Re: [asterisk-users] Transfer calls "on demand"

2015-12-29 Thread Daniel Heckl
You are searching for „Call Pickup“. It is implemented in Asterisk by default.

https://wiki.asterisk.org/wiki/display/AST/Call+Pickup 

Take a look under section „Configuration Options“.

Daniel

> Am 29.12.2015 um 07:53 schrieb Luca Bertoncello :
> 
> 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
> (lucab...@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



smime.p7s
Description: S/MIME cryptographic signature
-- 
_
-- 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

Re: [asterisk-users] Transfer calls "on demand"

2015-12-29 Thread Luca Bertoncello
Daniel Heckl  schrieb:

> You are searching for „Call Pickup“. It is implemented in Asterisk by
> default.
> 
> 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 *8)
;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 => _0049351222,1,Verbose(2,Call for Anika - [${CALLERID(num)}])
exten => _0049351222,Set(CHANNEL(pickupgroup)=1)
exten => _0049351222,n,Dial(local/222@anika_incoming)
exten => _0351222,1,Verbose(2,Call for Anika - [${CALLERID(num)}])
exten => _0351222,n,Dial(local/222@anika_incoming)
exten => _222,1,Verbose(2,Call for Anika - [${CALLERID(num)}])
exten => _222,n,Set(CALLERID(num)=${IF($[ "${CALLERID(num):0:3}" = "+49" 
]?0${CALLERID(num):3}:${CALLERID(num)})})  ; Damit das "+49" mit "0" ersetzt 
wird
exten => _222,n,Set(CHANNEL(musicclass)=default)
;;;exten => 
_222,n,Dial(SIP/0049351222/1@luca_for_anika_voip_mobile,19,RcxX)
exten => _222,n,Dial(SIP/0049351222,19,RcxX)
exten => _222,n,Verbose(2,Voicemail for Anika)
exten => _222,n,Set(CALLERID(name)=)   
; Damit in der E-Mail der AB nicht den Namen steht
exten => _222,n,VoiceMail(0049351222,us)
exten => _222,n,Hangup

Then I called the 222 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 222 continue to ring.
No error on the log of Asterisk...

Any suggestion?

Thanks
Luca Bertoncello
(lucab...@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

Re: [asterisk-users] Transfer calls "on demand"

2015-12-29 Thread Daniel Heckl
On top of the page: "Call pickup support added in Asterisk 11“

I think that is the problem. I do not know a solution for 1.8, but maybe 
someone other.

> Am 29.12.2015 um 10:20 schrieb Luca Bertoncello :
> 
> Daniel Heckl  schrieb:
> 
>> You are searching for „Call Pickup“. It is implemented in Asterisk by
>> default.
>> 
>> 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 
> *8)
> ;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 => _0049351222,1,Verbose(2,Call for Anika - [${CALLERID(num)}])
> exten => _0049351222,Set(CHANNEL(pickupgroup)=1)
> exten => _0049351222,n,Dial(local/222@anika_incoming)
> exten => _0351222,1,Verbose(2,Call for Anika - [${CALLERID(num)}])
> exten => _0351222,n,Dial(local/222@anika_incoming)
> exten => _222,1,Verbose(2,Call for Anika - [${CALLERID(num)}])
> exten => _222,n,Set(CALLERID(num)=${IF($[ "${CALLERID(num):0:3}" = "+49" 
> ]?0${CALLERID(num):3}:${CALLERID(num)})})  ; Damit das "+49" mit "0" ersetzt 
> wird
> exten => _222,n,Set(CHANNEL(musicclass)=default)
> ;;;exten => 
> _222,n,Dial(SIP/0049351222/1@luca_for_anika_voip_mobile,19,RcxX)
> exten => _222,n,Dial(SIP/0049351222,19,RcxX)
> exten => _222,n,Verbose(2,Voicemail for Anika)
> exten => _222,n,Set(CALLERID(name)=)  
>  ; Damit in der E-Mail der AB nicht den Namen steht
> exten => _222,n,VoiceMail(0049351222,us)
> exten => _222,n,Hangup
> 
> Then I called the 222 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 222 continue to ring.
> No error on the log of Asterisk...
> 
> Any suggestion?
> 
> Thanks
> Luca Bertoncello
> (lucab...@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



smime.p7s
Description: S/MIME cryptographic signature
-- 
_
-- 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

Re: [asterisk-users] Transfer calls "on demand"

2015-12-29 Thread Doug Lytle

Luca Bertoncello wrote:

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 *8)


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


Re: [asterisk-users] Transfer calls "on demand"

2015-12-29 Thread Luca Bertoncello
Doug Lytle  schrieb:

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

IT WORKS!!!

Thanks a lot!
Luca Bertoncello
(lucab...@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


[asterisk-users] Transfer calls "on demand"

2015-12-28 Thread Luca Bertoncello
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
(lucab...@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