vicidial.org

Hi all

i have installed asterisk 1.2.17 and astguiclinet 2.0.3.
i want to use them with sip, softphone xlite.,
i have studied the scracth_installation guide but in that there is configuration written for gs102 phone and other phone...

can any one tell me how to add extensions and users in sip.conf and extensnion.conf file for outbound calls... and local internal calls.,...

Regards
Sberg

Hi

sip.conf is basic configuration used to register with your sip provider
and extensions.

already given example to you in scratch instalaltion

[gs102]
disallow=all
allow=ulaw
allow=alaw
type=friend
username=gs102
secret=test
host=dynamic
dtmfmode=inband
defaultip=10.10.10.16
qualify=1000
mailbox=102

you can add your own extensions also below that config.

[1001]
disallow=all
allow=ulaw
allow=alaw
type=friend
username=1001
secret=1234
host=dynamic
dtmfmode=inband
qualify=1000
mailbox=1001

[1002]
disallow=all
allow=ulaw
allow=alaw
type=friend
username=1002
secret=1234
host=dynamic
dtmfmode=inband
qualify=1000
mailbox=1002


extension.conf also there in scratch installation

exten => 1001,1,Playback,transfer|skip ; "Please hold while..."
exten => 1001,2,Dial,sip/1001|20|to ; Ring, 20 secs max
exten => 1001,3,Voicemail,u1001 ; Send to voicemail...

exten => 1002,1,Playback,transfer|skip ; "Please hold while..."
exten => 1002,2,Dial,sip/1002|20|to ; Ring, 20 secs max
exten => 1002,3,Voicemail,u1002 ; Send to voicemail...


register with xlite with 1001 and 1002 in other PC
you can call each other


ram

Thanks ram,
After doing as u said i can use this users, extensions in the astguiclient...
bcoz i want them to use in the astguiclient... i want to do outbound part...
how can i do it ...

can u plse make me clear..

Regards,
Sberg

Hi


after adding the users

you need to add the same users in AGC also
from Admin login

add users from users menu, same what we have created in sip.conf

add phone under admin menu, same as users

then you will be able to make call
depending on the campaign settings

ready manager manual

ram

Hi

i want to test internally with out sip registered .
i want to check leads, etc all the faciliites ..is that possible in doing..

i can do normal calling from one user to another..in asterisk with xlite phone..but i want it to be done through astguiclient..

if any body know s this plse tell me..


Thanks in advance
Regards,
Sberg

Hi

i think should be possible

if you have 2 asterisk in your place

register each other as1 and as2

and install on as1 place agc, and as2 server only with Asterisk with extenstion.

take 2 PC and with x-lite

one pc rester withas2 server with extension in that server

make leads in agc point to that number

so when you login in as1 and use agc, start making calls

all calls will go to as2 extenstion, that way you can test.


let me know if this works

ram