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
