<html><head><style type='text/css'>p { margin: 0; }</style></head><body><div style='font-family: Arial, Helvetica, sans-serif; font-size: 12pt; color: #000000'>That was it! I had a value (412 and 413) set for each phone. This overwrote the caller ID that I was setting in the dialplan. Removing the contents of the fromuser field cleared this issue. <br><br>Thanks Olle!<br><br><div><span style="font-family: arial,helvetica,sans-serif;">Brett Woollum</span><br style="font-family: arial,helvetica,sans-serif;"><span style="font-family: arial,helvetica,sans-serif;">Brett@Woollum.com</span><br></div><br><br>----- Original Message -----<br>From: "Olle E. Johansson" <oej@edvina.net><br>To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com><br>Sent: Tuesday, November 9, 2010 11:30:27 PM GMT -08:00 US/Canada Pacific<br>Subject: Re: [asterisk-users] Asterisk 1.6 (realtime) - Overwritten CallerID(num) Problem<br><br><br>10 nov 2010 kl. 02.38 skrev Brett Woollum:<br><br>> Good idea Paul.<br>> <br>> My debug output:<br>> [Nov 9 17:33:39] VERBOSE[2923] netsock.c: == Using SIP RTP CoS mark 5<br>> [Nov 9 17:33:39] VERBOSE[4175] pbx.c: -- Executing [412@sipphones:1] Set("SIP/413-00000005", "CALLERID(num)=22222") in new stack<br>> [Nov 9 17:33:39] VERBOSE[4175] pbx.c: -- Executing [412@sipphones:2] NoOp("SIP/413-00000005", "CallerID(num) is: 22222") in new stack<br>> [Nov 9 17:33:39] VERBOSE[4175] pbx.c: -- Executing [412@sipphones:3] Dial("SIP/413-00000005", "SIP/412") in new stack<br>> [Nov 9 17:33:39] VERBOSE[4175] netsock.c: == Using SIP RTP CoS mark 5<br>> [Nov 9 17:33:39] VERBOSE[4175] app_dial.c: -- Called 412<br>> [Nov 9 17:33:40] VERBOSE[4175] app_dial.c: -- SIP/412-00000006 is ringing<br>> [Nov 9 17:33:44] VERBOSE[4175] pbx.c: == Spawn extension (sipphones, 412, 3) exited non-zero on 'SIP/413-00000005'<br>> [Nov 9 17:33:44] VERBOSE[4175] pbx.c: -- Executing [h@sipphones:1] Hangup("SIP/413-00000005", "") in new stack<br>> [Nov 9 17:33:44] VERBOSE[4175] pbx.c: == Spawn extension (sipphones, h, 1) exited non-zero on 'SIP/413-00000005'<br>> <br>> As you can see on line 3, CallerID(num) was successfully set to "22222". SIP/412 is dialed next. It receives the call, but with "412" as the Caller ID number - even though the real source of the call was extension 413. The name I set in CallerID(name) works fine. <br>> <br>> My Extensions.conf for that context:<br>> [sipphones]<br>> exten => 412,1,Set(CALLERID(num)=22222)<br>> exten => 412,1,Set(CALLERID(all)="TEST"<22222>)<br>> exten => 412,n,NoOp(CallerID(num) is: ${CALLERID(num)})<br>> exten => 412,n,Dial(SIP/412)<br>> exten => 412,n,NoOp(${CALLERID(num)})<br>> <br>> If I disable sippusers and sippeers in extconfig.conf and put 412 and 413 into sip.conf directly, this code works (ie: the CallerID(num) I set makes it out to the destination phone properly).<br>Have you set the fromuser= field in the realtime database?<br><br>/O<br>-- <br>_____________________________________________________________________<br>-- Bandwidth and Colocation Provided by http://www.api-digital.com --<br>New to Asterisk? Join us for a live introductory webinar every Thurs:<br> http://www.asterisk.org/hello<br><br>asterisk-users mailing list<br>To UNSUBSCRIBE or update options visit:<br> http://lists.digium.com/mailman/listinfo/asterisk-users<br></div></body></html>