Hello All,
Can anyone help me with this... This is what my program does: -
1) At certain time the system generates a ".call" and make a call to User A.
2) When User A picks up the phone call, system will play a menu select
option.
a) Press 1 to call your supervisor.
b) Press 2 to call your manager.
c) Press 3 to leave a voice message.
3) When the User A press 1 to call his supervisor... The system has to
put the User A on hold and place a call to the supervisor.
4) Once the supervisor picks up the call, User A has to be in session
with his supervisor.
Now I have already got part 1 and 2 done... but I am stuck with part 3
and 4.
This is how I generate my call to the supervisor: -
===================================
if($asm->connect())
{
$call = $asm->send_request('Originate',
array('Channel'=>"SIP/xo-out/$supervisor_num",
'Context'=>'default',
'Priority'=>1,
'Callerid'=>$cid));
$asm->disconnect();
}
One the *CLI I do see the call, but its failing: -
AGI Rx << STREAM FILE
/var/spool/asterisk//tmp//text2wav_e08db16aede0af38ebb90a1c69ee19e3 "" 0
AGI Tx >> 200 result=0 endpos=26224
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'phpagi' logged on from 127.0.0.1
> Channel SIP/xo-out-08f8ae10 was answered.
== Starting SIP/xo-out-08f8ae10 at default,,1 failed so falling back
to exten 's'
== Manager 'phpagi' logged off from 127.0.0.1
AGI Rx << STREAM FILE goodbye "" 0
Can anyone put some light what I am missing here... Why the call is
dropped on both end...?
Cheers,
Nitesh
_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
