3 Way Hangup

[bghayad]

Hello;

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, Vtiger V5

The call come to the agent, the agent do a transfer for the call using the SIP trunk to a destination mobile number, the problem that if the agent click on the Hangup button at the goautodial (after the transfer completed and both sides are talking) the call will be disconnected. What we need is the call to stay between the caller and the destination if the agent completed the transfer even if the agent clicked on the hangup button. At the phone, the call is finished (no more appearing, but at the agent screen, still the hangup button need to be clicked and that will cause the transferred call to be disconnected when clicking on that button). How we can fix this?

At the same time, I would that a hangup to be happen at the agent screen after hangup on the IP Phone device it self, but I am not able to know from where it can be done .. Although, I do not need the transfer to be disconnected if we click on the hangup button at the agent desktop.

Regards
Bilal

[williamconley]

there is a "leave 3-way call" button in the transfer-conf popup. this is used to allow the agent to continue to another call while leaving the Prospect and 3rd party together in the conference room.

[bghayad]

So the only way is to be done from the agent desktop goautodial screen, not directly from the IP Phone?
Regards
Bilal

[williamconley]

So the only way is to be done from the agent desktop goautodial screen, not directly from the IP Phone?
Regards
Bilal

Unless you use "remote agent", yes. Remote Agents, however, are not able to put information into the agent screen (because they don't have one ...). Although there is a "remote agent interface" which allows remote agents to log in and modify the client record after the fact. I believe the file is "vdremote.php" in the /agc/ folder.

[bghayad]

About using transfer via the goautodial desktop application, there are two issues:

1) When I do it from the agent desktop and I select consultative (because I need to transfer the caller from his mobile to a destination mobile or landline but not internal extension), I see the value of the channel that is IAX/127.0.0.1 which is correct. How I can correct the channel to use the right channel? From where it is taking IAX/127.0.0.1?

2) I think I have to put values in the Transfer-Conf DTMF 1 and Transfer-Conf Number 1 and others to be able to use the transfer. I am not able to know what these values need to be and if it is required to be able to do the transfer via the goautodial desktop application. Any advise?

Regards
Bilal

[williamconley]

About using transfer via the goautodial desktop application, there are two issues:

1) When I do it from the agent desktop and I select consultative (because I need to transfer the caller from his mobile to a destination mobile or landline but not internal extension), I see the value of the channel that is IAX/127.0.0.1 which is correct. How I can correct the channel to use the right channel? From where it is taking IAX/127.0.0.1?

2) I think I have to put values in the Transfer-Conf DTMF 1 and Transfer-Conf Number 1 and others to be able to use the transfer. I am not able to know what these values need to be and if it is required to be able to do the transfer via the goautodial desktop application. Any advise?

Regards
Bilal

you do not modify the channel, you follow the directions in the agent and/or manager's manual for transfers. none of which involve modifying the channel value

you do not need to use DTMF or Number values, those are shortcuts (which can be modified in the Campaign settings, but are not required)

[bghayad]

Hello;

Is it required to add CXFER in Transfer-Conf Number 1 in Campaign (D1)?
Anything related to Campaign VDAD extension need to be done? How?

There is a setting need to be done on the vicidial admin to be able to do the 3 WAY transfer .. meanwhile, it is not working.
I selected Transfer, then I typed the telephone number that I need to transfer the call for it and then I clicked on Dial With Customer, but it is not calling the customer and I am getting the followig:

go*CLI>
go*CLI>
[Apr 22 09:30:09] == Parsing '/etc/asterisk/manager.conf': [Apr 22 09:30:09] Found
[Apr 22 09:30:09] == Manager 'sendcron' logged on from 127.0.0.1
[Apr 22 09:30:09] -- Executing [950911347@default:1] AGI("Local/950911347@default-741a,2", "agi://127.0.0.1:4577/call_log") in new stack
[Apr 22 09:30:09] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Apr 22 09:30:09] -- Executing [950911347@default:2] Dial("Local/950911347@default-741a,2", "SIP/50911347@kems||tTo") in new stack
[Apr 22 09:30:09] -- Called 50911347@kems
[Apr 22 09:30:10] -- Got SIP response 480 "Temporarily Unavailable" back from 80.184.248.36
[Apr 22 09:30:10] -- SIP/kems-000004c5 is circuit-busy
[Apr 22 09:30:10] == Everyone is busy/congested at this time (1:0/1/0)
[Apr 22 09:30:10] -- Executing [950911347@default:3] Hangup("Local/950911347@default-741a,2", "") in new stack
[Apr 22 09:30:10] == Spawn extension (default, 950911347, 3) exited non-zero on 'Local/950911347@default-741a,2'
[Apr 22 09:30:10] -- Executing [h@default:1] DeadAGI("Local/950911347@default-741a,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----19-----CONGESTION----------") in new stack
[Apr 22 09:30:10] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Apr 22 09:30:10] == Refreshing DNS lookups.
[Apr 22 09:30:12] == Manager 'sendcron' logged off from 127.0.0.1

