[Asterisk-Users] Transfers (sip or asterisk "#' base) broken in certain scenario

[Chris A. Icide]

I've got an interesting scenario where transfers while getting an invite 
seem to break.


Here is the scenario:  You have a receptionist who has a 6 line phone (in 
this case, a polycom ip600 - also tested with a Cisco 7960) the 
receptionist has all six lines available for use (in the case of the cisco 
I tried registering all lines as one number as well as registering multiple 
lines and having the dialplan do roll-over).  The receptionist receives a 
call and begins to transfer the call and in the middle of transferring the 
call, another call is received.

This is what happens:

If the receptionist is using the cisco/polycom soft button labelled 
transfer, the transfer goes through, however, the receptionist never knows 
another call was coming in.  It went straight to the 'busy' priority (+101 
in the case of a single registered extension or +101, +101, +101, +101... 
all the way through a roll-over dialplan straight to busy handling even 
though 5 of the 6 lines were available with no active calls)

In the case of using # to effect a transfer, the receptionist hits pound 
and begins entering the phone number to transfer to and a call comes 
in.  Immediately the receptionist is send to the 'i' extension while doing 
the transfer, and the new call is presented (rings and LCD screen shows 
information).  In some cases depending on the timing of the new call (if 
it's received after pressing # but before entering an extension to transfer 
to) you can get the call back, place it on hold and take the new call, 
however if you are in the process of entering the number to transfer when 
the new call comes in, then the original call is immediately acted on.  In 
other words, if I was typing 2004 and had entered 20 when the new call came 
in, asterisk grabs the 20 and tries to transfer the call to it.  No matter 
what happens, the call is lost to the receptionist, unable to get it back, 
even if there is a valid 'i' handler.

Is there anyone out there who has a busy enough system to have seen this as 
well?  If so, how have you dealt with it?

The only solution I can think of is to place all inbound calls into a 
queue, then pass them to the receptionist as the only agent (permanent 
agent) of the queue.  Then set limits on the number of calls the 
receptionist is allowed to have incoming 1, outgoing 1 so the queue won't 
ring the receptionists phone unless there is no active sessions.



Relevant info below

Asterisk CVS-D2004.07.03.19.00.00-07/05/04-14:41:51
Polycom IP500 sip.ld version 1.2.0.0318
Cisco 7960 SIP image 6.0


sip.conf entries are simple and are not a factor in the problem - items 
that may be important from sip.conf are:
canreinvite=no     ; want asterisk in the media stream for features
type=friend          ; haven't tried this by creating a user and peer for 
each handset (yuck)

extensions.conf entries are either one of the following (tested against both)

exten => _XXXX,1,Dial(SIP/1000,30,t)
exten => _XXXX,2,Voicemail(u1000)
exten => _XXXX,3,Hangup
exten => _XXXX,102,Voicemail(b1000)
exten => _XXXX,103,Hangup

exten => _XXXX,1,Dial(SIP/1000,30,t)
exten => _XXXX,2,Voicemail(u1000)
exten => _XXXX,3,Hangup
exten => _XXXX,102,Dial(SIP/1001,30,t)
exten => _XXXX,103,Voicemail(u1000)
exten => _XXXX,104,Hangup
exten => _XXXX,203,Dial(SIP/1002,30,t)
.......
exten => _XXXX,506,Dial(SIP/1005,30,t)
exten => _XXXX,507,Voicemail(u1000)
exten => _XXXX,508,Hangup
exten => _XXXX,607,Voicemail(b1000)
exten => _XXXX,608,Hangup


-Chris