asterisk users - Apr 2005 - Sip transfer and redirect in a Company setting

I have an asterisk box setup and dialplan that is something like this:

(t1/pri)
    |
[incoming]
1234,1,Dial(SIP/secretary,30,rt)
1234,2,Voicemail(su1234@company)

Now the "t" in the dial lets the sec transfer with # and if the person
transferred to is unavail it goes to their voicemail -- that works great.

However if the sec tells her phone to redirect to another phone (CFWDall 
on a 7960) asterisk will redirect that call to that phone.  However it 
uses the sec's context to dial, which if redirecting internally included 
voicemail.

So if the sec redirects to another phone and that phone does not answer, 
the redirected phone's voicemail plays and not the companies.

I just wanted to see if anyone else had this problem (and a solution).

Jeb Campbell
jebc@c4solutions.net

If I understand your problem correctly, you have user a setup with vm
box a, and user b with vm box b, when secretary uses local callFWD
from phone a to phone b, vm of b picks up. And you want that if it was
redirected from phone a vm box of a should answer. I think (I never
tested this) that the RDNIS variable (${RDNIS}) will hold the CallerID
of phone a, which you can use in your dialplan to use for voicemail if
it exists, something like this will do:
exten => _1XX,1,Dial(SIP/${EXTEN},45,tr)
exten => _1XX,2,GotoIf($[${RDNIS} > 0 ]?10)
exten => _1XX,3,VoiceMail(u${EXTEN})
exten => _1XX,10,VoiceMail(u${RDNIS})

I'm not sure if DNID or RDNIS will work for SIP phones, but one of
those should work.
Another way to get this done (ugly), is to set a variable for the
channel before you use the Dial command, like this

exten => _1XX,1,SetVar(ORIGINAL_EXTEN=${EXTEN})
and then test if ${ORIGINAL_EXTEN} is different than ${EXTEN}

Look at this:
http://bugs.digium.com/bug_view_page.php?bug_id=0002590
this:
http://bugs.digium.com/bug_view_page.php?bug_id=0002763
and this:
http://www.voip-info.org/wiki-RDNIS

I hope this helps.

On 4/11/05, Jeb Campbell <jebc@c4solutions.net>
wrote:
> I have an asterisk box setup and dialplan that is something like this:
> 
> (t1/pri)
>    |
> [incoming]
> 1234,1,Dial(SIP/secretary,30,rt)
> 1234,2,Voicemail(su1234@company)
> 
> Now the "t" in the dial lets the sec transfer with # and if the
person
> transferred to is unavail it goes to their voicemail -- that works great.
> 
> However if the sec tells her phone to redirect to another phone (CFWDall
> on a 7960) asterisk will redirect that call to that phone.  However it
> uses the sec's context to dial, which if redirecting internally
included
> voicemail.
> 
> So if the sec redirects to another phone and that phone does not answer,
> the redirected phone's voicemail plays and not the companies.
> 
> I just wanted to see if anyone else had this problem (and a solution).
> 
> Jeb Campbell
> jebc@c4solutions.net
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