[asterisk-users] Interrupt VM and Steal a call.

[Robert Lister]

On Fri, Feb 22, 2008 at 09:05:17AM -0500, Michael Munger wrote:
> Two questions:
> 
> 1.       Does anyone have a good way to transfer a call from inside
> comedian mail to the current extension? The problem is: let's say the
> phone goes to voicemail after 4 rings, yet I don't hear it until the 3rd
> ring. I come running into my office but miss it by a split second. Is
> there a way I can barge in on the person leaving a message for my
> mailbox while they're leaving it?

I imagine that would be tricky to do once the call has been handed in to the 
voicemail application, as presently you are limited in what you can do to 
the call once it's gone in there. You might be able to locate the original 
SIP channel and bridge the calls, but I've no idea how you would track that 
properly. There is a way to make voicemail have a "press 0 to be transferred 
to somewhere else" option. We use that here and it works. Users can set up 
where they want the caller to be transferred to (usually a mobile) and then 
they can record on their outgoing message "leave me a message, or press 0 to 
try my mobile..."

Or, Sounds like a case for a few IP DECT cordless handsets to save all this 
running about! You might run into somebody carrying a boiling hot cup of 
coffee in your rush to answer the phone! (happens!)

We have a few Siemens C460 IP DECT Phones. The range and battery life on 
them is by far superior to any of the WiFi/SIP phones I've tried so far. I 
have a SIP/Wifi Nokia E65 that works great, but the battery life is not very 
good when the wifi is left on, and it was less than straightforward to set 
up!

> 2.       If a phone rings a receptionist desk, and the receptionist is
> down the hall, she wants to be able to dial an extension, and have that
> transfer the call from her desk to the phone she's currently on so she
> doesn't have to run to her desk. Is there a built in feature for this or
> do I have to code it?

There is a feature called pickup defined in features.conf:

pickupexten = *8

Restart asterisk if you need to change features.conf (in my experience just 
a reload when changing features.conf doesn't always work)

You then need to define your SIP/devices into pickup groups in sip.conf, for 
example:-

[500]
canreinvite=yes
nat=no
secret=...
dtmfmode=rfc2833
callgroup=1
pickupgroup=1

[501]
canreinvite=yes
nat=no
secret=...
dtmfmode=rfc2833
callgroup=1
pickupgroup=1

Then reload. Now, if extn 500 were ringing, picking up 501 and doing *8 will 
connect that call. 

For busier systems I believe there is a dialplan feature that enables 
"directed" pickup so you can pick up a specific extension, but I haven't 
played with that so I can't say how it works. That might be more suitable.

The callgroup defines what pickup group the device is in, and pickupgroup 
defines what groups (when that extension dials *8) that device can pick up.
A device can be in one callgroup but multiple pickup groups:-

pickupgroup=1,2

This is so that if you have many sites or departments, only people who sit 
within the range of the ringing phone can pick it up, and not get connected 
to some other random call incoming somewhere else.


Rob


-- 
Robert Lister     - London Internet Exchange - http://www.linx.net/
sip:robl at linx.net - inoc-dba:5459*710        - tel: +44 (0)20 7645 3510