Asterisk 1.6.2.20 on Debian Lenny

I’m finding that if no one answers an attended transfer (timeout set by
atxfernoanswertimeout), then the transferrer is handed back to the original
caller, and a beep is played.

In 1.4 I was able to indicate the timeout and failure by setting xferfailsound
to a custom recording, but this doesn’t seem to happen in 1.6

How can I indicate a timeout to the transferrer?

Many thanks

John

Here is a weird problem that I have had several reports on lately.
Customers using our 1.4 based product placing calls to large companies
(two examples – AT&T and ComEd, numbers for which I will provide below)
that have auto-attendants based on voice recog get disconnected – by the
remote end – when the “transfer to an agent” occurs. The message they
hear just before disconnect is that “your call cannot be completed from
this calling area”. We send all US 800 numbers via Vitelity, and I
somewhat suspect the problem may be there, because the same customers
call the same numbers from cell phones and have no issue. Perhaps
Vitelity or its upstream is being asked to make a transfer that isn’t
possible? I haven’t opened a ticket with them yet… my next call.

Anyway, I am interested to know if others are experiencing this in the
US. Try to call AT&T at 877-722-3755 and say “agent”. You will hear a
few DTMF tones, then either it will ring an agent or you will get the
“cannot be completed” message. Also ComED (electric company in the
midwest) at 800-334-7661. Hit “2″, then say “agent”. A different voice
on the “cannot be completed” message, but the effect of course is the
same!

Cheers,

Jeff LaCoursiere
SunFone

Hi, I’m seeing an odd issue at a recent installation and have been unable
to resolve it. Caller A calls Caller B and Caller B transfers to Caller
C. Using a blind transfer, if Caller C doesn’t answer then Caller A gets
Caller C’s voicemail. (as expected) However if doing an attended transfer
(Caller B simply hits transfer, then transfer without announcing) then
Caller A winds up in Caller B’s voicemail box if Caller C doesn’t answer.
I realize my users are misusing “attended transfer” but it still doesn’t
seem to work as expected. I have tried setting both canreinvite=no and
directmedia=no in sip.conf and it doesn’t seem to make a difference.

Asterisk is version 1.8.4.2. All phones are Polycom 650. I have another
installation that is on Asterisk 1.6 with a mixture of cisco and polycom
phones and I cannot reproduce the behavior there. Thanks in advance. If I
need to provide more information please let me know.

Thanks,
Jeff Roberts

Hello Folks!
I´m trying attended transfer with asterisk 1.6 and see the message
“WARNING[] features.c: Failed to play transfer sound!” once in a while when
the transfer failes.

Any idea what can be happening?

Thanks a lot!!!

Hi,

since version 1.4.12 the libpri package supports ETSI Explicit Call
Transfer feature:
http://downloads.asterisk.org/pub/telephony/libpri/ChangeLog-1.4.12

Does anyone know, how to use this feature in the dialplan? I can not
find any hints in the asterisk doc.

Best regards,
-Thorsten-