[Asterisk-Users] WARNING[19541]: res_features.c:844 builtin_atxfer: Did not read data.

[John Millican]

On Saturday November 26 2005 1:41 pm, John Millican wrote:
> On Saturday November 26 2005 1:26 pm, John Millican wrote:
> > Hello all,
> > I have just upgraded to 1.2 from 1.0.9 and am receiving and placing calls
> > as expected.  I have been trying to get atxfer working and am getting the
> > error message:
> >  WARNING[19541]: res_features.c:844 builtin_atxfer: Did not read data.
> > whenever I try a transfer.
> > In features.conf:
> > [general]
> > parkext => 700;
> > parkpos => 701-720;
> > context => parkedcalls;
> > ;parkingtime => 45;
> > transferdigittimeout => 3;
> > courtesytone = beep;
> > xfersound = beep;
> >
> > [featuremap]
> > blindxfer => #1		; Blind transfer
> > disconnect => *0		; Disconnect
> > automon => *1		; One Touch Record
> > atxfer => *2			; Attended transfer
> >
> > [applicationmap]
> > testfeature => #9,callee,Playback,tt-monkeys;
> >
> > in extensions.conf
> > [globals]
> > DYNAMIC_FEATURES=>automon#blindxfer#disconect#atxfer#testfeature
> >
> >
> > in CLI when attempting a transfer:
> > SIP/677-8544 answered Zap/1-1
> >     -- Started music on hold, class 'default', on channel 'Zap/1-1'
> >     -- Playing 'pbx-transfer' (language 'en')
> > Nov 26 13:13:33 WARNING[19541]: res_features.c:844 builtin_atxfer: Did
> > not read data.
> >     -- Playing 'beeperr' (language 'en')
> >     -- Stopped music on hold on Zap/1-1
> >
> > and then the channels are joined again as if nothing had happened.
> > I googled for the error message and searched voip-info.org but no results
> > on either.
>
> Sorry for the second post but thought I should add some info.
> setup is:
> PSTN ---> X100P in asterisk box -----> Linksys PA2-NA ----> phone1 on port
> 1 and Phone2 on port 2
> the linsys is set to g711 ulaw with inband signaling
>
> I am trying to transfer an incoming call from phone1 to phone 2
>
>
Okay let me try once again.  When I attempt a transfer either blind or 
attended i get the transfer prompt and then dial tone as I should.  Then what 
happens is when I press a digit the dial tone may or may not go away.  If I 
repeat that first digit I can sometimes get the dial tone to go away and 
asterisk accepts the remaining digits for the transfer without problem and 
the transfer happens.  I have tried increasing the dtmf playback level in the 
PAP2 from -16db all the way up to 0db.  This has not made any noticeable 
difference in detection.  I have also increased the DTMF playback length 
from .1 to .3 again no success.
Any help would be greatly appreciated.
Thank You
John Millican