<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'>Good idea Paul.<br><br>My debug output:<br><font size="2"><span style="font-family: Courier New,courier,monaco,monospace,sans-serif;">[Nov 9 17:33:39] VERBOSE[2923] netsock.c: == Using SIP RTP CoS mark 5</span><br style="font-family: Courier New,courier,monaco,monospace,sans-serif;"><span style="font-family: Courier New,courier,monaco,monospace,sans-serif;">[Nov 9 17:33:39] VERBOSE[4175] pbx.c: -- Executing [412@sipphones:1] Set("SIP/413-00000005", "CALLERID(num)=22222") in new stack</span><br style="font-family: Courier New,courier,monaco,monospace,sans-serif;"><span style="font-family: Courier New,courier,monaco,monospace,sans-serif;">[Nov 9 17:33:39] VERBOSE[4175] pbx.c: -- Executing [412@sipphones:2] NoOp("SIP/413-00000005", <span style="font-weight: bold;">"CallerID(num) is: 22222"</span>) in new stack</span><br style="font-family: Courier New,courier,monaco,monospace,sans-serif;"><span style="font-family: Courier New,courier,monaco,monospace,sans-serif;">[Nov 9 17:33:39] VERBOSE[4175] pbx.c: -- Executing [412@sipphones:3] Dial("SIP/413-00000005", "SIP/412") in new stack</span><br style="font-family: Courier New,courier,monaco,monospace,sans-serif;"><span style="font-family: Courier New,courier,monaco,monospace,sans-serif;">[Nov 9 17:33:39] VERBOSE[4175] netsock.c: == Using SIP RTP CoS mark 5</span><br style="font-family: Courier New,courier,monaco,monospace,sans-serif;"><span style="font-family: Courier New,courier,monaco,monospace,sans-serif;">[Nov 9 17:33:39] VERBOSE[4175] app_dial.c: -- Called 412</span><br style="font-family: Courier New,courier,monaco,monospace,sans-serif;"><span style="font-family: Courier New,courier,monaco,monospace,sans-serif;">[Nov 9 17:33:40] VERBOSE[4175] app_dial.c: -- SIP/412-00000006 is ringing</span><br style="font-family: Courier New,courier,monaco,monospace,sans-serif;"><span style="font-family: Courier New,courier,monaco,monospace,sans-serif;">[Nov 9 17:33:44] VERBOSE[4175] pbx.c: == Spawn extension (sipphones, 412, 3) exited non-zero on 'SIP/413-00000005'</span><br style="font-family: Courier New,courier,monaco,monospace,sans-serif;"><span style="font-family: Courier New,courier,monaco,monospace,sans-serif;">[Nov 9 17:33:44] VERBOSE[4175] pbx.c: -- Executing [h@sipphones:1] Hangup("SIP/413-00000005", "") in new stack</span><br style="font-family: Courier New,courier,monaco,monospace,sans-serif;"><span style="font-family: Courier New,courier,monaco,monospace,sans-serif;">[Nov 9 17:33:44] VERBOSE[4175] pbx.c: == Spawn extension (sipphones, h, 1) exited non-zero on 'SIP/413-00000005'</span></font><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><font size="2"><span style="font-family: Courier New,courier,monaco,monospace,sans-serif;">[sipphones]</span><br style="font-family: Courier New,courier,monaco,monospace,sans-serif;"><span style="font-family: Courier New,courier,monaco,monospace,sans-serif;">exten => 412,1,Set(CALLERID(num)=22222)</span><br style="font-family: Courier New,courier,monaco,monospace,sans-serif;"><span style="font-family: Courier New,courier,monaco,monospace,sans-serif;">exten => 412,1,Set(CALLERID(all)="TEST"<22222>)</span><br style="font-family: Courier New,courier,monaco,monospace,sans-serif;"><span style="font-family: Courier New,courier,monaco,monospace,sans-serif;">exten => 412,n,NoOp(CallerID(num) is: ${CALLERID(num)})</span><br style="font-family: Courier New,courier,monaco,monospace,sans-serif;"><span style="font-family: Courier New,courier,monaco,monospace,sans-serif;">exten => 412,n,Dial(SIP/412)</span><br style="font-family: Courier New,courier,monaco,monospace,sans-serif;"><span style="font-family: Courier New,courier,monaco,monospace,sans-serif;">exten => 412,n,NoOp(${CALLERID(num)})</span></font><br><span style="font-family: arial,helvetica,sans-serif;"><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><br>Brett Woollum</span><br style="font-family: arial,helvetica,sans-serif;"><div><span style="font-family: arial,helvetica,sans-serif;">Brett@Woollum.com</span><br></div><br><br>----- Original Message -----<br>From: "Paul Belanger" <paul.belanger@polybeacon.com><br>To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com><br>Sent: Tuesday, November 9, 2010 5:18:36 PM GMT -08:00 US/Canada Pacific<br>Subject: Re: [asterisk-users] Asterisk 1.6 (realtime) - Overwritten CallerID(num) Problem<br><br>On Tue, Nov 9, 2010 at 6:35 PM, Brett Woollum <brett@woollum.com> wrote:<br>> Nobody has any idea why the Caller ID is being overwritten when using<br>> Asterisk Realtime for the SIP users?<br>><br>No, perhaps you can _show_ us the problem.<br><br>https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information<br>-- <br>Paul Belanger | dCAP<br>Polybeacon | Consultant<br>Jabber: paul.belanger@polybeacon.com | IRC: pabelanger (Freenode) |<br>Blog: http://blog.polybeacon.com | Twitter: http://twitter.com/pabelanger<br><br>-- <br>_____________________________________________________________________<br>-- Bandwidth and Colocation Provided by http://www.api-digital.com --<br>New to Asterisk? 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