Re: [asterisk-users] Call forwarding in Asterisk

[Julian Beach]

Hello Kantharuban,

Friday, September 4, 2015, 8:19:28 AM, you wrote:

> Thanks for your info, What is the impact of the following line in
> dialpla Dial(SIP/19201/19202,300)

It  does  not  look like a valid format. If you are trying to dial two
SIP  devices  (19201  and  19202)  with  a timeout of 300 seconds, the
command would be

Dial(SIP/19201/19202,300)  and  you might want to consider some of
the  option  Dial options depending on what you do with the call after
it has been answered.

Have  a  look  at http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial
for  details  of  the  dial command, and the options or have a look at
Asterisk:  The  Definitive  Guide  which  will  tell  you  more  about
Originate and Local Channels, which you might also find useful.

http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/index.html

J

-- 
Best regards,
 Julianmailto:jb_s...@trink.co.uk


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Re: [asterisk-users] Call forwarding in Asterisk

[Kantharuban Ruban]

Hi ,
 I have gone through the link you have sent me , there i could find the
below lines,

*Dial() together with openining Jack ports for callee*






*Nescesarry if you want to "capture" a record in leg B with SoundPatty
exten =>
_X.,n,Dial(SIP/$PROVIDER/${EXTEN},60,M(connect-jack)[macro-connect-jack]exten
=> s,1,NoOp(${CHANNEL}) ; This is leg A, skipexten =>
s,2,Set(JACK_HOOK(manipulate,i(${CHANNEL}:input),o(${CHANNEL}:output))=on)Note:
only for asterisk 1.6.x*

Could you please tell me what does it do?


On Fri, Sep 4, 2015 at 2:56 PM, Julian Beach  wrote:

> Hello Kantharuban,
>
> Friday, September 4, 2015, 8:19:28 AM, you wrote:
>
> > Thanks for your info, What is the impact of the following line in
> > dialpla Dial(SIP/19201/19202,300)
>
> It  does  not  look like a valid format. If you are trying to dial two
> SIP  devices  (19201  and  19202)  with  a timeout of 300 seconds, the
> command would be
>
> Dial(SIP/19201/19202,300)  and  you might want to consider some of
> the  option  Dial options depending on what you do with the call after
> it has been answered.
>
> Have  a  look  at http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial
> for  details  of  the  dial command, and the options or have a look at
> Asterisk:  The  Definitive  Guide  which  will  tell  you  more  about
> Originate and Local Channels, which you might also find useful.
>
>
> http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/index.html
>
> J
>
> --
> Best regards,
>  Julianmailto:jb_s...@trink.co.uk
>
>
> --
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>



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*Ruban.S*
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Re: [asterisk-users] Call forwarding in Asterisk

[Kantharuban Ruban]

Hi,
Thanks for your info, What is the impact of the following line in
dialplan,

Dial(SIP/19201/19202,300)





On Thu, Sep 3, 2015 at 7:20 PM, Vinicius Fontes 
wrote:

> You might want to use the Originate() application instead. Check its usage
> by issuing the command 'core show application originate' on Asterisk CLI.
>
> 2015-09-03 9:09 GMT-03:00 Kantharuban Ruban :
>
>> Hello Group,
>>
>> I have a requirement to dialout some external number, once
>> the call is answered the same has to be forwarded to an Internal Queue.
>>
>> Please help me.
>>
>> I have tried calling with two SIP end point forwarding , even that is not
>> working,
>>
>> My dial plan line is , Dial(SIP/19201/19202,300)
>>
>>
>> --
>> *Best regards,*
>> *Ruban.S*
>>
>> --
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>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
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Re: [asterisk-users] Call forwarding in Asterisk

[Vinicius Fontes]

You might want to use the Originate() application instead. Check its usage
by issuing the command 'core show application originate' on Asterisk CLI.

2015-09-03 9:09 GMT-03:00 Kantharuban Ruban :

> Hello Group,
>
> I have a requirement to dialout some external number, once the
> call is answered the same has to be forwarded to an Internal Queue.
>
> Please help me.
>
> I have tried calling with two SIP end point forwarding , even that is not
> working,
>
> My dial plan line is , Dial(SIP/19201/19202,300)
>
>
> --
> *Best regards,*
> *Ruban.S*
>
> --
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>http://www.asterisk.org/hello
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> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [asterisk-users] Call forwarding from Phones and getting the referrer IP

[Ishfaq Malik]

On 24 October 2014 16:51, Ishfaq Malik i...@pack-net.co.uk wrote:

 Hi

 I'm using asterisk 1.8 but I'm sure this applies to other versions.

 If someone puts a call divert on a handset such as a Snom phone I get this
 type of SIP message on receipt of an inbound call:

 Got SIP response 302 Moved Temporarily back from xxx.xxx.xxx.xxx:x

 Which then triggers a local channel to make the call.

 Is there any way I can access that IP address inside my dialplan? I've
 done a ChanDump and there's no sign of it.

 Regards

 Ish


Bumping this as I originally sent it late on Friday. If anyone has any
idea, please let me know.


Thanks in Advance

Ish
-- 

Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http://www.pack-net.co.uk

Registered Address: PACKNET LIMITED, Duplex 2, Ducie House
37 Ducie Street
Manchester, M1 2JW
COMPANY REG NO. 04920552
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Re: [asterisk-users] Call forwarding from Phones and getting the referrer IP

[Scott Griepentrog]

After a quick perusal of the chan_sip.c code (from svn trunk), I'm not
seeing where the address (p-sa) logged in that message is passed to the
redirecting functions handling the 302, thus it is unlikely there is a way
to obtain it other than reading the log.

It wouldn't be hard to set a channel variable with that value however,
should you want to patch the code, possibly even submit that.


On Tue, Oct 28, 2014 at 7:05 AM, Ishfaq Malik i...@pack-net.co.uk wrote:

 On 24 October 2014 16:51, Ishfaq Malik i...@pack-net.co.uk wrote:

 Hi

 I'm using asterisk 1.8 but I'm sure this applies to other versions.

 If someone puts a call divert on a handset such as a Snom phone I get
 this type of SIP message on receipt of an inbound call:

 Got SIP response 302 Moved Temporarily back from xxx.xxx.xxx.xxx:x

 Which then triggers a local channel to make the call.

 Is there any way I can access that IP address inside my dialplan? I've
 done a ChanDump and there's no sign of it.

 Regards

 Ish


 Bumping this as I originally sent it late on Friday. If anyone has any
 idea, please let me know.


 Thanks in Advance

 Ish
 --

 Ishfaq Malik
 Department: VOIP Support
 Company: Packnet Limited
 t: +44 (0)845 004 4994
 f: +44 (0)161 660 9825
 e: i...@pack-net.co.uk
 w: http://www.pack-net.co.uk

 Registered Address: PACKNET LIMITED, Duplex 2, Ducie House
 37 Ducie Street
 Manchester, M1 2JW
 COMPANY REG NO. 04920552


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[image: Digium logo]
Scott Griepentrog
Digium, Inc · Software Developer
445 Jan Davis Drive NW · Huntsville, AL 35806 · US
direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090
Check us out at: http://digium.com · http://asterisk.org
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Re: [asterisk-users] Call Forwarding / Follow-Me on PRI

[Carlos Alvarez]

On Thu, Dec 27, 2012 at 2:41 PM, Barry D. Hassler
barry.hass...@gmail.comwrote:

 Friends,

 Curious if others have run into this scenario, and can shed further light
 on it. I am working with an installed base of systems using PRI circuits
 from several carriers, and the symptoms I relate occur across the board.


We have encountered it, and simply told the carriers to stop blocking it or
lose the business.  All but one did it, and we dropped their services.
 Don't know that there's a good work-around otherwise.

Is there a reason you don't just go all SIP, where 98% of the service
providers will accept any CLID?

-- 
Carlos Alvarez
TelEvolve
602-889-3003
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Re: [asterisk-users] call forwarding number from outside.

[Perenaster]

Hello,
I have a similar problem. Whenever a call comes in to my asterisk I handle
it like this:

exten = s,1, Answer()
exten = s, n, Dial(SIP/exten,20,fotT)
exten = s, 1, Hangup()

it works fine but in the SIP messages th IP-Address from Asterisk is in the
From field. For example I am calling from 123@134.32.220.33 then the SIP
message behind the Asterisk looks like

INVITE sip:2232@10.10.10.11 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.10 (Asterisk IP)
From: 123 sip:soft@10.10.10.10 (again Asterisk IP)


how can I change this?
Thanks
Tom

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sip:3...@perenaster.com
sip:3...@perenaster.com
sip:3...@perenaster.com
sip:3...@perenaster.com
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Re: [asterisk-users] call forwarding number from outside.

[Mike]

That`s the normal behavior of assisted transfers.  Try a blind/non-assisted
transfer, that should show the original callerid.

 

Mike

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alessio
Sent: Friday, July 29, 2011 2:56 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] call forwarding number from outside.

 

Hi!

I need help regarding the following problem:

when I receive a phone call to the PBX from the number 01234567890
rings the number 100, get up the phone, I transfer (assisted) to the number
100.
When the 100 number rings, the display shows the number of those who
transferred the call and not the number 01234567890.

How can you solve this problem?

Thanks and sorry for my English 

 

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Re: [asterisk-users] call forwarding number from outside.

[Alessio]

Thanks for the reply!

I've tried and works, but isn't possible with the transfer assisted?

thanks


From: Mike 
Sent: Friday, July 29, 2011 8:58 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion' 
Subject: Re: [asterisk-users] call forwarding number from outside.


That`s the normal behavior of assisted transfers.  Try a blind/non-assisted 
transfer, that should show the original callerid.

 

Mike

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alessio
Sent: Friday, July 29, 2011 2:56 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] call forwarding number from outside.

 

Hi!

I need help regarding the following problem:

when I receive a phone call to the PBX from the number 01234567890
rings the number 100, get up the phone, I transfer (assisted) to the number 100.
When the 100 number rings, the display shows the number of those who 
transferred the call and not the number 01234567890.

How can you solve this problem?

Thanks and sorry for my English 

 






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Re: [asterisk-users] call forwarding number from outside.

[Danny Nicholas]

The issue with assisted transfer is that the assisting transferer is a
second call

Outside - A

A answers

A calls B to tell them they have a call (call #2 with ID of A

A transfers Outside but the ID stays A

 

Blind Transfer

Outside - A

A answers

A blind transfers to B (1 call - keeps ID.

 

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alessio
Sent: Friday, July 29, 2011 2:10 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] call forwarding number from outside.

