[asterisk-users] Asterisk 1.6 (realtime) - Overwritten CallerID(num) Problem

[Brett Woollum]

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 at sipphones:1] Set("SIP/413-00000005", "CALLERID(num)=22222") in new stack 
[Nov 9 17:33:39] VERBOSE[4175] pbx.c: -- Executing [412 at sipphones:2] NoOp("SIP/413-00000005", "CallerID(num) is: 22222" ) in new stack 
[Nov 9 17:33:39] VERBOSE[4175] pbx.c: -- Executing [412 at 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 at 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 

Brett at Woollum.com 


----- Original Message ----- 
From: "Paul Belanger" <paul.belanger at polybeacon.com> 
To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com> 
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 <brett at woollum.com> 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: paul.belanger at polybeacon.com | IRC: pabelanger (Freenode) | 
Blog: http://blog.polybeacon.com | Twitter: http://twitter.com/pabelanger 

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