<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'>Hi Carlos.<br><br>Yes I did have fromuser set, which was the problem. I removed this for each extension and that solved the issue. <br><br>Thanks!<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: "Carlos Chavez" <cursor@telecomabmex.com><br>To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com><br>Sent: Wednesday, November 10, 2010 8:47:38 AM GMT -08:00 US/Canada Pacific<br>Subject: Re: [asterisk-users] Asterisk 1.6 (realtime) - Overwritten CallerID(num) Problem<br><br>On Tue, 2010-11-09 at 17:38 -0800, Brett Woollum wrote:
> Good idea Paul.
>
> My debug output:
> [Nov 9 17:33:39] VERBOSE[2923] netsock.c: == Using SIP RTP CoS mark
> 5
> [Nov 9 17:33:39] VERBOSE[4175] pbx.c: -- Executing
> [412@sipphones:1] Set("SIP/413-00000005", "CALLERID(num)=22222") in
> new stack
> [Nov 9 17:33:39] VERBOSE[4175] pbx.c: -- Executing
> [412@sipphones:2] NoOp("SIP/413-00000005", "CallerID(num) is: 22222")
> in new stack
> [Nov 9 17:33:39] VERBOSE[4175] pbx.c: -- Executing
> [412@sipphones:3] Dial("SIP/413-00000005", "SIP/412") in new stack
> [Nov 9 17:33:39] VERBOSE[4175] netsock.c: == Using SIP RTP CoS mark
> 5
> [Nov 9 17:33:39] VERBOSE[4175] app_dial.c: -- Called 412
> [Nov 9 17:33:40] VERBOSE[4175] app_dial.c: -- SIP/412-00000006 is
> ringing
> [Nov 9 17:33:44] VERBOSE[4175] pbx.c: == Spawn extension
> (sipphones, 412, 3) exited non-zero on 'SIP/413-00000005'
> [Nov 9 17:33:44] VERBOSE[4175] pbx.c: -- Executing
> [h@sipphones:1] Hangup("SIP/413-00000005", "") in new stack
> [Nov 9 17:33:44] VERBOSE[4175] pbx.c: == Spawn extension
> (sipphones, h, 1) exited non-zero on 'SIP/413-00000005'
>
> 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.
>
> My Extensions.conf for that context:
> [sipphones]
> exten => 412,1,Set(CALLERID(num)=22222)
> exten => 412,1,Set(CALLERID(all)="TEST"<22222>)
> exten => 412,n,NoOp(CallerID(num) is: ${CALLERID(num)})
> exten => 412,n,Dial(SIP/412)
> exten => 412,n,NoOp(${CALLERID(num)})
>
> 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).
>
Are you using the fromuser field in the realtime table? I had this
problem once when from user was set and user kept receiving that as the
callerid.
>
--
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001
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