 

Thanks for the reply!

 

I've tried and works, but isn't possible with the transfer assisted?

 

thanks

 

From: Mike mailto:l...@net-wall.com  

Sent: Friday, July 29, 2011 8:58 AM

To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
mailto:asterisk-users@lists.digium.com  

Subject: Re: [asterisk-users] call forwarding number from outside.

 

That`s the normal behavior of assisted transfers.  Try a blind/non-assisted
transfer, that should show the original callerid.

 

Mike

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alessio
Sent: Friday, July 29, 2011 2:56 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] call forwarding number from outside.

 

Hi!

I need help regarding the following problem:

when I receive a phone call to the PBX from the number 01234567890
rings the number 100, get up the phone, I transfer (assisted) to the number
100.
When the 100 number rings, the display shows the number of those who
transferred the call and not the number 01234567890.

How can you solve this problem?

Thanks and sorry for my English 

 

  _  

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Re: [asterisk-users] call forwarding number from outside.

[Eric Wieling]

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
 boun...@lists.digium.com] On Behalf Of Danny Nicholas
 Sent: Friday, July 29, 2011 9:06 AM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Cc: jim.smith...@debsinc.com
 Subject: Re: [asterisk-users] call forwarding number from outside.
 
 The issue with assisted transfer is that the assisting transferer is a 
 second
 call
 
 Outside - A
 
 A answers
 
 A calls B to tell them they have a call (call #2 with ID of A
 
 A transfers Outside but the ID stays A
 
 
 
 Blind Transfer
 
 Outside - A
 
 A answers
 
 A blind transfers to B (1 call - keeps ID.
 

From the output of core show application dial:

f: Force the callerid of the *calling* channel to be set as the
extension associated with the channel using a dialplan 'hint'. For example,
some PSTNs do not allow CallerID to be set to anything other than the
number assigned to the caller.


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Re: [asterisk-users] call forwarding number from outside.

[Kevin P. Fleming]

On 07/29/2011 09:12 AM, Eric Wieling wrote:



-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Friday, July 29, 2011 9:06 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Cc: jim.smith...@debsinc.com
Subject: Re: [asterisk-users] call forwarding number from outside.

The issue with assisted transfer is that the assisting transferer is a second
call

Outside -  A

A answers

A calls B to tell them they have a call (call #2 with ID of A

A transfers Outside but the ID stays A



Blind Transfer

Outside -  A

A answers

A blind transfers to B (1 call - keeps ID.



 From the output of core show application dial:

 f: Force the callerid of the *calling* channel to be set as the
 extension associated with the channel using a dialplan 'hint'. For example,
 some PSTNs do not allow CallerID to be set to anything other than the
 number assigned to the caller.


In Asterisk 1.8 and later, if the phones (endpoints) support it, the 
connected party display on the phone will update *after* the transfer 
has been completed to show who the person is talking to (not the person 
who performed the transfer).


--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com  www.asterisk.org

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Re: [asterisk-users] call forwarding number from outside.

[Danny Nicholas]

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P.
Fleming
Sent: Friday, July 29, 2011 8:49 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] call forwarding number from outside.

On 07/29/2011 09:12 AM, Eric Wieling wrote:

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- 
 boun...@lists.digium.com] On Behalf Of Danny Nicholas
 Sent: Friday, July 29, 2011 9:06 AM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Cc: jim.smith...@debsinc.com
 Subject: Re: [asterisk-users] call forwarding number from outside.

 The issue with assisted transfer is that the assisting transferer 
 is a second call

 Outside -  A

 A answers

 A calls B to tell them they have a call (call #2 with ID of A

 A transfers Outside but the ID stays A



 Blind Transfer

 Outside -  A

 A answers

 A blind transfers to B (1 call - keeps ID.


  From the output of core show application dial:

  f: Force the callerid of the *calling* channel to be set as the
  extension associated with the channel using a dialplan 'hint'. For
example,
  some PSTNs do not allow CallerID to be set to anything other than the
  number assigned to the caller.

In Asterisk 1.8 and later, if the phones (endpoints) support it, the
connected party display on the phone will update *after* the transfer has
been completed to show who the person is talking to (not the person who
performed the transfer).

--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at
www.digium.com  www.asterisk.org

Couple of questions - 
This magic trick is contained in app_dial?
Functionality is inherent to 1.8/10.X structure so we can't re-invent this
in our old 1.4/1.6 installs?



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Re: [asterisk-users] call forwarding number from outside.

[Kevin P. Fleming]

On 07/29/2011 10:41 AM, Danny Nicholas wrote:

snip


Couple of questions -
This magic trick is contained in app_dial?
Functionality is inherent to 1.8/10.X structure so we can't re-invent this
in our old 1.4/1.6 installs?


No, it's core functionality, implemented in the channel drivers and 
using control frames that pass through bridges. It would be a large 
amount of effort to implement it again in 1.4/1.6. It extends well 
beyond simple dialing, as it can receive updates across external 
protocols and pass them along, it handles call redirection, and various 
other features.


--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com  www.asterisk.org

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Re: [asterisk-users] call forwarding number from outside.

[Danny Nicholas]

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P.
Fleming
Sent: Friday, July 29, 2011 9:48 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] call forwarding number from outside.

On 07/29/2011 10:41 AM, Danny Nicholas wrote:

snip

 Couple of questions -
 This magic trick is contained in app_dial?
 Functionality is inherent to 1.8/10.X structure so we can't re-invent 
 this in our old 1.4/1.6 installs?

No, it's core functionality, implemented in the channel drivers and using
control frames that pass through bridges. It would be a large amount of
effort to implement it again in 1.4/1.6. It extends well beyond simple
dialing, as it can receive updates across external protocols and pass them
along, it handles call redirection, and various other features.

--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at
www.digium.com  www.asterisk.org

As I suspected sigh


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Re: [asterisk-users] call forwarding number from outside.

[Alessio]

So I can't do anything?

--
From: Kevin P. Fleming kpflem...@digium.com
Sent: Friday, July 29, 2011 4:48 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] call forwarding number from outside.


On 07/29/2011 10:41 AM, Danny Nicholas wrote:

snip


Couple of questions -
This magic trick is contained in app_dial?
Functionality is inherent to 1.8/10.X structure so we can't re-invent 
this

in our old 1.4/1.6 installs?


No, it's core functionality, implemented in the channel drivers and using 
control frames that pass through bridges. It would be a large amount of 
effort to implement it again in 1.4/1.6. It extends well beyond simple 
dialing, as it can receive updates across external protocols and pass them 
along, it handles call redirection, and various other features.


--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com  www.asterisk.org

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Re: [asterisk-users] call forwarding number from outside.

[Danny Nicholas]

Upgrade to 1.8/10.0

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alessio
Sent: Friday, July 29, 2011 10:04 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] call forwarding number from outside.

So I can't do anything?

--
From: Kevin P. Fleming kpflem...@digium.com
Sent: Friday, July 29, 2011 4:48 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] call forwarding number from outside.

 On 07/29/2011 10:41 AM, Danny Nicholas wrote:

 snip

 Couple of questions -
 This magic trick is contained in app_dial?
 Functionality is inherent to 1.8/10.X structure so we can't re-invent 
 this in our old 1.4/1.6 installs?

 No, it's core functionality, implemented in the channel drivers and 
 using control frames that pass through bridges. It would be a large 
 amount of effort to implement it again in 1.4/1.6. It extends well 
 beyond simple dialing, as it can receive updates across external 
 protocols and pass them along, it handles call redirection, and various
other features.

 --
 Kevin P. Fleming
 Digium, Inc. | Director of Software Technologies
 Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: 
 kpfleming
 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at 
 www.digium.com  www.asterisk.org

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Re: [asterisk-users] call forwarding number from outside.

[Alessio]

ok I'll do it Monday, and how you handle it with the version 1.10?

Thanks

--
From: Danny Nicholas da...@debsinc.com
Sent: Friday, July 29, 2011 5:05 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion' 
asterisk-users@lists.digium.com

Subject: Re: [asterisk-users] call forwarding number from outside.


Upgrade to 1.8/10.0

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alessio
Sent: Friday, July 29, 2011 10:04 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] call forwarding number from outside.

So I can't do anything?

--
From: Kevin P. Fleming kpflem...@digium.com
Sent: Friday, July 29, 2011 4:48 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] call forwarding number from outside.


On 07/29/2011 10:41 AM, Danny Nicholas wrote:

snip


Couple of questions -
This magic trick is contained in app_dial?
Functionality is inherent to 1.8/10.X structure so we can't re-invent
this in our old 1.4/1.6 installs?


No, it's core functionality, implemented in the channel drivers and
using control frames that pass through bridges. It would be a large
amount of effort to implement it again in 1.4/1.6. It extends well
beyond simple dialing, as it can receive updates across external
protocols and pass them along, it handles call redirection, and various

other features.


--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype:
kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at
www.digium.com  www.asterisk.org

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Re: [asterisk-users] call forwarding

[salaheddine elharit]

Hi Rizwan



Thank you for your help i will test this solution and i will update you as
soon as i have any result.



Kind Regards

2011/4/4 Rizwan Hisham rizwanhas...@gmail.com

 Do this:

 exten= _0522XX,1,Goto(${CONTEXT},0520${EXTEN:4:},1)

 you can also use the dial command for this as well

 exten= _0522XX,1,Dial(Local/0520${EXTEN:4}@${SOMECONTEXT})

 replace ${CoNTEXT} and ${SOMECONTEXT} with name of your contexts which
 contains 0520 numbers.

 I have not tested it, you can try it on your setup.


   On Mon, Apr 4, 2011 at 7:00 PM, salaheddine elharit 
 salah.elharit...@gmail.com wrote:

   Hello list,

 i have one question related to call forwarding.

 i have 2 number for the inbound and i want to configure asterisk like
 that.

 When the customer call the first number 0522XX the call will be
 forwarding automatically to anther number 0520xx

 Does anybody have a solution to this problem.

 Thanks and Regards.

 --
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 --
 Best Ragards
 Rizwan Qureshi
 VoIP/Asterisk Engineer
 Axvoice Inc.

 V: +92 (0)  6767 26
 E: rizwanhas...@gmail.com
 W: www.axvoice.com


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Re: [asterisk-users] call forwarding

[Rizwan Hisham]

Do this:

exten= _0522XX,1,Goto(${CONTEXT},0520${EXTEN:4:},1)

you can also use the dial command for this as well

exten= _0522XX,1,Dial(Local/0520${EXTEN:4}@${SOMECONTEXT})

replace ${CoNTEXT} and ${SOMECONTEXT} with name of your contexts which
contains 0520 numbers.

