Throw a Call from one server to another server

[udy786]

Hi,

I have two vicidial server. Both server located on different location on public IP. In current setup, We have configured a IVR on server A. Now point is, when customer will press 1 then we want to throw that call on server B in any In-group. We have to throw that call (1 Pressed) on server B because that Campaign running on server B.

Here is flow of my setup on server A:-

1. Call received on Inbound number 123-456-7890 (DID)
2. In DID, we have called Call Menu
3. in Call Menu we have configured as per user Input.

From server A, need to throw call on server B when will receive 1 DTMF from user end.

Bother server Details:-
Asterisk 1.4.39.1-vici
VERSION: 2.4-309a
BUILD: 110430-1642
Installed from GoAutodial CD 2.1.


Please suggest or guide me.

[striker]

1. interconnect two server via sip trunk or iax trunk
link : http://www.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/connecting_two_asterisk.html

2. create a DID in server B in admin
3. dial that DID @ sipserverB in the call menu ( proper dialplan to dial that DID via the serverB sip trunk)

[udy786]

We have same setting using DID. Like we have dialed a DID from server A when customer will press 1 and same DID we have configured on server B. Its working.

But its making double cost because dialing again a DID on press 1 . We want to dial or call any dialplan / agi / script from server A or B so that call will go directly on server B. Using this we can save a DID costing and per call charge. Its not only for money, it will increase connectivity time because if will dial any new DID then again it will take 4 or 5 sec. so we can save this 4 to 5 sec. time also.

[williamconley]

Set up Server A to have Server B as a "Carrier" under Admin->Servers. Use the credentials of a Phone from server B. So Server A will then be a "phone" on server B and can call just like any other phone on server B. Now you just need to make that phone call to "trunkinbound" so its calls will be treated like a normal DID. This is done by modifying the phones "Context" entry in server B to "trunkinbound".

Now when you use server A to call to server B to that DID, it will actually be a direct call between the servers and land in the DID just like it came from a telco. But YOU are the telco. No money changed hands.

Not only that, but the DID can be any number you like (71523, 9999, whatever). Since it is a direct call between two servers you control, you'll even be able to troubleshoot easier because you can access the asterisk command line of both servers until it works. And you don't need to stop dialing with your regular method to set it up. Switch over when you've got it figured out without even stopping the campaign. Done it lots of times.

[udy786]

Hi,
I tried to configure as you guided me. Now calls going on server B. Also its going in DID and Inbound group. When my agents are not logged on, when we are getting message as we have configured in Inbound Group for that DID. "No One available to take your calls". But problem is when agents are logged on for that Inbound Group then calls landing on agents screen (Agents of server B) but not getting any voice. At same time on server A, agents getting busy tone and on CLI of server getting CHANUNAVAIL.

A have configured SIP on server A with name of "uday" and dialplan 977. All dialed with 977 prefix will go on server B. On server B, a DID 9377579349 created. So from server A, I have dialed 9377579349 from agent interface. In Campaing, dial prefix is 977. Using this process, call gone on server B.

Server A Dialplan:-

Code: Select all

exten => _977.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _977.,n,Dial(SIP/${EXTEN:4}@uday,,tTo)
exten => _977.,n,Hangup

Server A SIP:-

Code: Select all

[uday]
username=uday
type=peer
qualify=1000
insecure=very
port=5060
host=192.168.1.208
dtmfmode=rfc2833
fromdomain=uday
canreinvite=no
disallow=all
allow=alaw
trustrpid=yes
sendrpid=yes
insecure=invite

On server B, call received in DID with exten => 9377579349,1,AGI(agi-DID_route.agi).

Server A CLI:-

Code: Select all

go*CLI>
    -- Executing [97719377579349@default:1] AGI("SIP/8002-0000017c", "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 [97719377579349@default:2] Dial("SIP/8002-0000017c", "SIP/9377579349@uday||tTo") in new stack
    -- Called 9377579349@uday
    -- SIP/uday-0000017d answered SIP/8002-0000017c
    -- Executing [h@default:1] DeadAGI("SIP/8002-0000017c", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----9-----9") in new stack
    -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----9-----9 completed, returning 0
  == Spawn extension (default, 97719377579349, 2) exited non-zero on 'SIP/8002-0000017c'
go*CLI>


