Inbound Calls

[chill_master]

Hi,

I was able to setup VicidialNow a long time ago. My outbound call is doing fine. But I can't setup my incoming calls. I don't know how to set a dial plan to forward the incoming call to the DID I created. I followed the tutorial in Managers Manual, but there's no item there that discusses about the dial plan.

Can anybody point me to a right direction?

[Op3r]

reread the manager's manual?

Its all there. as for 2.0.5 its now an asterisk configurator

[chill_master]

I got it now... I totally did not check with my VOIP provider about how many digits they send.

I was able to receive already an incoming call.

Thank you so much Op3r. Probably I was just too anxious about it.

By the way, it wasn't mentioned in the manual on how the vicidial handles the call when there are more than 1 agent who is online and part of the group that is suppose to take the incoming calls. Does vicidial ring all agents? Does it find which agent is free if some are taking some calls?

[gardo]

A campaign where all agents are logged-in does the ACD (automatic call distribution). Depending on your campaign settings, it will find the next agent who's free. However, it won't ring any agents at all. The behavior of inbound campaigns is almost the same as outbound campaigns.

[oshonubi]

Hello,

Have similar challenges based on PSTN provider. I have a nmuber from the provider, I tried it but never worked. Is there any difference in the configuration if SIP and ZAP. My PSTN provider gave me 10 digits and I added it through the DID but not sure of what the dial plan should.

The number is (0)7098141858, while the country code is 234 . The initMy confusion so far has been between the DID and the dial plan. Which one of them will affect the incoming call.

Again, is there any difference in the way the dial plan is configure between Zap and SIP/IAX. Zap does not have any authentication and codec issue.

[gardo]

The dialplan is almost the same whether you're using SIP/IAX or Zap. You just need to configure Asterisk first for your PSTN lines. Make sure everything works properly with your Asterisk setup. You can post the output of your Asterisk CLI when dialing.

[oshonubi]

Hi Gardo,

Thanks for the response. I believe my PSTN line is working. This is because the call routes to the softphone. However, please find attached the following information:

My dial plan setting:

exten => s,1,AGI(agi://127.0.0.1:4577/call_log)
exten => s,2,Dial(SIP/201,20,to )
exten => s,3,Hangup

when I used other settings, I got "Sorry that is not a valid extension"

The other settings are

exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91NXXNXXXXXX,2,Dial(SIP/201,20,to )
exten => _91NXXNXXXXXX,3,Hangup


exten => _X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _X.,2,Dial(SIP/201,20,to)
exten => _X.,3,Hangup

exten => _9234XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _9234XXXXXXXXXX,2,Dial(SIP/201,20,to)
exten => _9234XXXXXXXXXX,3,Hangup

The 234 is my country code, while the remaining digits are as provided by the providers.

This is my cli output

Aug 21 14:30:42 NOTICE[14166]: chan_zap.c:6248 ss_thread: Got event 18 (Ring Begin)...
Aug 21 14:30:43 NOTICE[14166]: chan_zap.c:6248 ss_thread: Got event 2 (Ring/Answered)...


Again I tried it using my pbx line, I did not get any result apart from using the following dial plan:

exten => s,1,AGI(agi://127.0.0.1:4577/call_log)
exten => s,2,Dial(SIP/201,20,to )
exten => s,3,Hangup

My questions are:

1. Does DID configuration have any effect on receiving calls?

2. Does dial plan have anything to do with the leads in the list for this to work well?

Will really appreciate your response.

[gardo]

1. Yes it does.If the dialplan for your DID is not configured properly, Asterisk won't accept those calls.

2. Yes. Dialing the numbers on your list without the correct dialplan won't work. The system will dial the numbers. However, it won't connect your calls and you'll get some error messages on your CLI.

My questions are:

1. Does DID configuration have any effect on receiving calls?

2. Does dial plan have anything to do with the leads in the list for this to work well?

Will really appreciate your response.

[oshonubi]

Hi Gardo,

Thanks for the response. Based on my dial plan, I have the "LIVE CALL" up now. However there is a continuous loop in the dial. There is a continuous incoming call to the agent interface. Unfortunately the call is not a real call. It is a sort of simulated call from vicidial itself. What I mean by that is that the call seems to be from one of the leads in the list, however, there is no "valid" PSTN line connected, rather it was my PBX that was connected to the vicidial system. Please kindly advice on what can be done on this.

Whenever, I use the vicidial dial plan below,

exten => _9234XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _9234XXXXXXXXXX,2,Dial(SIP/201,20,to)
exten => _9234XXXXXXXXXX,3,Hangup

and I tried to dial into it, I get "Sorry, that is not a valid extension"

However, whenever I use this other one below,

exten => s,1,AGI(agi://127.0.0.1:4577/call_log)
exten => s,2,Dial(SIP/201,20,to )
exten => s,3,Hangup

My dial-in goes to the softphone not the agent interface.

Kindly assist.

[chill_master]

that is because of this:

exten => s,2,Dial(SIP/201,20,to )

it dials the extension used by the softphone.

I suggest follow carefully the managers manual. In my case, I didn't add anything in my dial plan.

just make sure of the DID.