On 15/09/09 3:12 AM, Stanisław Pitucha wrote: > 2009/9/14 Olle E. Johansson<[email protected]>: >> Make sure that each device has a TRANSFER_CONTEXT dialplan variable. > > What about a situation where sip devices register at a proxy in front > of many asterisks and asterisks authorise all calls from that proxy? > I.e. I don't have any devices that asterisk would know about. That way > as far as asterisk is concerned, the call is a simple trunk call and > the B side (in A->B call) doesn't trigger any TRANSFER_CONTEXT setting > when doing a transfer.
If your users are not connected to Asterisk and Asterisk just sees all calls as origination from your proxy, surely the place to sort this out would be the proxy. Can you not set a variable in the proxy before sending the call to Asterisk and use the sip header function to retrieve it once in Asterisk? -- Cheers, Matt Riddell Director _______________________________________________ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
