Nobody has any idea why the Caller ID is being overwritten when using Asterisk Realtime for the SIP users?
Brett Woollum [email protected] ----- Original Message ----- From: "Brett Woollum" <[email protected]> To: [email protected] Sent: Sunday, November 7, 2010 3:08:50 PM GMT -08:00 US/Canada Pacific Subject: [asterisk-users] Asterisk 1.6 (realtime) - Overwritten CallerID(num) Problem Hello, I am running Asterisk 1.6 with Realtime enabled for my SIP users and peers. The backend is a MySQL database running through the ODBC backend in Asterisk. At this point everything works in terms of phones registering, placing calls between them, etc. However, I am having a problem setting the Caller ID number whenever I am using the Realtime database for the SIP users/peers. If I use a static sip.conf configuration instead of the database, everything works fine. Unfortunately a static sip.conf file won't work in my application. In this example: exten => 412,1,Set(CALLERID(all)="TEST"<22222>) exten => 412,n,NoOp(CallerID(num) is: ${CALLERID(num)}) ;;;PS: This shows the correct number of "22222" on the CLI console... exten => 412,n,Dial(SIP/412) Whenever another phone calls extension 412, the call is forwarded to SIP/412 and should have "TEST" as the CallerID name and "22222" as the CallerID number. But, whenever I am using the realtime backend, the caller ID number always displays on the destination phone as that phone's username. Meaning, if phone SIP/412 receives the call from the example above, the caller ID name displayed is "TEST" but the caller ID number is always "412". What could be causing this? Brett Woollum [email protected] -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
