HELP.... DID ringing on the phone used to dial the number

[gtalbos]

Hi,

just a newbie here.
we were ale to setup vicibox redux 4.0.3; Vicidial VERSION: 2.8-407a BUILD: 130709-1350 from; Asterisk 1.4.44 from scratch. I'm also using eyebeam softphone 1.1 3007n stamp 17816.

We were able to setup carriers, campaigns, phones, and users. Outbound calls are doing just fine. We can make calls to vonage, mobiles and landlines.

We are now trying to setup our inbound calls. We've made inbound campaign, in group, and DID. But when we tried to call our DID number, "its ringing on the phone that we used to dial the DID, but there's nothing happened on vicidial".

Please help...

CARRIER SETUP

Account entry

[Provider]
type=friend
host=xx.xx.xx.xx
canreinvite=no
disallow=all
allow=g729
allow=ulaw
dtmfmode=rfc2833
qualify=yes
nat=yes
context=trunkinbound

Dial plan entry

exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91NXXNXXXXXX,n,Dial(SIP/${EXTEN:1}@provider,,tTor)
exten => _91NXXNXXXXXX,n,Hangup


CLI notes upon login to inbound campaign

[Aug 12 14:59:17] == Parsing '/etc/asterisk/manager.conf': [Aug 12 14:59:17] Found
[Aug 12 14:59:17] == Manager 'sendcron' logged on from 127.0.0.1
[Aug 12 14:59:20] > Channel SIP/zerone-00000272 was answered.
[Aug 12 14:59:20] -- Executing [8600053@default:1] MeetMe("SIP/zerone-00000272", "8600053|F") in new stack
[Aug 12 14:59:20] == Parsing '/etc/asterisk/meetme.conf': [Aug 12 14:59:20] Found
[Aug 12 14:59:20] == Parsing '/etc/asterisk/meetme-vicidial.conf': [Aug 12 14:59:20] Found
[Aug 12 14:59:20] -- Created MeetMe conference 1023 for conference '8600053'
[Aug 12 14:59:20] -- <SIP/zerone-00000272> Playing 'conf-onlyperson' (language 'en')
[Aug 12 14:59:21] == Manager 'sendcron' logged off from 127.0.0.1
vicibox*CLI> [Aug 12 14:59:17] == Parsing '/etc/asterisk/manager.conf': [Aug 12 14:59:17] Found
No such command '[Aug 12 14:59:17] == Parsing '/etc/asterisk/manager.conf': [Aug 12 14:59:17] Found' (type 'help [Aug 12' for other possible commands)
vicibox*CLI> [Aug 12 14:59:17] == Manager 'sendcron' logged on from 127.0.0.1
No such command '[Aug 12 14:59:17] == Manager 'sendcron' logged on from 127.0.0.1' (type 'help [Aug 12' for other possible commands)
vicibox*CLI> [Aug 12 14:59:20] > Channel SIP/zerone-00000272 was answered.
No such command '[Aug 12 14:59:20] > Channel SIP/zerone-00000272 was answered.' (type 'help [Aug 12' for other possible commands)
vicibox*CLI> [Aug 12 14:59:20] -- Executing [8600053@default:1] MeetMe("SIP/zerone-00000272", "8600053|F") in new stack
No such command '[Aug 12 14:59:20] -- Executing [8600053@default:1] MeetMe("SIP/zerone-00000272", "8600053|F") in new stack' (type 'help [Aug 12' for other possible commands)
vicibox*CLI> [Aug 12 14:59:20] == Parsing '/etc/asterisk/meetme.conf': [Aug 12 14:59:20] Found
No such command '[Aug 12 14:59:20] == Parsing '/etc/asterisk/meetme.conf': [Aug 12 14:59:20] Found' (type 'help [Aug 12' for other possible commands)
vicibox*CLI> [Aug 12 14:59:20] == Parsing '/etc/asterisk/meetme-vicidial.conf': [Aug 12 14:59:20] Found
No such command '[Aug 12 14:59:20] == Parsing '/etc/asterisk/meetme-vicidial.conf': [Aug 12 14:59:20] Found' (type 'help [Aug 12' for other possible commands)
vicibox*CLI> [Aug 12 14:59:20] -- Created MeetMe conference 1023 for conference '8600053'
No such command '[Aug 12 14:59:20] -- Created MeetMe conference 1023 for conference '8600053'' (type 'help [Aug 12' for other possible commands)
vicibox*CLI> [Aug 12 14:59:20] -- <SIP/zerone-00000272> Playing 'conf-onlyperson' (language 'en')
No such command '[Aug 12 14:59:20] -- <SIP/zerone-00000272> Playing 'conf-onlyperson' (language 'en')' (type 'help [Aug 12' for other possible commands)
vicibox*CLI> [Aug 12 14:59:21] == Manager 'sendcron' logged off from 127.0.0.1
No such command '[Aug 12 14:59:21] == Manager 'sendcron' logged off from 127.0.0.1' (type 'help [Aug 12' for other possible commands)
vicibox*CLI> vicibox*CLI>
No such command 'vicibox*CLI>' (type 'help vicibox*CLI>' for other possible commands)
[Aug 12 15:00:01] == Parsing '/etc/asterisk/manager.conf': [Aug 12 15:00:01] Found
[Aug 12 15:00:01] == Manager 'sendcron' logged on from 127.0.0.1
[Aug 12 15:00:01] == Parsing '/etc/asterisk/manager.conf': [Aug 12 15:00:01] Found
[Aug 12 15:00:01] == Manager 'sendcron' logged on from 127.0.0.1
[Aug 12 15:00:01] == Manager 'sendcron' logged off from 127.0.0.1
[Aug 12 15:00:01] == Manager 'sendcron' logged off from 127.0.0.1
[Aug 12 15:00:06] == Parsing '/etc/asterisk/manager.conf': [Aug 12 15:00:06] Found
[Aug 12 15:00:06] == Manager 'sendcron' logged on from 127.0.0.1
[Aug 12 15:00:06] == Manager 'sendcron' logged off from 127.0.0.1
vicibox*CLI>


