Nobody has any idea why the Caller ID is being overwritten when using Asterisk Realtime for the SIP users?
Brett Woollum br...@woollum.com ----- Original Message ----- From: "Brett Woollum" <br...@woollum.com> To: asterisk-users@lists.digium.com 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 br...@woollum.com -- _____________________________________________________________________ -- 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