I have not tested it, you can try it on your setup.


On Mon, Apr 4, 2011 at 7:00 PM, salaheddine elharit 
salah.elharit...@gmail.com wrote:

 Hello list,

 i have one question related to call forwarding.

 i have 2 number for the inbound and i want to configure asterisk like that.

 When the customer call the first number 0522XX the call will be
 forwarding automatically to anther number 0520xx

 Does anybody have a solution to this problem.

 Thanks and Regards.

 --
 _
 -- 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




-- 
Best Ragards
Rizwan Qureshi
VoIP/Asterisk Engineer
Axvoice Inc.

V: +92 (0)  6767 26
E: rizwanhas...@gmail.com
W: www.axvoice.com
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Re: [asterisk-users] call forwarding callerID

[Gerard]

We have a T1 of sorts, ATT ip flex reach basically voip over a t1 
line i think. I will ask them and see what they say, I'm already able to 
set our outgoing callerID to any number we own, just no other ones..

 there some other way to handle this?

 It depends on the technology and the carrier.

 A simple POTS line and you're out of luck.

 If you have a T1 (i.e. ISDN-PRI) and a co-operative carrier, it may just
 work or they may enable it if requested.

 You could always use a co-operative SIP carrier (like Vitelity). A penny
 or 2 per minute will keep your someone happy.



-- 
Gerard Saraber
Network Admin.
Rarcoa, Inc
(630) 654-2580 x199
(630) 654-3556 (fax)
(630) 915-4122 (cell)

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Re: [asterisk-users] call forwarding callerID

[Gerard]

On 10/13/10 14:52, Danny Nicholas wrote:
 I think FOLLOWME is going to fix this for you

Can you elaborate please? is this a feature from our carrier? or 
something that will be built into asterisk? sounds like a useful fix :)

-- 
Gerard Saraber
Network Admin.
Rarcoa, Inc
(630) 654-2580 x199
(630) 654-3556 (fax)
(630) 915-4122 (cell)

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Re: [asterisk-users] call forwarding callerID

[Danny Nicholas]

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gerard
Sent: Thursday, October 14, 2010 11:24 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] call forwarding callerID

On 10/13/10 14:52, Danny Nicholas wrote:
 I think FOLLOWME is going to fix this for you

Can you elaborate please? is this a feature from our carrier? or 
something that will be built into asterisk? sounds like a useful fix :)

-- 
Gerard Saraber
Network Admin.
Rarcoa, Inc
(630) 654-2580 x199
(630) 654-3556 (fax)
(630) 915-4122 (cell)

Check this link
http://www.voip-info.org/wiki/view/Asterisk+cmd+FollowMe

A simpler solution (perhaps) would be a forwarding context like this

[forward-with-announce]
Exten = s,1,dial(DAHDI/g1/w${ARG1},30,mKkTt)
Exten = s,n,playback(followme/call-from)
Exten = s,n,SayDigits(${ARG2})

Exten = 393,1,Set(ARG1=201212)
Exten = 393,2,Set(ARG2=${EXTEN})
Exten = 393,3,Goto(forward-with-announce,s,1)

Dependent on carrier and other considerations, you can also spoof the
caller-id.  That's a different google-search.


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Re: [asterisk-users] call forwarding callerID

[Gerard]

ah-ha,
thank you very much, that's what I found when googling, I'll ask my user 
and see if Asterisk announcing the call is acceptable to him, if I can't 
spoof the callerID.

Followme would alternatively work pretty well, press 1 to accept the 
call etc. is a pretty nice feature, I'll see if that works for him.

Thanks!

On 10/14/10 11:41, Danny Nicholas wrote:
 Check this link
 http://www.voip-info.org/wiki/view/Asterisk+cmd+FollowMe

 A simpler solution (perhaps) would be a forwarding context like this

 [forward-with-announce]
 Exten =  s,1,dial(DAHDI/g1/w${ARG1},30,mKkTt)
 Exten =  s,n,playback(followme/call-from)
 Exten =  s,n,SayDigits(${ARG2})

 Exten =  393,1,Set(ARG1=201212)
 Exten =  393,2,Set(ARG2=${EXTEN})
 Exten =  393,3,Goto(forward-with-announce,s,1)

 Dependent on carrier and other considerations, you can also spoof the
 caller-id.  That's a different google-search.


-- 
Gerard Saraber
Network Admin.
Rarcoa, Inc
(630) 654-2580 x199
(630) 654-3556 (fax)
(630) 915-4122 (cell)

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Re: [asterisk-users] call forwarding callerID

[Danny Nicholas]

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gerard
Sent: Thursday, October 14, 2010 1:47 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] call forwarding callerID

ah-ha,
thank you very much, that's what I found when googling, I'll ask my user 
and see if Asterisk announcing the call is acceptable to him, if I can't 
spoof the callerID.

Followme would alternatively work pretty well, press 1 to accept the 
call etc. is a pretty nice feature, I'll see if that works for him.

Thanks!

On 10/14/10 11:41, Danny Nicholas wrote:
 Check this link
 http://www.voip-info.org/wiki/view/Asterisk+cmd+FollowMe

 A simpler solution (perhaps) would be a forwarding context like this

 [forward-with-announce]
 Exten =  s,1,dial(DAHDI/g1/w${ARG1},30,mKkTt)
 Exten =  s,n,playback(followme/call-from)
 Exten =  s,n,SayDigits(${ARG2})

 Exten =  393,1,Set(ARG1=201212)
 Exten =  393,2,Set(ARG2=${EXTEN})
 Exten =  393,3,Goto(forward-with-announce,s,1)

 Dependent on carrier and other considerations, you can also spoof the
 caller-id.  That's a different google-search.


-- 
Gerard Saraber
Network Admin.
Rarcoa, Inc
(630) 654-2580 x199
(630) 654-3556 (fax)
(630) 915-4122 (cell)

-- 
FWIW, there are also professional spoofing services but they cost
$0.02-$0.05/minute.


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Re: [asterisk-users] call forwarding callerID

[Danny Nicholas]

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gerard
Sent: Wednesday, October 13, 2010 2:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] call forwarding callerID

Hi list,
This is not necessarily an asterisk issue, but a lot of you guys know 
way more then me, so I have a question:
someone at my company sets his phone to forward calls to his cellphone, 
so someone calls our office, call is forwarded to his cell, and the 
callerID that shows up on his cell is of course our office number, 
because asterisk originates a new call to his cell and then bridges the two.
so he told me, a partner of his, at his office does the same thing, and 
when he does it, the callerID shows up as coming from the initial 
caller, not from his office.

so here's the schematic:
customer - our office ---callforward-- cellphone

so should I call ATT and ask them to unlock our callerID so I can set 
the outgoing callerID to the customer's number in my dialplan? or is 
there some other way to handle this?

I appreciate any input,
Thanks!
-- 
Gerard Saraber
Network Admin.
Rarcoa, Inc
(630) 654-2580 x199
(630) 654-3556 (fax)
(630) 915-4122 (cell)

I think FOLLOWME is going to fix this for you


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Re: [asterisk-users] call forwarding callerID

[Steve Edwards]

On Wed, 13 Oct 2010, Gerard wrote:

 This is not necessarily an asterisk issue, but a lot of you guys know 
 way more then me, so I have a question: someone at my company sets his 
 phone to forward calls to his cellphone, so someone calls our office, 
 call is forwarded to his cell, and the callerID that shows up on his 
 cell is of course our office number, because asterisk originates a new 
 call to his cell and then bridges the two. so he told me, a partner of 
 his, at his office does the same thing, and when he does it, the 
 callerID shows up as coming from the initial caller, not from his 
 office.

 so here's the schematic: customer - our office ---callforward-- 
 cellphone

 so should I call ATT and ask them to unlock our callerID so I can set 
 the outgoing callerID to the customer's number in my dialplan? or is 
 there some other way to handle this?

It depends on the technology and the carrier.

A simple POTS line and you're out of luck.

If you have a T1 (i.e. ISDN-PRI) and a co-operative carrier, it may just 
work or they may enable it if requested.

You could always use a co-operative SIP carrier (like Vitelity). A penny 
or 2 per minute will keep your someone happy.

-- 
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] Call Forwarding

[Stefan Schmidt]

Dan Journo schrieb:

  

 Since it isn't behaving like I want, is there any way to disable the 
 feature that allows a SIP phone to perform call forwarding?

  

 Thanks

 Dan

  

Hello,

in asterisk 1.6.x there is a Dial option i which suppress a 302 redirect 
which is very nice when dialing more than one phone at once, but you can 
use it also if you just dial one channel.

see output of core show application dial:

   i- Asterisk will ignore any forwarding requests it may receive on 
this
   dial attempt.


best regards

steve

-- 
Für weitere Fragen stehen wir gerne unter v...@sil.at oder
059944 - 2440 zur Verfügung.

Mit freundlichen Grüssen
-- 
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-
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A-1160 Wien // Fax 059944-9000 // www.sil.at  //
- 


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Re: [asterisk-users] Call Forwarding

[Dan Journo]

 in asterisk 1.6.x there is a Dial option

Sorry, any solutions for Asterisk 1.4?

Thanks
Dan

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Re: [asterisk-users] Call Forwarding Lopp Prevention

[Doug Lytle]

Paradise Dove wrote:
 i have two extensions which have call forwarding enabled when they are
 busy to forward the caller to each other.

 11 ==on busy== 12
 12 ==on busy== 11


   

exten = 11,1,Set(GROUP()=Loop11_Detect)
exten = 11,n,NoOP(Loop Detect for Extension 11: 
${GROUP_COUNT(Loop11_Detect)})
exten = 11,n,GotoIf($[ ${GROUP_COUNT(Loop11_Detect)}  2 ]?11,100)
exten = 11,n,Dial(SIP/12)

exten = 11,100,Voicemail([EMAIL PROTECTED]|b)
exten = 11,101,Hangup(17)


Doug

-- 
Ben Franklin quote:

Those who would give up Essential Liberty to purchase a little Temporary 
Safety, deserve neither Liberty nor Safety.


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Re: [asterisk-users] Call forwarding-in india

[Grygoriy Dobrovolskyy]

well give us details

2008/3/10, sandeep [EMAIL PROTECTED]:

  Hi All,
 Can any body tell how to enable call forward facility in INDAI
 for an asterisk IPPBX.