When we tried calling our DID, there's no activity in CLI

Please Hep...
Thanks in Advance...

[williamconley]

vicidial cannot "handle" a call that never arrives at the server.

so for inbound your first goal is to cause the call to arrive at the server (even if it fails at that point, since any failure can be resolved ... but if the call never arrives at the server there's nothing to resolve!).

so the question is: HOW do you get the call to arrive at the server?

Glad you asked.

1) VOIP Carrier web site often has a method to specify the IP to send the call. If you do this and the call never arrives: is the Vicidial server's firewall open for the sending carrier's IP? Or is your router stopping the call from reaching the Vicidial server? In this case, be sure to point UDP port 5060 to Vicidial. If this results in a call with no sound, point UDP port range 10000-25000 to it as well.

2) Registration: In many carrier systems, you can create an account of some sort (with a user/pass associated) and point the DID to that account. Then you register the Vicidial server to that account and the calls will be sent through that "tunnel" into Vicidial. Registration from Vicidial to the carrier will associate both your IP and the Port through your firewall with the account at your carrier. So a call to that DID will be sent to the proper IP and even the proper port. Often this method will remove the need to forward ports through the router.

In both cases it may be necessary to have an externip=xx.xx.xx.xx (with your router's external ip) in sip.conf. Of course, this setting also affects outbound calls and if those are working you likely do not require this setting or already have it configured. Remember to change this setting if your IP changes!

Happy Hunting! Welcome to the Party!

[gtalbos]

Hi williamconley,

Thanks for your reply,

I was also thinking that the call might not able to arrived at the server, just wanted to make sure.

I'll be contacting our provider about this...

Keep you posted...


Thanks

[gtalbos]

Hi williamconley,

Just wanted to give you guys an update about the issue.

Its already been resolved, the issue is with the providers server, it seems that they're using a different IP to send the call to our server. they already fixed it, and the inbound call is already working.

Thanks for the support

[Boutchou]

2) "Registration: In many carrier systems, you can create an account of some sort (with a user/pass associated) and point the DID to that account."

Hi William, please can you help me how to point a DID to a sip account please