Re: [Asterisk-Users] Limiting incoming SIP calls & OriginalCallerID on transfer
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" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> 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" <[EMAIL PROTECTED]>
> > > To: <[EMAIL PROTECTED]>
> > > 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
>
Re: [Asterisk-Users] Limiting incoming SIP calls & OriginalCallerID on transfer
Hi Erik, Can you post your dial plan from incoming PSTN to the receptionist? David - Original Message - From: "Erik Barker" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> 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" <[EMAIL PROTECTED]> > > To: <[EMAIL PROTECTED]> > > 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 > > > > > > ___ > > > Asterisk-Users mailing list > > > [EMAIL PROTECTED] > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > To UNSUBSCRIBE or update options visit: > > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
