Sponsor: VoiceMeUp - Corporate & Wholesale VoIP Services

VoIP Mailing List Archives
Mailing list archives for the VoIP community
 SearchSearch 

[asterisk-users] Transfer calls "on demand"


 
Post new topic   Reply to topic    VoIP Mailing List Archives Forum Index -> Asterisk Users
View previous topic :: View next topic  
Author Message
lucabert at lucabert.de
Guest





PostPosted: Tue Dec 29, 2015 1:54 am    Post subject: [asterisk-users] Transfer calls "on demand" Reply with quote

Hi list!

Right now I configured my Asterisk to forward the calls for the number X to
both phones (mine and the phone of my wife).
It works, of course, but I'm not enthusiast...

I see what we have at office: if one phone rings, other phones in the same
group can "catch the call", so that if a colleague is not present, another
colleague can catch the call.

I'd like to have the same procedure at home. I think, Asterisk can do that,
but I have no idea how to implement this.

Shortly: what I want is that every phone rings only on calls for the own
number, and I can catch the call from the other phone, if for example my wife
is not at home, for example pressing "*5#" or other key combination.

Thanks a lot for your suggestion!

Luca Bertoncello
(lucabert@lucabert.de)

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Back to top
lucabert at lucabert.de
Guest





PostPosted: Tue Dec 29, 2015 4:20 am    Post subject: [asterisk-users] Transfer calls "on demand" Reply with quote

Daniel Heckl <daniel.heckl@gmail.com> schrieb:

Quote:
You are searching for „Call Pickup“. It is implemented in Asterisk by
default.

https://wiki.asterisk.org/wiki/display/AST/Call+Pickup
<https://wiki.asterisk.org/wiki/display/AST/Call+Pickup> Take a look under
section „Configuration Options“.

Hi, Daniel!

Thanks for your answer...
I'm using Asterisk 1.8.30.0 on an OpenWRT-Router.
I found the configuration for call pickup in the sip.conf and features.conf,
so I tried to activate it...
Unfortunately, unsuccessfully...

So, my sip.conf:

callgroup=1,3-4 ; We are in caller groups 1,3,4
pickupgroup=1,3-5 ; We can do call pick-p for call group 1,3,4,5

my features.conf:

; Pickup Options
;
pickupexten = *8 ; Configure the pickup extension. (default is *Cool
;pickupsound = beep ; to indicate a successful pickup (default: no sound)
;pickupfailsound = beeperr ; to indicate that the pickup failed (default: no sound)

my users.conf:

[general]
callgroup = 1
pickupgroup = 1

my extensions.conf:

[anika_incoming]
exten => _00493512222222,1,Verbose(2,Call for Anika - [${CALLERID(num)}])
exten => _00493512222222,Set(CHANNEL(pickupgroup)=1)
exten => _00493512222222,n,Dial(local/2222222@anika_incoming)
exten => _03512222222,1,Verbose(2,Call for Anika - [${CALLERID(num)}])
exten => _03512222222,n,Dial(local/2222222@anika_incoming)
exten => _2222222,1,Verbose(2,Call for Anika - [${CALLERID(num)}])
exten => _2222222,n,Set(CALLERID(num)=${IF($[ "${CALLERID(num):0:3}" = "+49" ]?0${CALLERID(num):3}:${CALLERID(num)})}) ; Damit das "+49" mit "0" ersetzt wird
exten => _2222222,n,Set(CHANNEL(musicclass)=default)
;;;exten => _2222222,n,Dial(SIP/00493512222222&local/1@luca_for_anika_voip_mobile,19,RcxX)
exten => _2222222,n,Dial(SIP/00493512222222,19,RcxX)
exten => _2222222,n,Verbose(2,Voicemail for Anika)
exten => _2222222,n,Set(CALLERID(name)=) ; Damit in der E-Mail der AB nicht den Namen steht
exten => _2222222,n,VoiceMail(00493512222222,us)
exten => _2222222,n,Hangup

Then I called the 2222222 with my mobile phone and I tried to get the call
from the other phone, calling the *8.
Unfortunately I get an error (invalid number) on the display of the phone,
and the phone 2222222 continue to ring.
No error on the log of Asterisk...

Any suggestion?

Thanks
Luca Bertoncello
(lucabert@lucabert.de)

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Back to top
support at drdos.info
Guest





PostPosted: Tue Dec 29, 2015 4:52 am    Post subject: [asterisk-users] Transfer calls "on demand" Reply with quote

Luca Bertoncello wrote:
Quote:
So, my sip.conf:

callgroup=1,3-4 ; We are in caller groups 1,3,4
pickupgroup=1,3-5 ; We can do call pick-p for call group 1,3,4,5

Keep it simple for testing. My sip.conf on a working Asterisk system below:

[4220](stemplet)
defaultuser=4220
pickupgroup=1
callgroup=1
callerid = Operator <4220>

[4232](stemplet)
defaultuser=4232
mailbox = 4232@sip
context = sip
callgroup=1
pickupgroup=1
callerid = Judy K. <4232>


My features.conf:

; Pickup Options
;
pickupexten = *7 ; Configure the pickup extension.
(default is *Cool

This has worked for me on all version of Asterisk that we've used; we're
currently running 11. In the above, The operator can pickup calls for
Judy and vise versa using *7.

You'll also want to make sure your phone's digit map understands that *7
is a valid key combination. We're using Polycom phones.

Doug


--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Back to top
lucabert at lucabert.de
Guest





PostPosted: Tue Dec 29, 2015 5:09 am    Post subject: [asterisk-users] Transfer calls "on demand" Reply with quote

Doug Lytle <support@drdos.info> schrieb:

Quote:
Keep it simple for testing. My sip.conf on a working Asterisk system below:

IT WORKS!!!

Thanks a lot!
Luca Bertoncello
(lucabert@lucabert.de)

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Back to top
Display posts from previous:   
Post new topic   Reply to topic    VoIP Mailing List Archives Forum Index -> Asterisk Users All times are GMT - 5 Hours
Page 1 of 1

 
Jump to:  
You cannot post new topics in this forum
You cannot reply to topics in this forum
You cannot edit your posts in this forum
You cannot delete your posts in this forum
You cannot vote in polls in this forum


Powered by phpBB © 2001, 2005 phpBB Group

VoiceMeUp - Corporate & Wholesale VoIP Services