How to open inbound calls in vicidial ?

[ostas]

Dear collages,

Please help me to make inbound calls work in vicidila software. My outbound calls are working properly but I can not to set up inbound connection to work.

Let me kindly ask you if a configuration of my carrier is proper.

Account Entry:

[callcentrevoip1]
type=peer
host=sip.extravoip.pl
port=5060
outboundproxy=sip.extravoip.pl
outboundproxyport=5060
fromdomain=sip.extravoip.pl
context=extravoip
fromuser=XXXXXXXX
username=XXXXXXXX
secret=XXXXXXXXXX
canreinvite=no
dtmfmode=rfc2833
nat=no
disallow=all
allow=ulaw
allow=g729
rtptimeout=60
dtmfmode=RFC2833
insecure=no


Dial plan:

exten => _3348XXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _3348XXXXXXXXX,2,Dial(${CALLCENTREVOIP1}/${EXTEN:2},60,tTor)
exten => _3348XXXXXXXXX,3,Hangup


Please help me to diagnose a problem. What do I need to configure inbound calls ?? I would like to attache each DID to separate agent. How to do it??


Version of my soft is the following:

VERSION: 2.6-375a
BUILD: 120831-1523
© 2012 ViciDial Group


Regards, Jacek

[Michael_N]

you can read about it in manager manual

http://www.vicidial.org/store.php

[rishuk]

I can do this for you @ 50 USD, it ll take hardly 2 hours mail me at advik1981@gmail.com

[williamconley]

1) Welcome to the Party!

2) Please post your installation method (the name of the .iso with the version included ...).

3) Inbound calls are handled based on two basic stages: The first stage involves getting the call to route to your server. This is done either at a control panel at your carrier's website/account or by "registering" a sip/iax trunk to the carrier (that is the purpose of registration: "send my calls here!"). Once the calls have been routed to your server, they will arrive in a sip context based on the IP address of the sending carrier. This is where the second stage comes in: routing the call to the Vicidial agi scripting. This is simple: "context=trunkinbound". This goes in the account entry of the carrier in question. The system will look for a "host=xx.xx.xx.xx" matching the ip/domain of the sender of the call. When it finds a match, it will check "context=" and send the call in extensions.conf to that context. If you check "trunkinbound" in extensions.conf, you will note that the only thing that can possibly happen in this context ... is to send the call to the inbound DID script. Thus turning control of the call over to Vicidial's control apparatus.

So: first get the call sent to the server, then put "context=trunkinbound" into your sip account and you're done. Watch the call happen in the "asterisk cli" (in sip debug mode if possible) and send us the results if this does not work.

But remember: If there is no activity in the asterisk sip debug cli when the DID is called ... the call is not routing to this server and NO changes on this server can route the call until that has been resolved. If it is necessary to register to get the calls routed, then work on registration before continuing, but if registration is not part of your carrier's process, you may need to open a port (udp 5060) through your firewall from the carrier's ip to receive these calls.

Happy Hunting.

[boybawang]

add a context=trunkinbound to your carrier , itwill look like this

[callcentrevoip1]
type=peer
host=sip.extravoip.pl
port=5060
outboundproxy=sip.extravoip.pl
outboundproxyport=5060
fromdomain=sip.extravoip.pl
fromuser=XXXXXXXX
username=XXXXXXXX
secret=XXXXXXXXXX
canreinvite=no
dtmfmode=rfc2833
nat=no
disallow=all
allow=ulaw
allow=g729
rtptimeout=60
dtmfmode=RFC2833
insecure=no
context=trunkinbound

then go to admin interface then inbound then add a did and then add the incoming number