 Regards,
 Sandeep.S

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Re: [asterisk-users] Call forwarding-in india

[Godwin Stewart]

On Mon, 10 Mar 2008 16:22:45 +0530, sandeep [EMAIL PROTECTED]
wrote:

 Can any body tell how to enable call forward facility in INDAI
 for an asterisk IPPBX.

Why would it be different in India from anywhere else?

-- 
Godwin Stewart - Horwich IT services

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Re: [asterisk-users] Call forwarding

[Stefan Wintermeyer]

Am 15.02.2007 um 14:06 schrieb Dominik Zalewski:

exten = _*21*X.,2,Set(DB(CFIM/${CALLERID(NUM)})=${EXTEN:4})


Just use ${CALLERID(num)} and not ${CALLERID(NUM)}.

  Stefan

--
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Re: [asterisk-users] Call forwarding

[Dominik Zalewski]

On Thursday 15 February 2007 03:22:58 pm Stefan Wintermeyer wrote:
 Am 15.02.2007 um 14:06 schrieb Dominik Zalewski:
  exten = _*21*X.,2,Set(DB(CFIM/${CALLERID(NUM)})=${EXTEN:4})

 Just use ${CALLERID(num)} and not ${CALLERID(NUM)}.

Stefan


it didnt help :(  Is there is other way to implement call forwarding?
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Re: [asterisk-users] Call forwarding

[Steve Davies]

On 2/15/07, Dominik Zalewski [EMAIL PROTECTED] wrote:

Hi All,

I'm using asterisk 1.2.15 and call forwarding doesnt work for me.

from my extensions.conf:

; Unconditional Call Forward
exten = _*21*X.,1,NoCDR
exten = _*21*X.,2,Set(DB(CFIM/${CALLERID(NUM)})=${EXTEN:4})
exten = _*21*X.,3,Playback(vm-saved)
exten = _*21*X.,4,Hangup

exten = #21#,1,NoCDR
exten = #21#,2,DBdel(CFIM/${CALLERID(NUM)})
exten = #21#,3,Playback(auth-thankyou)
exten = #21#,4,Hangup


debug from asterisk CLI:

-- Executing NoCDR(SIP/dzalewski-081afaf0, ) in new stack
Feb 15 15:00:19 NOTICE[32307]: cdr.c:443 ast_cdr_free: CDR on
channel 'SIP/dzalewski-081afaf0' not posted
Feb 15 15:00:19 NOTICE[32307]: cdr.c:445 ast_cdr_free: CDR on
channel 'SIP/dzalewski-081afaf0' lacks end
-- Executing Set(SIP/dzalewski-081afaf0, DB(CFIM/200)=204) in new
stack
-- Executing Playback(SIP/dzalewski-081afaf0, vm-saved) in new stack
-- Playing 'vm-saved' (language 'en')
-- Executing Hangup(SIP/dzalewski-081afaf0, ) in new stack
  == Spawn extension (from-internal, *21*204, 4) exited non-zero
on 'SIP/dzalewski-081afaf0'



Above you are setting and clearing some database entries. What in your
dialplan are you using to act upon these values? You need something
resembling Example 1 on this page:
http://www.voip-info.org/wiki/view/Asterisk+call+forwarding Which
takes your saved values and acts on them.

Or perhaps I am misunderstanding something?

Steve
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Re: [asterisk-users] Call forwarding

[Pavel Jezek]

you just post only call forward activation part of dialplan,
but you must also make dialplan part, that reflect, how is set this 
callforward mark,

ie. if callforward is set, dial that number, if not, dial peer...



Dominik Zalewski wrote:

Hi All,

I'm using asterisk 1.2.15 and call forwarding doesnt work for me. 


from my extensions.conf:

; Unconditional Call Forward 
exten = _*21*X.,1,NoCDR 
exten = _*21*X.,2,Set(DB(CFIM/${CALLERID(NUM)})=${EXTEN:4}) 
exten = _*21*X.,3,Playback(vm-saved) 
exten = _*21*X.,4,Hangup 

exten = #21#,1,NoCDR 
exten = #21#,2,DBdel(CFIM/${CALLERID(NUM)}) 
exten = #21#,3,Playback(auth-thankyou) 
exten = #21#,4,Hangup



debug from asterisk CLI:

-- Executing NoCDR(SIP/dzalewski-081afaf0, ) in new stack
Feb 15 15:00:19 NOTICE[32307]: cdr.c:443 ast_cdr_free: CDR on 
channel 'SIP/dzalewski-081afaf0' not posted
Feb 15 15:00:19 NOTICE[32307]: cdr.c:445 ast_cdr_free: CDR on 
channel 'SIP/dzalewski-081afaf0' lacks end
-- Executing Set(SIP/dzalewski-081afaf0, DB(CFIM/200)=204) in new 
stack

-- Executing Playback(SIP/dzalewski-081afaf0, vm-saved) in new stack
-- Playing 'vm-saved' (language 'en')
-- Executing Hangup(SIP/dzalewski-081afaf0, ) in new stack
  == Spawn extension (from-internal, *21*204, 4) exited non-zero 
on 'SIP/dzalewski-081afaf0'


Thank you in advance,

Dominik


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Re: [asterisk-users] Call forwarding

[Dominik Zalewski]

On Thursday 15 February 2007 04:00:52 pm Pavel Jezek wrote:
 you just post only call forward activation part of dialplan,
 but you must also make dialplan part, that reflect, how is set this
 callforward mark,
 ie. if callforward is set, dial that number, if not, dial peer...

Do you have any example of this diaplan part?

Thanks,

Dominik
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Re: [asterisk-users] Call forwarding

[Paul Hales]

With the call forward button on the phone? ;)

PaulH


 Stefan
 
 
 it didnt help :(  Is there is other way to implement call forwarding?
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Re: [asterisk-users] Call Forwarding Using Asterisk

[jk]

Thank you Ram,
Can you give me some example, how can I  do that.

-Jk

ram wrote:

Hi
 
its possible

you need mention in the config
 
Ram


 
On 10/17/06, *jk* [EMAIL PROTECTED] 
mailto:[EMAIL PROTECTED] wrote:


Can I do this with Asterisk,

Call comes to Asterisk Server (Master), Then master just forwards
calls
to other slave asterisk servers one by one.
Like this
Master forward 1st call to Slave 1,
Second call to Slave 2,
Third call to slave 1
Fourth call to slave 2.


Is it possible? I will appreciate if some one help me with this.

Thank you,
-Jai
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Re: [asterisk-users] Call Forwarding not working for extension in queue, why?

[Dovid B]

What do you mean by that it is forwarded. Is it set 
on the phone or do you have it set in que memeber.

  - Original Message - 
  From: 
  Zeeshan 
  Zakaria 
  To: Asterisk Users Mailing List - 
  Non-Commercial Discussion 
  Sent: Wednesday, October 04, 2006 8:56 
  AM
  Subject: [asterisk-users] Call Forwarding 
  not working for extension in queue, why?
  Extension 200 is member of a queue. At night time, it is 
  forwarded to a different number. Now when this extension is dialed directly, 
  call forwarding works, but when a call comes into the queue, ext. 200 keeps on 
  ringing and doesn't get forwarded. Why is that and how to fix it? -- Zeeshan A Zakaria 
  
  

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Re: [asterisk-users] Call Forwarding not working for extension in queue, why?

[Zeeshan Zakaria]

I pick up extension 200, dial *72 and forward it to another number. When a call comes in to the queue, it dials extension 200 along with the other extensions. I expect queue not to dial extension 200 but to dial the forwarded number which it doesn't do and keep ringing extension 200, and there is nobody to pick it up. Why it doesn't dial the forwarded number?
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Re: [asterisk-users] Call Forwarding in SIP.conf

[broadbandvoice]

Thanks all. It works fine now.

-- Original message -- From: "Tim St. Pierre" [EMAIL PROTECTED]  ___  --Bandwidth and Colocation provided by Easynews.com --   asterisk-users mailing list  To UNSUBSCRIBE or update options visit:  http://lists.digium.com/mailman/listinfo/asterisk-users 

---BeginMessage---
Check your Dial() string to make sure that you haven't mistyped and put 
gafachi-o instead of gafachi-out.  Specifiying the full host name will also 
work.

As a hint, you can refresh these changes with out restarting your server (and 
therefore without disrupting any calls in progress)

extensions reload will refresh the extensions file
reload will reload all your configs
sip reload will reload only sip configs (and re-register everything)

Very handy when working on an active machine.


On September 8, 2006 14:19, [EMAIL PROTECTED] wrote:
 Tim, this is the way I have Gafachi set up in sip.conf and works well with
 channels that have an ATA attached to it but not the virtual one. I have
 changed the host in extensions.conf to the .sip.gafachi.com.
 But I have calls on the server and cannot restart it yet. I'll keep you
 posted and thanks for the feedback.

 [gafachi-out]
 type=peer
 secret=xx
 username=x
 fromuser=x
 fromdomain=xxx
 host=.sip.gafachi.com
 ;usereqphone=yes; This provider requires ;user=phone on
 URI ;nat=yes
 rtptimeout=60
 dtmfmode=rfc2833


 -- Original message --
 From: Tim St. Pierre [EMAIL PROTECTED]

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-- 
Tim St. Pierre

IP telephony specialist
sip://[EMAIL PROTECTED]
Toronto: 647 722 6930
Toll-Free 1 888 488 6940
[EMAIL PROTECTED]


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Re: [asterisk-users] Call Forwarding in SIP.conf

[broadbandvoice]

I have a follow up question. How do I pass on the caller ID of the call I'm forwarding to the other party? I can pass on the channels caller ID but prefer to pass on the forwarding party's number instead.

-- Original message -- From: [EMAIL PROTECTED] 
Thanks all. It works fine now.

-- Original message -- From: "Tim St. Pierre" [EMAIL PROTECTED]  ___  --Bandwidth and Colocation provided by Easynews.com --   asterisk-users mailing list  To UNSUBSCRIBE or update options visit:  http://lists.digium.com/mailman/listinfo/asterisk-users 

---BeginMessage---
---BeginMessage---
Check your Dial() string to make sure that you haven't mistyped and put 
gafachi-o instead of gafachi-out.  Specifiying the full host name will also 
work.

As a hint, you can refresh these changes with out restarting your server (and 
therefore without disrupting any calls in progress)

extensions reload will refresh the extensions file
reload will reload all your configs
sip reload will reload only sip configs (and re-register everything)

Very handy when working on an active machine.


