> ----- Original Message ----
> From: Don Smith <[EMAIL PROTECTED]>
> To: [email protected]
> Sent: Thursday, 31 January, 2008 4:46:27 PM
> Subject: Re: [asterisk-users] Default delay time for Attended call
>
> A call comes in from the PSTN, Asterisk answers it, it goes to the directory,
> and then to the extension the caller designates and the user at that
> extension answers. The user at the extension then wants to transfer the call
> to another extension; on the Cisco 7940 they push the “more” soft key, then
> the “Transfer” soft key, then enter the extension number they want to
> transfer to, and hit the “dial” soft key. The user at the new extension
> answers and the talks to the user doing the transfer. They agree to transfer
> the call to the new extension and the person who got the original call then
> hits the “transfer” soft key and hangs up. 6 seconds later the caller and
> the new extension can talk to each other. The line at the new extension is
> silent for those 6 seconds.
Hi Don,
There is no setting you can adjust on Asterisk. For an attended transfer there
is no further interaction with the Asterisk dialplan once the transfer button
on your Cisco is pressed it's all handled automagically in the SIP channel and
channel.c.
When I do an attended transfer through Asterisk the transfer takes about 1
second to complete, 6 seconds seems like a very large amount of time for the
operation. With an attended transfer there are no new audio streams to set up
for Asterisk and therefore no NAT that could be slowing things down as the
audio trys to get through. The only thing that springs to mind would be if the
REFER request from your Cisco was getting dropped along the way and taking a
few re-transmits to get through. That wouldn't be likely to happen everytime
though so a consistent 6 second delay is puzzling.
Regards,
Greyman.
Make the switch to the world's best email. Get the new Yahoo!7 Mail now.
www.yahoo7.com.au/worldsbestemail
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
