Found
[Nov 13 17:54:53]   == Manager 'sendcron' logged on from 127.0.0.1
[Nov 13 17:54:53]   == Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-d1c8,2'
[Nov 13 17:54:53]     -- Executing [h@default:1] DeadAGI("Local/8600051@default-d1c8,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Nov 13 17:54:53]     -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0
[Nov 13 17:54:53]     -- Executing [h@default:1] DeadAGI("Local/8600051@default-d1c8,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----14-----4") in new stack
[Nov 13 17:54:53]     -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----14-----4 completed, returning 0
[Nov 13 17:54:53]     -- Hungup 'IAX2/WAHLSERVER-9775'
[Nov 13 17:54:53]   == Spawn extension (default, 999990123456789, 2) exited non-zero on 'Local/8600051@default-d1c8,1'


extension
Code: Select all
;Verbindung IAX Gateway ka-vici-dialer2 - Extern
exten => _9999[1-9].,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _9999[1-9].,2,Dial(IAX2/WAHLSERVER:test@192.168.203.33/${EXTEN},35,tTo)
exten => _9999[1-9].,3,Hangup
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Postby covarrubiasgg » Tue Nov 16, 2010 1:02 pm

gmcust3 wrote:Instead of tTo , Try tToR.


This works for me using vicidial VERSION: 2.4-283 BUILD: 100929-1203 on asterisk 1.4.27.1-vici from vicibox redux 3.0.6.

Just to let you know, thanks!
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Postby phil_discount » Tue Nov 16, 2010 5:54 pm

i tried to call over my iax gateway which is using a SIP provider and i get no ringback...
then i changed progressinband=yes in sip.conf on iax gateway and i get ringback dialing out over SIP and manual call.

-> problem is not iax
-> problem is my dahdi configuration for pbx
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Postby phil_discount » Sat Nov 20, 2010 10:24 am

any idea what i can change in dahdi configuration?
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Postby phil_discount » Thu Nov 25, 2010 12:21 pm

ok i found the problem

with tToR it works, but not every time if i call a mobile phone.
if i call a normal analog/digital phone, i allways hear ringing....

on all my tests i use my mobile phone...so that was the problem

thanks all!
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Re: Ringing sound on manual dial

Postby yuvrajkc » Thu Feb 07, 2013 5:04 am

An easy way would be to update the dial command:

Code: Select all
exten => _981NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _981NXXNXXXXXX,n,Dial(SIP/CarrierName/${EXTEN:2},,tTorL(300000)m(rng))
exten => _981NXXNXXXXXX,n,Hangup


Next create a musicOnHold with name "rng".
Place a ringing sound under that MOH.

I hope it makes sense and helps someone :)
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