On September 8, 2006 14:19, [EMAIL PROTECTED] wrote:
 Tim, this is the way I have Gafachi set up in sip.conf and works well with
 channels that have an ATA attached to it but not the virtual one. I have
 changed the host in extensions.conf to the .sip.gafachi.com.
 But I have calls on the server and cannot restart it yet. I'll keep you
 posted and thanks for the feedback.

 [gafachi-out]
 type=peer
 secret=xx
 username=x
 fromuser=x
 fromdomain=xxx
 host=.sip.gafachi.com
 ;usereqphone=yes; This provider requires ;user=phone on
 URI ;nat=yes
 rtptimeout=60
 dtmfmode=rfc2833


 -- Original message --
 From: Tim St. Pierre [EMAIL PROTECTED]

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-- 
Tim St. Pierre

IP telephony specialist
sip://[EMAIL PROTECTED]
Toronto: 647 722 6930
Toll-Free 1 888 488 6940
[EMAIL PROTECTED]


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Re: [asterisk-users] Call Forwarding in SIP.conf

[Tim St. Pierre]

If you don't set the callerID in the channel, it will get passed on as-is.  
Don't change it, and it will stay the same.

-TIm

On September 9, 2006 12:27, [EMAIL PROTECTED] wrote:
 I have a follow up question. How do I pass on the caller ID of the call I'm
 forwarding to the other party? I can pass on the channels caller ID but
 prefer to pass on the forwarding party's number instead.

 -- Original message --
 From: [EMAIL PROTECTED]

 Thanks all. It works fine now.

 -- Original message --
 From: Tim St. Pierre [EMAIL PROTECTED]

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-- 
Tim St. Pierre

IP telephony specialist
sip://[EMAIL PROTECTED]
Toronto: 647 722 6930
Toll-Free 1 888 488 6940
[EMAIL PROTECTED]


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Re: [asterisk-users] Call Forwarding in SIP.conf

[broadbandvoice]

I tried both of them but it still goes asID unavailable. First I commented it out, that did not work and left it blank and that did not work either. Below is the sample in sip.conf

[4305]type=frienduser=4305secret=xxx;context=from-sipcallerid= ; left it blank but did not get passed on!host=dynamicnat=yesqualify=yescanreinvite=nodtmfmode=rfc2833

-- Original message -- From: "Tim St. Pierre" [EMAIL PROTECTED]  ___  --Bandwidth and Colocation provided by Easynews.com --   asterisk-users mailing list  To UNSUBSCRIBE or update options visit:  http://lists.digium.com/mailman/listinfo/asterisk-users 

---BeginMessage---
If you don't set the callerID in the channel, it will get passed on as-is.  
Don't change it, and it will stay the same.

-TIm

On September 9, 2006 12:27, [EMAIL PROTECTED] wrote:
 I have a follow up question. How do I pass on the caller ID of the call I'm
 forwarding to the other party? I can pass on the channels caller ID but
 prefer to pass on the forwarding party's number instead.

 -- Original message --
 From: [EMAIL PROTECTED]

 Thanks all. It works fine now.

 -- Original message --
 From: Tim St. Pierre [EMAIL PROTECTED]

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-- 
Tim St. Pierre

IP telephony specialist
sip://[EMAIL PROTECTED]
Toronto: 647 722 6930
Toll-Free 1 888 488 6940
[EMAIL PROTECTED]


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Re: [asterisk-users] Call Forwarding in SIP.conf

[William Piper]

whatever the did is needs to be put in the extensions.conf  told to dial your cellphone. 
Example:

exten = _011123445566,1,Dial,SIP/[EMAIL PROTECTED] 

assuming that your using a SIP carrier, replace 1234567890 with your cellphone  1.2.3.4 with the carrier's IP or carriers context name in sip.conf.
bp
On 9/8/06, [EMAIL PROTECTED] [EMAIL PROTECTED]
 wrote:



I'm using it for virtual numbers. I have international virtual number from a DID provider and want to forward it to my cell phone.

In Sip.conf I have the channel


[4305]
type=friend
user=4305
secret=
;context=from-sip
callerid=
host=dynamic
nat=yes
canreinvite=no
dtmfmode=rfc2833
;incominglimit=1
;[EMAIL PROTECTED]
;disallow=all
;allow=ulaw
;allow=alaw
;allow=g723.1 ; Asterisk only supports g723.1 pass-thru!

and in extensions.conf I have

exten = 4305,1,Dial(SIP/4305,120,rt) ; permit transfer

This had worked in the past when I forwarded it through the Linksys ATA but now have run out of ATA's.


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Re: [asterisk-users] Call Forwarding in SIP.conf

[broadbandvoice]

It sounds like a good idea, I tried it and get this error


Sep 8 09:52:17 WARNING[27193]: chan_sip.c:1968 create_addr: No such host: gafachi-o
Sep 8 09:52:17 NOTICE[27193]: app_dial.c:1011 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)

In Extensions.conf I have
exten = 4305,1,Dial(SIP/[EMAIL PROTECTED]) ; permit transfer

In Sip.conf I have
[4305]
type=friend
user=4305
secret=
;context=from-sip
callerid=
host=dynamic
nat=yes
canreinvite=no
dtmfmode=rfc2833
;incominglimit=1
;[EMAIL PROTECTED]
;disallow=all
;allow=ulaw
;allow=alaw
;allow=g723.1 ; Asterisk only supports g723.1 pass-thru!



-- Original message -- From: "William Piper" [EMAIL PROTECTED] 
whatever the did is needs to be put in the extensions.conf  told to dial your cellphone. 
Example:

exten = _011123445566,1,Dial,SIP/[EMAIL PROTECTED] 

assuming that your using a SIP carrier, replace 1234567890 with your cellphone  1.2.3.4 with the carrier's IP or carriers context name in sip.conf.
bp
On 9/8/06, [EMAIL PROTECTED] [EMAIL PROTECTED]  wrote: 



I'm using it for virtual numbers. I have international virtual number from a DID provider and want to forward it to my cell phone.

In Sip.conf I have the channel


[4305]
type=friend
user=4305
secret=
;context=from-sip
callerid=
host=dynamic
nat=yes
canreinvite=no
dtmfmode=rfc2833
;incominglimit=1
;[EMAIL PROTECTED]
;disallow=all
;allow=ulaw
;allow=alaw
;allow=g723.1 ; Asterisk only supports g723.1 pass-thru!

and in extensions.conf I have

exten = 4305,1,Dial(SIP/4305,120,rt) ; permit transfer

This had worked in the past when I forwarded it through the Linksys ATA but now have run out of ATA's.


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Re: [asterisk-users] Call Forwarding in SIP.conf

[Tim St. Pierre]

Do you have gafachi-o in your sip.conf?

Since it's not a valid host name, you need to have an entry in sip.conf to 
tell asterisk how to make a call to gafachi-o.

That's why it is telling you No such host.



On September 8, 2006 12:57, [EMAIL PROTECTED] wrote:
 It sounds like a good idea, I tried it and get this error

 Sep  8 09:52:17 WARNING[27193]: chan_sip.c:1968 create_addr: No such host:
 gafachi-o Sep  8 09:52:17 NOTICE[27193]: app_dial.c:1011 dial_exec_full:
 Unable to create channel of type 'SIP' (cause 3 - No route to destination)

 In Extensions.conf I have
 exten = 4305,1,Dial(SIP/[EMAIL PROTECTED])  ; permit transfer

 In Sip.conf I have
 [4305]
 type=friend
 user=4305
 secret=
 ;context=from-sip
 callerid=
 host=dynamic
 nat=yes
 canreinvite=no
 dtmfmode=rfc2833
 ;incominglimit=1
 ;[EMAIL PROTECTED]
 ;disallow=all
 ;allow=ulaw
 ;allow=alaw
 ;allow=g723.1   ; Asterisk only supports g723.1 pass-thru!



 -- Original message --
 From: William Piper [EMAIL PROTECTED]

 whatever the did is needs to be put in the extensions.conf  told to dial
 your cellphone. Example:

 exten = _011123445566,1,Dial,SIP/[EMAIL PROTECTED]

 assuming that your using a SIP carrier, replace 1234567890 with your
 cellphone  1.2.3.4 with the carrier's IP or carriers context name in
 sip.conf.

 bp

 On 9/8/06, [EMAIL PROTECTED] [EMAIL PROTECTED]  wrote:
 I'm using it for virtual numbers. I have international virtual number from
 a DID provider and want to forward it to my cell phone.

 In Sip.conf I have the channel

 [4305]
 type=friend
 user=4305
 secret=
 ;context=from-sip
 callerid=
 host=dynamic
 nat=yes
 canreinvite=no
 dtmfmode=rfc2833
 ;incominglimit=1
 ;[EMAIL PROTECTED]
 ;disallow=all
 ;allow=ulaw
 ;allow=alaw
 ;allow=g723.1   ; Asterisk only supports g723.1 pass-thru!

 and in extensions.conf I have

 exten = 4305,1,Dial(SIP/4305,120,rt)  ; permit transfer

 This had worked in the past when I forwarded it through the Linksys ATA but
 now have run out of ATA's.


 -- Original message --
 From: Tim St. Pierre  [EMAIL PROTECTED]

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-- 
Tim St. Pierre

IP telephony specialist
sip://[EMAIL PROTECTED]
Toronto: 647 722 6930
Toll-Free 1 888 488 6940
[EMAIL PROTECTED]


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Re: [asterisk-users] Call Forwarding in SIP.conf

[broadbandvoice]

Tim, this is the way I have Gafachi set up in sip.conf and works well with channels that have anATA attached to it but not the virtual one. I have changed the host in extensions.conf to the .sip.gafachi.com. But I have calls on the server and cannot restart it yet. I'll keep you posted and thanks for the feedback.

[gafachi-out]type=peersecret=xxusername=xfromuser=xfromdomain=xxxhost=.sip.gafachi.com;usereqphone=yes ; This provider requires ";user=phone" on URI;nat=yesrtptimeout=60dtmfmode=rfc2833

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---BeginMessage---
Do you have gafachi-o in your sip.conf?

Since it's not a valid host name, you need to have an entry in sip.conf to 
tell asterisk how to make a call to gafachi-o.

That's why it is telling you No such host.