Regards
Bilal

[williamconley]

what number are you actually trying to dial? if it is 950911347, the "9" is being stripped off as your dial prefix to choose a carrier. Add another 9 to the front 9950911347. Perhaps it would help if you showed a successful call to a regular number so we can see your normal dial pattern.

[Apr 22 09:30:09] -- Executing [950911347@default:2] Dial("Local/950911347@default-741a,2", "SIP/50911347@kems||tTo") in new stack
[Apr 22 09:30:09] -- Called 50911347@kems

[bghayad]

By the way: What is the story of the Local in the line (because I do not see the Local if I am doing a normal outside call directly from the IP Phone)? Does it effect?

[Apr 22 09:30:09] -- Executing [950911347@default:2] Dial("Local/950911347@default-741a,2", "SIP/50911347@kems||tTo") in new stack

Also I need to notify that the problem that the call is failing from the service provider "Got SIP response 480 Temporarily Unavailable", but if I do a transfer via the IP Phone directly, the call is not drop and it is going fine ... I feel there is a setting or something need to be allowed or enabled to be able to do this transfer from the desktop, but what this?

[Apr 22 09:30:10] -- Got SIP response 480 "Temporarily Unavailable" back from 80.184.248.36
[Apr 22 09:30:10] -- SIP/kems-000004c5 is circuit-busy

I am trying to call 50911347 and I added the 9 because when we do normally any outside call, we put 9 in the beginning of the number. The context vicidial-auto is included in [default] context and the pattern as below:

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

; Local Server: 10.149.1.130
exten => _010*149*001*130*.,1,Goto(default,${EXTEN:16},1)
; VICIDIAL Carrier: Kems - Kems
; Signup an account at http://goautodial.com/voip-store/
exten => _9NXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _9NXXXXXXX,2,Dial(SIP/${EXTEN:1}@kems,,tTo)
exten => _9NXXXXXXX,3,Hangup


Regards
Bilal

[williamconley]

you may be experiencing a callerid caused hangup or any one of several other issues.

you should try a sip-debug and see if you can detect a difference in the invite packet sent to request this call and a regular call that succeeds.

and change your 3-way call callerid on the campaign, too.

[bghayad]

I think you are right .. it is something related to the provider authentication in case of 3 Way transfer.

I will tell you what is the story, to be able to place outside call, it was required to send the callerid (the callerid of the phone extension) to be callerid="22281258" <22281258> and I putted this in the extension configuration to be able to place the outside call.

The question is: How this can be done in the 3 Way transfer specifically that it is using the term Local as shown in the sent logs .. From where to send the callerid and the fromuser parameters of vicidial to be able to do the 3 Way transfer call which will place outside call?

Regards
Bilal

[bghayad]

How I can set the callerid="22281258" <22281258> for the dialer?
In other words, the difference between the transfer via the IP Phone and the transfer via the 3 Way transfer that is existed on the application that the 3 Way transfer is using the Local while the transfer from the IP Phone is using the IP Phone it self.

This is the result if I am doing the transfer from the IP Phone:

[Apr 22 09:32:18] -- Executing [950911347@default:2] Dial("SIP/501-000004c6", "SIP/50911347@kems||tTo") in new stack
[Apr 22 09:32:18] -- Called 50911347@kems
[Apr 22 09:32:21] -- SIP/kems-000004c7 is ringing

This is the result when i am doing the transfer from the application using the 3 Way transfer:

[Apr 22 09:30:09] -- Executing [950911347@default:2] Dial("Local/950911347@default-741a,2", "SIP/50911347@kems||tTo") in new stack
[Apr 22 09:30:09] -- Called 50911347@kems
[Apr 22 09:30:10] -- Got SIP response 480 "Temporarily Unavailable" back from 80.184.248.36
[Apr 22 09:30:10] -- SIP/kems-000004c5 is circuit-busy

At the IP Phone, I am using the following to be able to place outside call (the callerid="22281258" <22281258> is very important to be set, otherwise my provider will reject the call), so how I can set the caller for the dialer in case I am using 3 Way transfer from the application, so my provider will authenticate me?

