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 [...@sipphones:1]
Set("SIP/413-00000005", "CALLERID(num)=22222") in new stack
[Nov 9 17:33:39] VERBOSE[4175] pbx.c: -- Executing [...@sipphones:2]
NoOp("SIP/413-00000005", "CallerID(num) is: 22222" ) in new stack
[Nov 9 17:33:39] VERBOSE[4175] pbx.c: -- Executing [...@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 [...@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).
Brett Woollum
[email protected]
----- Original Message -----
From: "Paul Belanger" <[email protected]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<[email protected]>
Sent: Tuesday, November 9, 2010 5:18:36 PM GMT -08:00 US/Canada Pacific
Subject: Re: [asterisk-users] Asterisk 1.6 (realtime) - Overwritten
CallerID(num) Problem
On Tue, Nov 9, 2010 at 6:35 PM, Brett Woollum <[email protected]> wrote:
> Nobody has any idea why the Caller ID is being overwritten when using
> Asterisk Realtime for the SIP users?
>
No, perhaps you can _show_ us the problem.
https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information
--
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: [email protected] | IRC: pabelanger (Freenode) |
Blog: http://blog.polybeacon.com | Twitter: http://twitter.com/pabelanger
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