On September 8, 2006 12:57, [EMAIL PROTECTED] wrote:
 It sounds like a good idea, I tried it and get this error

 Sep  8 09:52:17 WARNING[27193]: chan_sip.c:1968 create_addr: No such host:
 gafachi-o Sep  8 09:52:17 NOTICE[27193]: app_dial.c:1011 dial_exec_full:
 Unable to create channel of type 'SIP' (cause 3 - No route to destination)

 In Extensions.conf I have
 exten = 4305,1,Dial(SIP/[EMAIL PROTECTED])  ; permit transfer

 In Sip.conf I have
 [4305]
 type=friend
 user=4305
 secret=
 ;context=from-sip
 callerid=
 host=dynamic
 nat=yes
 canreinvite=no
 dtmfmode=rfc2833
 ;incominglimit=1
 ;[EMAIL PROTECTED]
 ;disallow=all
 ;allow=ulaw
 ;allow=alaw
 ;allow=g723.1   ; Asterisk only supports g723.1 pass-thru!



 -- Original message --
 From: William Piper [EMAIL PROTECTED]

 whatever the did is needs to be put in the extensions.conf  told to dial
 your cellphone. Example:

 exten = _011123445566,1,Dial,SIP/[EMAIL PROTECTED]

 assuming that your using a SIP carrier, replace 1234567890 with your
 cellphone  1.2.3.4 with the carrier's IP or carriers context name in
 sip.conf.

 bp

 On 9/8/06, [EMAIL PROTECTED] [EMAIL PROTECTED]  wrote:
 I'm using it for virtual numbers. I have international virtual number from
 a DID provider and want to forward it to my cell phone.

 In Sip.conf I have the channel

 [4305]
 type=friend
 user=4305
 secret=
 ;context=from-sip
 callerid=
 host=dynamic
 nat=yes
 canreinvite=no
 dtmfmode=rfc2833
 ;incominglimit=1
 ;[EMAIL PROTECTED]
 ;disallow=all
 ;allow=ulaw
 ;allow=alaw
 ;allow=g723.1   ; Asterisk only supports g723.1 pass-thru!

 and in extensions.conf I have

 exten = 4305,1,Dial(SIP/4305,120,rt)  ; permit transfer

 This had worked in the past when I forwarded it through the Linksys ATA but
 now have run out of ATA's.


 -- Original message --
 From: Tim St. Pierre  [EMAIL PROTECTED]

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-- 
Tim St. Pierre

IP telephony specialist
sip://[EMAIL PROTECTED]
Toronto: 647 722 6930
Toll-Free 1 888 488 6940
[EMAIL PROTECTED]


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Re: [asterisk-users] Call Forwarding in SIP.conf

[Tim St. Pierre]

Check your Dial() string to make sure that you haven't mistyped and put 
gafachi-o instead of gafachi-out.  Specifiying the full host name will also 
work.

As a hint, you can refresh these changes with out restarting your server (and 
therefore without disrupting any calls in progress)

extensions reload will refresh the extensions file
reload will reload all your configs
sip reload will reload only sip configs (and re-register everything)

Very handy when working on an active machine.


On September 8, 2006 14:19, [EMAIL PROTECTED] wrote:
 Tim, this is the way I have Gafachi set up in sip.conf and works well with
 channels that have an ATA attached to it but not the virtual one. I have
 changed the host in extensions.conf to the .sip.gafachi.com.
 But I have calls on the server and cannot restart it yet. I'll keep you
 posted and thanks for the feedback.

 [gafachi-out]
 type=peer
 secret=xx
 username=x
 fromuser=x
 fromdomain=xxx
 host=.sip.gafachi.com
 ;usereqphone=yes; This provider requires ;user=phone on
 URI ;nat=yes
 rtptimeout=60
 dtmfmode=rfc2833


 -- Original message --
 From: Tim St. Pierre [EMAIL PROTECTED]

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-- 
Tim St. Pierre

IP telephony specialist
sip://[EMAIL PROTECTED]
Toronto: 647 722 6930
Toll-Free 1 888 488 6940
[EMAIL PROTECTED]


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Re: [asterisk-users] Call Forwarding in SIP.conf

[Tim St. Pierre]

Call forwarding doesn't go in sip.conf, it has to go in the dialplan.

You set up your outbound provider in sip.conf

In your dialplan, you use the Dial application like this:
exten = _NXXNXX,1,Dial(SIP/outgoingprovider/${EXTEN})

This will dial out to a PSTN number based on the extension passed to it.

What is it you want to do?  Call forwarding on not-registered or no answer?

That needs a database and a macro.  What is your goal?

-Tim

On September 7, 2006 17:14, [EMAIL PROTECTED] wrote:
 I looked through the forums but could not find exactly what I needed. I
 need help setting up call forwarding in sip.conf, where the call forwards
 to PSTN number without a sip phone but just the channels in sip.conf
 without any hardware or softphone. Any help will be greatly appreciated.

-- 
Tim St. Pierre

IP telephony specialist
sip://[EMAIL PROTECTED]
Toronto: 647 722 6930
Toll-Free 1 888 488 6940
[EMAIL PROTECTED]


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Re: [asterisk-users] Call Forwarding in SIP.conf

[broadbandvoice]

I'm using it for virtual numbers. I have international virtual number from a DID provider and want to forward it to my cell phone.

In Sip.conf I have the channel


[4305]
type=friend
user=4305
secret=
;context=from-sip
callerid=
host=dynamic
nat=yes
canreinvite=no
dtmfmode=rfc2833
;incominglimit=1
;[EMAIL PROTECTED]
;disallow=all
;allow=ulaw
;allow=alaw
;allow=g723.1 ; Asterisk only supports g723.1 pass-thru!

and in extensions.conf I have

exten = 4305,1,Dial(SIP/4305,120,rt) ; permit transfer

This had worked in the past when I forwarded it through the Linksys ATA but now have run out of ATA's.


-- Original message -- From: "Tim St. Pierre" [EMAIL PROTECTED]  ___  --Bandwidth and Colocation provided by Easynews.com --   asterisk-users mailing list  To UNSUBSCRIBE or update options visit:  http://lists.digium.com/mailman/listinfo/asterisk-users 

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RE: [asterisk-users] call forwarding to mobile phone

[Sam Tam]

You will need an asterisk server + X100P +
GSM Gateway say from cyber-telecom.net


You can config the X100P with GSM Gateway like what you would do with an normal
Phone line and use it to dial in or out between VoIP and GSM Network











From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rodrigo Mercado
Sent: Tuesday, July 18, 2006 5:34
PM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re: [asterisk-users] call
forwarding to mobile phone







I need information / documents or configurations of asterisk with other
Telephonic head offices(plants), for your help , thank











sorry for my english, i speek spanish only.

















atte,
Rodrigo M







On 7/18/06, Lito
Lampitoc [EMAIL PROTECTED]
wrote: 



is there a way I can do
call forwarding to mobile phone without using a gsm gateway? my landline is
capable of calling a gsm network.



On 7/18/06, Sam Tam
 [EMAIL PROTECTED] wrote:








Get an GSM Gateway from cyber-telecom.net











From: [EMAIL PROTECTED]
[mailto:
[EMAIL PROTECTED] ] On
Behalf Of Lito Lampitoc
Sent: Tuesday, July 18, 2006 4:57
PM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: [asterisk-users] call
forwarding to mobile phone









Hello
all,

Is it possible to forward a call received by the asterisk server to a mobile
phone? 
If yes, how? a link or reference is highly appreciated. 

thanks

Lito








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RE: [asterisk-users] call forwarding to mobile phone

[Sam Tam]

Yes Get an X100P

Sam











From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lito Lampitoc
Sent: Tuesday, July 18, 2006 5:16
PM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re: [asterisk-users] call
forwarding to mobile phone





is there a way I can do
call forwarding to mobile phone without using a gsm gateway? my landline is capable
of calling a gsm network.



On 7/18/06, Sam Tam
[EMAIL PROTECTED] wrote:







Get an GSM Gateway from cyber-telecom.net











From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
] On Behalf Of Lito Lampitoc
Sent: Tuesday, July 18, 2006 4:57
PM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: [asterisk-users] call
forwarding to mobile phone









Hello
all,

Is it possible to forward a call received by the asterisk server to a mobile
phone? 
If yes, how? a link or reference is highly appreciated.

thanks

Lito










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RE: [asterisk-users] call forwarding to mobile phone

[Dave Cotton]

On Wed, 2006-07-19 at 19:04 +0800, Sam Tam wrote:
 You will need an asterisk server + X100P + GSM Gateway say from
 cyber-telecom.net

Not forgetting that the above person IS cyber-telecom.net.

Therefore his advice is not impartial.


-- 
Dave Cotton [EMAIL PROTECTED]

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RE: [asterisk-users] call forwarding to mobile phone

[Steven]

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sam Tam
Sent: 19 July 2006 12:04
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] call forwarding to mobile phone

You will need an asterisk server + X100P + GSM Gateway say from
cyber-telecom.net

You can config the X100P with GSM Gateway like what you would do with an
normal Phone line and use it to dial in or out between VoIP and GSM Network


From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rodrigo
Mercado
Sent: Tuesday, July 18, 2006 5:34 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] call forwarding to mobile phone

I need information / documents or configurations of asterisk with other
Telephonic head offices(plants), for your help , thank
 
sorry for my english, i speek spanish only.
 
 
atte,
Rodrigo M

 
On 7/18/06, Lito Lampitoc [EMAIL PROTECTED] wrote: 
is there a way I can do call forwarding to mobile phone without using a gsm
gateway? my landline is capable of calling a gsm network.
On 7/18/06, Sam Tam  [EMAIL PROTECTED] wrote: 
Get an GSM Gateway from cyber-telecom.net
 

From: [EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED] ] On Behalf Of Lito Lampitoc
Sent: Tuesday, July 18, 2006 4:57 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] call forwarding to mobile phone
 
Hello all,

Is it possible to forward a call received by the asterisk server to a mobile
phone? 
If yes, how? a link or reference is highly appreciated. 

thanks

Lito

Hi Sam,

Uhm... Wah? Your saying to call a mobile number you need a gsm gateway? What
have you been smoking and where can I get some?

Last I heard you can use a standard telephone line.. One of us must be on
cloud nine!