[501]
username=501
secret=*********
accountcode=501
callerid="22281258" <22281258>
mailbox=501
context=default
disallow=all
allow=ulaw
type=friend
host=dynamic

Regards
Bilal

[williamconley]

Use SIP debug to see the technical differences in the calls.

You cannot modify the "Callerid NAME" in a Vicidial call unless you use the loopback dial method, which is complex. Callerid name is used by Vicidial to control/manage the calls (since callerid name is not functional inside the US and this system was built at a time when that was the only guaranteeed "still there" variable available).

[bghayad]

Actually if I remove the callerid="22281258" <22281258> in the extension configuration, I will get the same error in the call:

[Apr 22 09:30:10] -- Got SIP response 480 "Temporarily Unavailable" back from 80.184.248.36
[Apr 22 09:30:10] -- SIP/kems-000004c5 is circuit-busy

I would to know, what is the meaning of Local and the meaning of the 2 in this line

"Local/950911347@default-741a,2"

Can you tell me william something:

In the 3 Way transfer using the application, and when clicking on Dial with customer: in this case, the call is considered to be generated from the dialer and not the phone, right? So that is the mean of the Local?

I am sure that the callerid is making the problem .. But, how to resolve it?

Regards
Bilal

[williamconley]

use sip debug. you'll get real details of the technicalities of the call. sip headers especially are quite often the culprit in these situations.

[bghayad]

Hello William;

Manual Dial and 3 Way transfer are giving the same result and in both situation, it seems the problem is related to the Remote-Party-ID as I did a lot of attempts ... Also, I know that my carrier is requesting from me to put the callerid of the IP Phone to by my primary DID number and that is what I did in the Phone configuration (callerid=22281258) otherwise the IP Phones it self will not work and will give the same error. I put the Campaign Caller ID to be 22281258 but still the Remote-Party-ID is not giving the same result what it is giving when calling from the IP Phone (because when I am calling from the IP Phone, I see the Remote-Party-ID is the 22281258). Kindly find below the sip debug from the asterisk CLI (this for manual transfer and same thing for the 3 Way transfer). I believe if I can place manual outbound call (from the application it self), then I will be able to do 3 Way transfer.

[May 6 13:15:03] --- (8 headers 0 lines) ---
[May 6 13:15:03] -- Got SIP response 480 "Temporarily Unavailable" back from 80.184.248.36
[May 6 13:15:03] Transmitting (NAT) to 80.184.248.36:5060:
ACK sip:50911347@80.184.248.36;cpd=on SIP/2.0
Via: SIP/2.0/UDP 10.149.1.130:5060;branch=z9hG4bK21ca309c;rport
From: "DV860501W000000t001W" <sip:1831100@10.100.5.125>;tag=as28945553
To: <sip:50911347@80.184.248.36;cpd=on>;tag=1524663509
Contact: <sip:1831100@10.100.5.125>
Call-ID: 75e547f31e9b4f8348108d0749e2a2a0@10.100.5.125
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "DV860501W000000t001W" <sip:0000000000@10.149.1.130>;privacy=off;screen=no
Content-Length: 0


---
[May 6 13:15:03] -- SIP/kems-000000c8 is circuit-busy
[May 6 13:15:03] == Everyone is busy/congested at this time (1:0/1/0)
[May 6 13:15:03] -- Executing [950911347@default:3] Hangup("Local/8600051@default-1123,1", "") in new stack
[May 6 13:15:03] == Spawn extension (default, 950911347, 3) exited non-zero on 'Local/8600051@default-1123,1'
[May 6 13:15:03] -- Executing [h@default:1] DeadAGI("Local/8600051@default-1123,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----19-----CONGESTION----------") in new stack
[May 6 13:15:03] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[May 6 13:15:03] Really destroying SIP dialog '75e547f31e9b4f8348108d0749e2a2a0@10.149.1.130' Method: INVITE
[May 6 13:15:03] == Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-1123,2'
[May 6 13:15:03] -- Executing [h@default:1] DeadAGI("Local/8600051@default-1123,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----N

Regards
Bilal

[williamconley]

how was this sample call generated? from a phone in a logged in agent session manual dial button? with or without override? or was it straight from the keypad of the phone through asterisk to the carrier?

have you modified the callerid of the phone in admin->phones?

[bghayad]

how was this sample call generated? from a phone in a logged in agent session manual dial button?

Yes.

with or without override?

Yes.

or was it straight from the keypad of the phone through asterisk to the carrier?

No.

have you modified the callerid of the phone in admin->phones?

Yes, I set Outbound CallerID: 22281258

Appreciate the help please.

Regards
Bilal