Problem Transferring Calls

Hello, I am rather new to Asterisk and the Flash Operator
Panel, however I do
know that I have Asterisk working as I am able to make phone
calls between 2
SIP phones.  

I then installed and tried both version .26 and .27 of the
Flash Operator Panel,
but can't get it to work completely. I'm trying to drag
one phone to the other
to create a call, however when I do this the first phone
rings and when I
answer, the 2nd never rings and the first simply hangs up. I
ran op_server.pl
with debug on and it looks like the callerid variable is
never getting set so
it doesn't know where to dial back to.

** DIGEST_EVENT: GOT
SIP/D3877|ocupado1|&incoming,[]|1153925740.55-0|SIP/D387
7-9666
** DIGEST_EVENT: canal: SIP/D3877
** DIGEST_EVENT: quehace: ocupado1
** DIGEST_EVENT: dos: &incoming,[] <------ BLANK????
** DIGEST_EVENT: uniqueid: 1153925740.55-0
** DIGEST_EVENT: canalid: SIP/D3877-9666

Below is my op_buttons.cfg:

[SIP/D3877]
Position=1
Label="D3877"
callerid="D3877"
Extension=D3877
Context=internal
Icon=1

[SIP/D3878]
Position=n
Label="D3878"
callerid="D3878"
Extension=D3878
Context=internal
Icon=1

Here is the output I get from the Asterisk console as well:

-- Executing Goto("SIP/D3877-9666",
"internal") in new stack
Jul 26 09:55:45 NOTICE[11816]: pbx.c:1753
pbx_extension_helper: No such label
'internal' in extension 's' in context 'default'
Jul 26 09:55:45 WARNING[11816]: pbx.c:6514
ast_parseable_goto: Priority
'internal' must be a number > 0, or valid label
== Spawn extension (default, s, 1) exited non-zero on
'SIP/D3877-9666'

Below is my extensions.conf, however I don't think this is
the problem.

[general]
static=yes
writeprotect=yes

[globals]
HARD1=SIP/D3877
HARD2=SIP/D3878
SOFT1=SIP/3876
SOFT2=SIP/3875

[internal] 
exten => 1001,1,Goto(conferences,1001,1)

exten => 3877,n,Dial($,15,wt)
exten => 3877,n,Voicemail(u$internal)  
exten => 3877,102,Voicemail(b$intenral)

exten => 3878,1,Dial($,15,wt)
exten => 3878,2,Voicemail(u$internal)  
exten => 3878,102,Voicemail(b$intenral)

exten => i,1,playback(tt-weasels) 

[conferences]
exten => 1001,1,MeetMe(1001)

Any help is appreciated.

Thanks,
Brett

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Hi,

On Jul 27, 2006, at 12:45 PM, brettbcytelcom.com wrote:

> Hello, I am rather new to Asterisk and the Flash
Operator Panel, 
> however I do
> know that I have Asterisk working as I am able to make
phone calls 
> between 2
> SIP phones.
>
> I then installed and tried both version .26 and .27 of
the Flash 
> Operator Panel,
> but can't get it to work completely. I'm trying to
drag one phone to 
> the other
> to create a call, however when I do this the first
phone rings and 
> when I
> answer, the 2nd never rings and the first simply hangs
up. I ran 
> op_server.pl
> with debug on and it looks like the callerid variable
is never getting 
> set so
> it doesn't know where to dial back to.

> <snip>

> Below is my op_buttons.cfg:
>
> [SIP/D3877]
> Position=1
> Label="D3877"
> callerid="D3877"
> Extension=D3877
                       ^^^^^^
The extension is wrong. You cannot dial that from the phone,
change 
Extension in every button to the number you dial on the
phone to ring 
that extension (probably 3877)

Regards,

--
Nicolas Gudino
Buenos Aires - Argentina



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