Steve Daniels

-- 
No virus found in this outgoing message.
Checked by AVG Free Edition.
Version: 7.1.394 / Virus Database: 268.10.1/391 - Release Date: 18/07/2006
 

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RE: [asterisk-users] call forwarding to mobile phone

[Sam Tam]

Get an GSM Gateway from cyber-telecom.net











From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lito Lampitoc
Sent: Tuesday, July 18, 2006 4:57
PM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: [asterisk-users] call
forwarding to mobile phone





Hello all,

Is it possible to forward a call received by the asterisk server to a mobile
phone? 
If yes, how? a link or reference is highly appreciated.

thanks

Lito






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Re: [asterisk-users] call forwarding to mobile phone

[Lito Lampitoc]

is there a way I can do call forwarding to mobile phone without using a gsm gateway? my landline is capable of calling a gsm network.On 7/18/06, Sam Tam
 [EMAIL PROTECTED] wrote:













Get an GSM Gateway from 
cyber-telecom.net











From:
[EMAIL PROTECTED]

[mailto:[EMAIL PROTECTED]
] On Behalf Of Lito Lampitoc
Sent: Tuesday, July 18, 2006 4:57
PM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: [asterisk-users] call
forwarding to mobile phone





Hello all,

Is it possible to forward a call received by the asterisk server to a mobile
phone? 
If yes, how? a link or reference is highly appreciated.

thanks

Lito







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Re: [asterisk-users] call forwarding to mobile phone

[Rodrigo Mercado]

I need information / documents or configurations of asterisk with other Telephonic head offices(plants), for your help , thank

sorry for my english, i speek spanish only.


atte,Rodrigo M
On 7/18/06, Lito Lampitoc [EMAIL PROTECTED] wrote:

is there a way I can do call forwarding to mobile phone without using a gsm gateway? my landline is capable of calling a gsm network.
On 7/18/06, Sam Tam 
[EMAIL PROTECTED] wrote: 




Get an GSM Gateway from 
cyber-telecom.net





From: 
[EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED] ] On Behalf Of Lito LampitocSent: Tuesday, July 18, 2006 4:57 PM
To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [asterisk-users] call forwarding to mobile phone


Hello all,Is it possible to forward a call received by the asterisk server to a mobile phone? If yes, how? a link or reference is highly appreciated.
thanksLito
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Re: [asterisk-users] call forwarding to mobile phone

[Woodoo People .pGa!]

 is there a way I can do call forwarding to mobile phone without using a gsm
 gateway? my landline is capable of calling a gsm network.

[from-gsm]
exten = s,1,Dial(Zap/$your_mobile)

that's all

-- 
WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com
[EMAIL PROTECTED]@RedHat.users
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Re: [asterisk-users] call forwarding to mobile phone

[Lito Lampitoc]

thanks a lot!On 7/19/06, Woodoo People .pGa! [EMAIL PROTECTED] wrote:
 is there a way I can do call forwarding to mobile phone without using a gsm gateway? my landline is capable of calling a gsm network.[from-gsm]exten = s,1,Dial(Zap/$your_mobile)that's all
--WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com[EMAIL PROTECTED]]iCQ#33118021[wpeople.on.iRCNet
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Re: [Asterisk-Users] Call forwarding for particular extension

[Giovanni Miano]

exten = 2006,2,goto(s-${DIALSTATUS},1)exten = s-BUSY,1,DIAL(SIP/sipura3)exten = s-NOANSWER,1,exten = s-www.voip-info.org/wiki-Asterisk+variable+DIALSTATUS
Cheers,Giovanni Miano2006/1/6, nr k [EMAIL PROTECTED]
:Hi allI need to configure call forwarding for particularextension is 
busy.how to configure this my extension configuration is like following.exten = 2006,1,Dial(SIP/sipura2)regardsramakrishnan.n__
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-- Giovanni Miano
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Re: [Asterisk-Users] Call forwarding for particular extension

[nr k]

Hi All

Thanks for ur reply.My phone having 2 line with same
extension and also I configure voicemail if the user
not pickup the phone within 25 seconds for tht
extension but i want if my line 1 is busy then forward
the call to some other extension .my config is like
following.my phone having the adhoc conference
facility so tht I need 2 lines I am using SIPURA IP
phones.pls do the needful...

exten = 2007,1,Dial(SIP/sipura3,25,r)
exten = 2007,2,VoiceMail([EMAIL PROTECTED])

regards
ramakrishnan.n

--- Giovanni Miano [EMAIL PROTECTED] wrote:

 exten = 2006,2,goto(s-${DIALSTATUS},1)
 exten = s-BUSY,1,DIAL(SIP/sipura3)
 exten = s-NOANSWER,1,
 exten = s-
 

www.*voip-info*.org/wiki-Asterisk+variable+DIALSTATUS
 
 Cheers,
 Giovanni Miano
 
 2006/1/6, nr k [EMAIL PROTECTED]:
 
  Hi all
 
  I need to configure call forwarding for particular
  extension is busy.how to configure this my
 extension
  configuration is like following.
 
 
  exten = 2006,1,Dial(SIP/sipura2)
 
 
  regards
  ramakrishnan.n
 
 
 
  __
  Yahoo! DSL – Something to write home about.
  Just $16.99/mo. or less.
  dsl.yahoo.com
 
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Re: [Asterisk-Users] Call Forwarding with Account Code.. can it be done?

[C F]

Create a context for that ATA that always applies the account code in
the DP before it you issue the dial command.

On 12/6/05, Matt [EMAIL PROTECTED] wrote:
 I want to allow my users to be able to
 Call Forward Unconditional
 Call Forward Busy
 Call Forward No Answer

 And curently I am doing this via my ATA and phone settings, however
 this has the problem that when a call is forwarded it goes out without
 an accountcode (Even though the ATA is forwarding the call), and hence
 I can't track the call!

 Can someone suggest a way to either fix this so that accountcodes go
 into the CDRs when the ATA/phone forwards the call, or to do the three
 forwarding types directly on asterisk?
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Re: [Asterisk-Users] Call Forwarding with Account Code.. can it be done?

[Andy Kuo]

I use SetAccount(${EXTEN}) when the extension gets the call. The original dialed extension will be recorded as AccountCode in CDR, before the call is forwarded. The 1st field in CDR will be the extension your customer, the 2nd will be the caller (source), the 3rd will be the forwared number.


It works for me pretty well.

Andy
On 12/6/05, Matt [EMAIL PROTECTED] wrote:
I want to allow my users to be able toCall Forward UnconditionalCall Forward Busy
Call Forward No AnswerAnd curently I am doing this via my ATA and phone settings, howeverthis has the problem that when a call is forwarded it goes out withoutan accountcode (Even though the ATA is forwarding the call), and hence
I can't track the call!Can someone suggest a way to either fix this so that accountcodes gointo the CDRs when the ATA/phone forwards the call, or to do the threeforwarding types directly on asterisk?
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Re: [Asterisk-Users] Call Forwarding with Account Code.. can it be done?

[Matt]

Hrmm that works except that my accountcode is not the extension of the
customer/user, but is a distinct accountcode (ID).

Oooo... you are setting the accountcode when you GET the
call.  I guess I could do that... before I go to do too much work, is
there a way to get asterisk to know the accountcode for the inbound
call?

On 12/6/05, Andy Kuo [EMAIL PROTECTED] wrote:
 I use SetAccount(${EXTEN}) when the extension gets the call.  The original
 dialed extension will be recorded as AccountCode in CDR, before the call is
 forwarded.  The 1st field in CDR will be the extension your customer, the
 2nd will be the caller (source), the 3rd will be the forwared number.

 It works for me pretty well.

 Andy


 On 12/6/05, Matt [EMAIL PROTECTED] wrote:
 
  I want to allow my users to be able to
  Call Forward Unconditional
  Call Forward Busy
  Call Forward No Answer
 
  And curently I am doing this via my ATA and phone settings, however
  this has the problem that when a call is forwarded it goes out without
  an accountcode (Even though the ATA is forwarding the call), and hence
  I can't track the call!
 
  Can someone suggest a way to either fix this so that accountcodes go
  into the CDRs when the ATA/phone forwards the call, or to do the three
  forwarding types directly on asterisk?
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Re: [Asterisk-Users] Call Forwarding with Account Code.. can it be done?

[C F]

I'm not sure what you are trying to set it to, I'm assuming that some
of the stuff you want is available here:
http://www.voip-info.org/wiki-asterisk+variables
or in README.variables in /usr/src/asteriks (or one of the sub folders)
Look at RDNIS or DNID, either one might have the dialded number (which
is the extension).
In any case if you create an outgoing context just for that device,
then you shouldn't have a problem setting it to whatever you want, as
only that device will use it.

On 12/6/05, Matt [EMAIL PROTECTED] wrote:
 Hrmm that works except that my accountcode is not the extension of the
 customer/user, but is a distinct accountcode (ID).

 Oooo... you are setting the accountcode when you GET the
 call.  I guess I could do that... before I go to do too much work, is
 there a way to get asterisk to know the accountcode for the inbound
 call?

 On 12/6/05, Andy Kuo [EMAIL PROTECTED] wrote:
  I use SetAccount(${EXTEN}) when the extension gets the call.  The original
  dialed extension will be recorded as AccountCode in CDR, before the call is
  forwarded.  The 1st field in CDR will be the extension your customer, the
  2nd will be the caller (source), the 3rd will be the forwared number.
 
  It works for me pretty well.
 
  Andy
 
 
  On 12/6/05, Matt [EMAIL PROTECTED] wrote:
  
   I want to allow my users to be able to
   Call Forward Unconditional
   Call Forward Busy
   Call Forward No Answer
  
   And curently I am doing this via my ATA and phone settings, however
   this has the problem that when a call is forwarded it goes out without
   an accountcode (Even though the ATA is forwarding the call), and hence
   I can't track the call!
  
   Can someone suggest a way to either fix this so that accountcodes go
   into the CDRs when the ATA/phone forwards the call, or to do the three
   forwarding types directly on asterisk?
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Re: [Asterisk-Users] Call Forwarding with Account Code.. can it be done?

[Matt]

Right, but I can't create a context for every device :)

On 12/6/05, C F [EMAIL PROTECTED] wrote:
 I'm not sure what you are trying to set it to, I'm assuming that some
 of the stuff you want is available here:
 http://www.voip-info.org/wiki-asterisk+variables
 or in README.variables in /usr/src/asteriks (or one of the sub folders)
 Look at RDNIS or DNID, either one might have the dialded number (which
 is the extension).
 In any case if you create an outgoing context just for that device,
 then you shouldn't have a problem setting it to whatever you want, as
 only that device will use it.

 On 12/6/05, Matt [EMAIL PROTECTED] wrote:
  Hrmm that works except that my accountcode is not the extension of the
  customer/user, but is a distinct accountcode (ID).
 
  Oooo... you are setting the accountcode when you GET the
  call.  I guess I could do that... before I go to do too much work, is
  there a way to get asterisk to know the accountcode for the inbound
  call?
 
  On 12/6/05, Andy Kuo [EMAIL PROTECTED] wrote:
   I use SetAccount(${EXTEN}) when the extension gets the call.  The original
   dialed extension will be recorded as AccountCode in CDR, before the call 
   is
   forwarded.  The 1st field in CDR will be the extension your customer, the
   2nd will be the caller (source), the 3rd will be the forwared number.
  
