Hello asterisk-users,
I setup my asterisk to support several features like
automon,blindxfer,atxfer,parkcall etc. by using features.conf and the
global variable
DYNAMIC_FEATURES=automon#blindxfer#atxfer#parkcall#disconnect in
extension.conf. Every Dial() command in my diaplan has the appropriate
parameters out of {tTkWwW}.
For calls from my SIP phones everything works fine. Pressing #1 will
transfer, pressing ## will automon and so on.
But if I originate the call from either a call file (which is used by a
callback application on my setup) or the manager api (which is used by a
webadressbook, which automatically dials and connects to the phone on
your desk) these features are not available even the manager api jumps
to the same context as a normal call from a phone and uses the same
Dial() command with {tTwWkK}. This means, that alle the nice feature
keys which "normally" work do not if I originate the call over the
webfrontend.
Therefore, either the global variable is not known to the call which
came from the call file or the manager api or something else is going
wrong.
Here's my call file:
Channel: SIP/tol/06151154260
Account: t-online
CallerID: 03222XXXXXXX <03222XXXXXXX>
MaxRetries: 4
RetryTime: 15
WaitTime: 60
Context: doCallBackVolkerStage2
Extension: s
Priority: 1
(the Context doCallBackVolkerStage2 will run DISA).
and that's the PHP code orgiginating the calls:
$socket = fsockopen("127.0.0.1","5038", $errno, $errstr, $timeout);
fputs($socket, "Action: Login\r\n");
fputs($socket, "UserName: webdial\r\n");
fputs($socket, "Secret: XXXXXX\r\n\r\n");
fputs($socket, "Action: Originate\r\n");
fputs($socket, "Channel: $channel\r\n");
fputs($socket, "Exten: $tonumber\r\n");
fputs($socket, "CallerID: $callerid\r\n");
fputs($socket, "Context: doLocalCalls\r\n");
fputs($socket, "MaxRetries: 1\r\n");
fputs($socket, "RetryTime: 15\r\n");
fputs($socket, "WaitTime: 60\r\n");
fputs($socket, "Priority: 1\r\n\r\n");
fputs($socket, "SetLanguage: de\r\n\r\n");
fputs($socket, "Action: Logoff\r\n\r\n");
(the Context doLocalCalls is the same as on any other SIP phone. It will run
Dial().
Any hint on what I'm doing wrong or where to check else?
Regards
Volker
signature.asc
Description: Digital signature
_______________________________________________ --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
