vicidial.org

Hi,

I'm a newbie on this, but have plenty of experience using Linux.

My SIP Trunking is Vitelity and I have used the configuration provided by them to set up the carrier (sip.conf and extension.conf) from within VD admin panel.

Using Zoipper softphone I can't receive the call made by ViciDial. I can call between extensions, and I can even dial from Zoipper directly.

On asterisk CLI I get the message whenever an agent signs in:

Code: Select all

channel.c:5429 __ast_request_and_dial: Unable to request channel SIP/xxx_1001


On the agent screen after some seconds of logging in:

Code: Select all

Logged in as User : 1001 on Phone: SIP/keth_1001

No one is in your session: 8600051


Registration is:

Code: Select all

register => xxxx:XXXXXXXXXX@inbound31 . vitelity . net:5060


My sip.conf is:

Code: Select all

[vitel-inbound]
type=friend
dtmfmode=auto
host=inbound31 . vitelity . net
context=inbound
         
username=[[user]]
secret=[[pwd]]
            
allow=all
insecure=port,invite
canreinvite=no
         
[vitel-outbound]
type=friend
dtmfmode=auto
host=outbound . vitelity . net
username=[[user]]
fromuser=[[user]]
secret=[[pwd]]
trustrpid=yes
sendrpid=yes
allow=all
canreinvite=no


extensions.conf is:

Code: Select all

[outbound]
exten => _NXXNXXXXXX,1,goto(outbound,1,1)
exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@vitel-outbound)
exten => _011.,1,Dial(SIP/${EXTEN}@vitel-outbound)


Any thought where the problem is? Thanks.

sip show peers

sip show registry

you will likely find that your sip peer is unreachable or unregistered

switch to IP based authentication (if carrier) or reduce your keepalive (registration timeout, if phone)

we only charge $25 to set up a carrier or phone by the way ...