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

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

[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&local/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)

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

[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&local/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)
> 
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Re: [asterisk-users] Transfer calls "on demand"

[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


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

[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)

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[asterisk-users] AMI StarAstAPI.php not working for commands like SIPpeers with more output

[Thomas]

Hi,
Iam using StarAstAPI.php.
If Iam sending Action commands like 'reload' everythink works fine.

If I send an Action like 'SIPpeers' I get:

  ["Response:"]=>
  string(8) " Success"
  ["ActionID:"]=>
  string(2) " 2"
  ["EventList:"]=>
  string(6) " start"
  ["Message:"]=>
  string(29) " Peer status list will follow"


But how I receive the peer list?

best regards
Thomas





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[asterisk-users] Looking for call queues data

[Tech Support]

All;

I'm working on a project that analyzes Asterisk call queue data that I
plan on releasing as open source. The project is going well and when
finished, will be very comprehensive. What I am looking for is actual call
queue data and its associated CDR's. The busier the better. The data will go
absolutely nowhere, will be deleted when finished,  and I have no problem
anonymizing the data if needed. If anyone feels that they want to help out,
please shoot me an email at supp...@voipbusiness.us. Like I said, this is an
open source project.

Thanks in Advance;

John V.  

 

Tech Support

Tech Support

VoIP Business Solutions

240-215-3479 x325

  supp...@voipbusiness.us

 

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[asterisk-users] Signaling ringing on other extension

[Luca Bertoncello]

Hi again!

With the "call pickup"-function I can now pickup a call directed to another
phone in my Asterisk. Very nice.
My problem, now, is that I can't see on my phone, that the other phone (in
another room) rings.

Is it possible to signal the incoming call on other extension? I use two
phones "Thomson ST2022".

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

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[asterisk-users] Questions about API Asterisk Java

[pierre.guyard]

Hello everyone,

I have some questions about one of the Asterisk API called Asterisk Java:


* How did it work?

* How can we use it in order to connect one external program to 
Asterisk?

* Can we use it with Asterisk 12?

Thank you in advance,
Pierre

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