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lucabert at lucabert.de Guest
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Posted: Tue Dec 29, 2015 1:54 am Post subject: [asterisk-users] Transfer calls "on demand" |
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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)
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lucabert at lucabert.de Guest
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Posted: Tue Dec 29, 2015 4:20 am Post subject: [asterisk-users] Transfer calls "on demand" |
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Daniel Heckl <daniel.heckl@gmail.com> schrieb:
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 *
;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)
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support at drdos.info Guest
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Posted: Tue Dec 29, 2015 4:52 am Post subject: [asterisk-users] Transfer calls "on demand" |
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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
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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 *
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
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lucabert at lucabert.de Guest
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Posted: Tue Dec 29, 2015 5:09 am Post subject: [asterisk-users] Transfer calls "on demand" |
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Doug Lytle <support@drdos.info> schrieb:
Quote: | Keep it simple for testing. My sip.conf on a working Asterisk system below:
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IT WORKS!!!
Thanks a lot!
Luca Bertoncello
(lucabert@lucabert.de)
--
_____________________________________________________________________
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