Server B CLI:-

Code: Select all

[Jun 21 13:54:12]     -- Executing [9377579349@default:1] AGI("SIP/udayi-000000e5", "agi-DID_route.agi") in new stack
[Jun 21 13:54:12]     -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-DID_route.agi
[Jun 21 13:54:12]     -- AGI Script Executing Application: (Monitor) Options: (wav|/var/spool/asterisk/monitor/MIX/20140621135412_9377579349_0000000000)
[Jun 21 13:54:12]     -- AGI Script agi-DID_route.agi completed, returning 0
[Jun 21 13:54:12]     -- Executing [99909*4***DID@default:1] Answer("SIP/udayi-000000e5", "") in new stack
[Jun 21 13:54:12]     -- Executing [99909*4***DID@default:2] AGI("SIP/udayi-000000e5", "agi-VDAD_ALL_inbound.agi") in new stack
[Jun 21 13:54:12]     -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_inbound.agi
[Jun 21 13:54:13]     -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Jun 21 13:54:13]     -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Jun 21 13:54:13]   == Parsing '/etc/asterisk/manager.conf': [Jun 21 13:54:13] Found
[Jun 21 13:54:13]   == Manager 'sendcron' logged on from 127.0.0.1
[Jun 21 13:54:13]     -- Executing [192*168*001*208*78600051@default:1] Set("Local/192*168*001*208*78600051@default-edf1,2", "extension=192*168*001*208*78600051") in new stack
[Jun 21 13:54:13]     -- Executing [192*168*001*208*78600051@default:2] Dial("Local/192*168*001*208*78600051@default-edf1,2", "SIP/192*168*001*208*78600051||M(recording|192*168*001*208*78600051|||)") in new stack
[Jun 21 13:54:13]   == Manager 'sendcron' logged off from 127.0.0.1
[Jun 21 13:54:13]   == Everyone is busy/congested at this time (1:0/0/1)
[Jun 21 13:54:13]     -- Executing [h@default:1] DeadAGI("Local/192*168*001*208*78600051@default-edf1,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----20-----CHANUNAVAIL----------") in new stack
[Jun 21 13:54:13]     -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----20-----CHANUNAVAIL---------- completed, returning 0
[Jun 21 13:54:14]     -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Jun 21 13:54:14]     -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Jun 21 13:54:14]   == Parsing '/etc/asterisk/manager.conf': [Jun 21 13:54:14] Found
[Jun 21 13:54:14]   == Manager 'sendcron' logged on from 127.0.0.1
[Jun 21 13:54:14]     -- Executing [58600051@default:1] MeetMe("Local/58600051@default-4c2e,2", "8600051|Fmq") in new stack
[Jun 21 13:54:14]        > Channel Local/58600051@default-4c2e,1 was answered.
[Jun 21 13:54:14]     -- Executing [8309@default:1] Answer("Local/58600051@default-4c2e,1", "") in new stack
[Jun 21 13:54:14]     -- Executing [8309@default:2] Monitor("Local/58600051@default-4c2e,1", "wav|20140621-232413_0000000000") in new stack
[Jun 21 13:54:14]     -- Executing [8309@default:3] Wait("Local/58600051@default-4c2e,1", "3600") in new stack
[Jun 21 13:54:14]     -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Jun 21 13:54:14]     -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Jun 21 13:54:14]     -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Jun 21 13:54:14]     -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Jun 21 13:54:14]     -- AGI Script agi-VDAD_ALL_inbound.agi completed, returning 0
[Jun 21 13:54:14]     -- Executing [192*168*001*208*8600051@default:1] Set("SIP/udayi-000000e5", "extension=192*168*001*208*8600051") in new stack
[Jun 21 13:54:14]     -- Executing [192*168*001*208*8600051@default:2] Dial("SIP/udayi-000000e5", "SIP/192*168*001*208*8600051||M(recording|192*168*001*208*8600051|||)") in new stack
[Jun 21 13:54:15]   == Manager 'sendcron' logged off from 127.0.0.1
[Jun 21 13:54:15]   == Everyone is busy/congested at this time (1:0/0/1)
[Jun 21 13:54:15]   == Auto fallthrough, channel 'SIP/udayi-000000e5' status is 'CHANUNAVAIL'
[Jun 21 13:54:16]   == Manager 'sendcron' logged off from 127.0.0.1
[Jun 21 13:54:21]     -- Executing [h@default:1] DeadAGI("SIP/udayi-000000e5", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----20-----CHANUNAVAIL----------") in new stack
[Jun 21 13:54:21]     -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----20-----CHANUNAVAIL---------- completed, returning 0
go*CLI>


Please let me if I am doing anything wrong or did any mistake.

[williamconley]

No sound is usually a firewall issue. Are these servers on the same physical network?

[udy786]

Yes both server in same network.