[Asterisk-Users] Limiting incoming SIP calls & OriginalCallerID on transfer

[Erik Barker]

OK,

I've fixed the '#' transfer problem. We setup a macro for dialing staff
extensions, however, it was missing the 'tr' options on the Dial
application:

[macro-staff-extension]
; Macro for Staff Extensions
exten => s,1,Dial(${ARG2},20,tr)  <------
exten => s,2,Voicemail(u${ARG1})
exten => s,102,Voicemail(b${ARG1})
exten => s,103,Hangup

I added the 'tr' and we can now perform call transfers while preserving
the correct CallerID information.

Thanks,

-- 
Erik Barker
Sr. Systems Engineer
NetNation Communications Inc.
http://www.netnation.com | http://www.domainpeople.com

On Tue, 2004-04-20 at 03:16, David Liu wrote:
> Hi Erik,
> 
> Can you post your dial plan from incoming PSTN to the receptionist?
> 
> David
> 
> ----- Original Message ----- 
> From: "Erik Barker" <erikb at netnation.com>
> To: <asterisk-users at lists.digium.com>
> Sent: Wednesday, April 21, 2004 4:37 AM
> Subject: Re: [Asterisk-Users] Limiting incoming SIP calls & OriginalCallerID
> on transfer
> 
> 
> > Thanks for the info David,
> >
> > I'll look at getting the '#' transfer option working again.... I had it
> > working at some point where we used it to park calls, however, it does
> > not appear to work anymore.
> >
> >
> > -- 
> > Erik Barker
> >
> > On Mon, 2004-04-19 at 11:13, David Liu wrote:
> > > Hi Erik,
> > >
> > > >From my experience with Polycom phones, I can answer you on your
> TRANSFER
> > > and Caller ID issue.  For Polycom, the transfer behavior is consultation
> > > transfer.  In consultation transfer mode, the caller ID of the
> transferer is
> > > passed to the ringing extension.  To actually pass the caller ID of the
> > > incoming caller on the PSTN, you would want to do a blind transfer.  So
> far,
> > > I have only figured to use the Asterisk transfer option # to do blind
> > > transfer.  And this assumes you have the t option enabled on the dial
> plan
> > > to the receptionist.
> > >
> > > Hope this helps.
> > > David
> > > ----- Original Message ----- 
> > > From: "Erik Barker" <erikb at netnation.com>
> > > To: <asterisk-users at lists.digium.com>
> > > Sent: Tuesday, April 20, 2004 6:19 PM
> > > Subject: [Asterisk-Users] Limiting incoming SIP calls & Original
> CallerID on
> > > transfer
> > >
> > >
> > > > I have 2 issues which I need to resolve on our production Asterisk
> > > > server:
> > > >
> > > >
> > > > We are currently using Polycom IP600 VOIP phones for our office which
> > > > are capable of handling 2 calls per SIP registration. What we're
> finding
> > > > is when staff are on the phone, Asterisk will pass them a second call
> > > > which will show up on their display, and an audible beep is heard over
> > > > the phone (regular call waiting). I would like to limit the number of
> > > > calls sent to each phone to 1 call only; otherwise respond as being
> > > > busy. I have looked at trying to accomplish this in the sip.conf by
> > > > using the 'incominglimit' and 'outgoinglimit' parameters, however, the
> > > > only one that *seems* to work is the 'incominglimit'. This prevents
> > > > further calls from reaching the phones, rings busy, but does not allow
> > > > our phones to initiate a 2nd call OR transfer their existing call. The
> > > > 'outgoinglimit' parameter does not seem to have any effect on limiting
> > > > whatsoever. Is there a way to limit calls passed to the phones from
> > > > Asterisk and also allow each phone to initiate 2 calls or transfer
> calls
> > > > (disable call waiting)??
> > > >
> > > > I have also looked at the WIKI for the parameters listed above and it
> > > > *appears* that 'outgoinglimit' should do what I want, however it also
> > > > states that this function has been disabled??
> > > >
> > > > "The _outgoinglimit__ is currently disabled in the source code of the
> > > > SIP channel."
> > > >
> > >
> http://www.voip-info.org/tiki-index.php?page=Asterisk%20sip%20incominglimit
> > > >
> > > >
> > > >
> > > > My second problem is that when external calls are transferred by our
> > > > receptionist to other staff members, the CallerID of course changes to
> > > > her Name instead of the original caller. Is there a way (in the
> > > > extensions logic or other) to preserve this CallerID information so
> that
> > > > staff members receive calls with the proper CallerID information?
> > > >
> > > >
> > > > Thanks,
> > > >
> > > >
> > > > -- 
> > > > Erik Barker
> > > >
> > > > _______________________________________________
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