   It works for me pretty well.
  
   Andy
  
  
   On 12/6/05, Matt [EMAIL PROTECTED] wrote:
   
I want to allow my users to be able to
Call Forward Unconditional
Call Forward Busy
Call Forward No Answer
   
And curently I am doing this via my ATA and phone settings, however
this has the problem that when a call is forwarded it goes out without
an accountcode (Even though the ATA is forwarding the call), and hence
I can't track the call!
   
Can someone suggest a way to either fix this so that accountcodes go
into the CDRs when the ATA/phone forwards the call, or to do the three
forwarding types directly on asterisk?
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Re: [Asterisk-Users] Call Forwarding with Account Code.. can it be done?

[C F]

Why cant you create a context for every device?
It can be just a one line context or a context that is based on a
template. I don't see why you cant.

On 12/6/05, Matt [EMAIL PROTECTED] wrote:
 Right, but I can't create a context for every device :)

 On 12/6/05, C F [EMAIL PROTECTED] wrote:
  I'm not sure what you are trying to set it to, I'm assuming that some
  of the stuff you want is available here:
  http://www.voip-info.org/wiki-asterisk+variables
  or in README.variables in /usr/src/asteriks (or one of the sub folders)
  Look at RDNIS or DNID, either one might have the dialded number (which
  is the extension).
  In any case if you create an outgoing context just for that device,
  then you shouldn't have a problem setting it to whatever you want, as
  only that device will use it.
 
  On 12/6/05, Matt [EMAIL PROTECTED] wrote:
   Hrmm that works except that my accountcode is not the extension of the
   customer/user, but is a distinct accountcode (ID).
  
   Oooo... you are setting the accountcode when you GET the
   call.  I guess I could do that... before I go to do too much work, is
   there a way to get asterisk to know the accountcode for the inbound
   call?
  
   On 12/6/05, Andy Kuo [EMAIL PROTECTED] wrote:
I use SetAccount(${EXTEN}) when the extension gets the call.  The 
original
dialed extension will be recorded as AccountCode in CDR, before the 
call is
forwarded.  The 1st field in CDR will be the extension your customer, 
the
2nd will be the caller (source), the 3rd will be the forwared number.
   
It works for me pretty well.
   
Andy
   
   
On 12/6/05, Matt [EMAIL PROTECTED] wrote:

 I want to allow my users to be able to
 Call Forward Unconditional
 Call Forward Busy
 Call Forward No Answer

 And curently I am doing this via my ATA and phone settings, however
 this has the problem that when a call is forwarded it goes out without
 an accountcode (Even though the ATA is forwarding the call), and hence
 I can't track the call!

 Can someone suggest a way to either fix this so that accountcodes go
 into the CDRs when the ATA/phone forwards the call, or to do the three
 forwarding types directly on asterisk?
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Re: [Asterisk-Users] Call Forwarding with Account Code.. can it be done?

[Darren Wiebe]

You have to do that from the dialplan.  I have a script that looks up 
the DID in a database and sets the accountcode.  It does some other 
stuff also but that could easily be cut out.  It's part of ASTPP.  Drop 
me a line if you need a copy.


Darren Wiebe

Matt wrote:


Hrmm that works except that my accountcode is not the extension of the
customer/user, but is a distinct accountcode (ID).

Oooo... you are setting the accountcode when you GET the
call.  I guess I could do that... before I go to do too much work, is
there a way to get asterisk to know the accountcode for the inbound
call?

On 12/6/05, Andy Kuo [EMAIL PROTECTED] wrote:
 


I use SetAccount(${EXTEN}) when the extension gets the call.  The original
dialed extension will be recorded as AccountCode in CDR, before the call is
forwarded.  The 1st field in CDR will be the extension your customer, the
2nd will be the caller (source), the 3rd will be the forwared number.

It works for me pretty well.

Andy


On 12/6/05, Matt [EMAIL PROTECTED] wrote:
   


I want to allow my users to be able to
Call Forward Unconditional
Call Forward Busy
Call Forward No Answer

And curently I am doing this via my ATA and phone settings, however
this has the problem that when a call is forwarded it goes out without
an accountcode (Even though the ATA is forwarding the call), and hence
I can't track the call!

Can someone suggest a way to either fix this so that accountcodes go
into the CDRs when the ATA/phone forwards the call, or to do the three
forwarding types directly on asterisk?
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http://lists.digium.com/mailman/listinfo/asterisk-users
   


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--
Darren Wiebe
[EMAIL PROTECTED]
Aleph Communications
ASTPP - Open Source Voip Billing  Calling Cards
www.aleph-com.net/astpp

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Re: [Asterisk-Users] Call Forwarding

[Jesse Keating]

On Thu, 2005-10-20 at 14:54 -0400, Dave Morrow wrote:
 Hi all.  I am attempting to setup a dial plan which will allow me to
 forward an extension.  I have followed the instructions in
 http://www.voip-info.org/wiki/index.php?page=Asterisk%20call%
 20forwarding however it does not work correctly.  Does anyone have
 some expertise they could lend.
 
 Not sure if it matters, but when I setup as in these instructions, and
 attempt to call forward my phone, asterisk logs when in fact I am
 attempting to forward to extension 8001 ;

Post your extensions.conf excerpt where you're trying to do the
forwarding.  I do something as silly-easy as:

exten = 5799,1,Goto(sipphones,5713,1)

Which takes calls coming into 5799 and instead directs them to 5713
within the sipphones context.

-- 
Jesse Keating
GameHouse -- Systems Engineer

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Re: [Asterisk-Users] Call forwarding

[Adrian A]

Thanks, that actually helps a lot.  
One problem I have (kind of unrelated) is with the AGI script
requiring two arguments.  You have:
exten = s,1,AGI(forward-get.agi,internal,${MACRO_EXTEN})
On my Asterisk installation that somehow passes the two arguments
internal and ${MACRO_EXTEN} as one argument to the bash script causing
the blank check for ${exten} to exit the script.  I have even tried
other suggestions such as:
exten = s,1,AGI(forward-get.agi|internal${MACRO_EXTEN}) or
exten = s,1,AGI,forward-get.agi,internal ${MACRO_EXTEN}
I'm running a recent version of CVS HEAD.
Asterisk CVS-HEAD built by [EMAIL PROTECTED] on a i686
running Linux on 2005-07-07 18:42:16



On 7/25/05, Cullin J. Wible [EMAIL PROTECTED] wrote:
 1) You could use asterisk realtime and a mysql database.
 
 2) You could use an asterisk database and allow users to set call forwarding
 by calling an extension.
 
 3) You could write some scripts to use an external database (what we did)
 and either allow users to update their forwarding options via a web page or
 telephone.
 
 I have attached some simple shell AGI-scripts and parts of our dial-plan so
 you can see how it all works. We authenticate against the mysql voicemail
 database and then our standard extension macro checks the database, possibly
 adding another channel to the dial command.
 
 I hope this helps.
 
 Cullin
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Adrian A
 Sent: Monday, July 25, 2005 4:15 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [Asterisk-Users] Call forwarding
 
 Is there an easy way to allow the users to go to a webpage or dial an
 extension and enter a phone number that their extension can be
 forwarded to?
 I'm using SER+Asterisk so doing this in sip.conf for example would not
 work since all users are registered to SER.  Currently in
 extensions.conf I have:
 exten = s,2,Dial(SIP/[EMAIL PROTECTED],20)
 Is there a way to check that the user at ${ARG1} has setup forwarding
 and retrieve the forwarding destination?
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Re: [Asterisk-Users] Call forwarding

[Adrian A]

Turns out it should actually be:
exten = s,1,AGI(agi-script.agi|arg1|arg2)

On 7/26/05, Adrian A [EMAIL PROTECTED] wrote:
 Thanks, that actually helps a lot.
 One problem I have (kind of unrelated) is with the AGI script
 requiring two arguments.  You have:
 exten = s,1,AGI(forward-get.agi,internal,${MACRO_EXTEN})
 On my Asterisk installation that somehow passes the two arguments
 internal and ${MACRO_EXTEN} as one argument to the bash script causing
 the blank check for ${exten} to exit the script.  I have even tried
 other suggestions such as:
 exten = s,1,AGI(forward-get.agi|internal${MACRO_EXTEN}) or
 exten = s,1,AGI,forward-get.agi,internal ${MACRO_EXTEN}
 I'm running a recent version of CVS HEAD.
 Asterisk CVS-HEAD built by [EMAIL PROTECTED] on a i686
 running Linux on 2005-07-07 18:42:16
 
 
 
 On 7/25/05, Cullin J. Wible [EMAIL PROTECTED] wrote:
  1) You could use asterisk realtime and a mysql database.
 
  2) You could use an asterisk database and allow users to set call forwarding
  by calling an extension.
 
  3) You could write some scripts to use an external database (what we did)
  and either allow users to update their forwarding options via a web page or
  telephone.
 
  I have attached some simple shell AGI-scripts and parts of our dial-plan so
  you can see how it all works. We authenticate against the mysql voicemail
  database and then our standard extension macro checks the database, possibly
  adding another channel to the dial command.
 
  I hope this helps.
 
  Cullin
 
  -Original Message-
  From: [EMAIL PROTECTED]
  [mailto:[EMAIL PROTECTED] On Behalf Of Adrian A
  Sent: Monday, July 25, 2005 4:15 PM
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  Subject: [Asterisk-Users] Call forwarding
 
  Is there an easy way to allow the users to go to a webpage or dial an
  extension and enter a phone number that their extension can be
  forwarded to?
  I'm using SER+Asterisk so doing this in sip.conf for example would not
  work since all users are registered to SER.  Currently in
  extensions.conf I have:
  exten = s,2,Dial(SIP/[EMAIL PROTECTED],20)
  Is there a way to check that the user at ${ARG1} has setup forwarding
  and retrieve the forwarding destination?
  ___
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RE: [Asterisk-Users] Call forwarding

[Cullin J. Wible]

1) You could use asterisk realtime and a mysql database.

2) You could use an asterisk database and allow users to set call forwarding
by calling an extension.

3) You could write some scripts to use an external database (what we did)
and either allow users to update their forwarding options via a web page or
telephone.

I have attached some simple shell AGI-scripts and parts of our dial-plan so
you can see how it all works. We authenticate against the mysql voicemail
database and then our standard extension macro checks the database, possibly
adding another channel to the dial command.

I hope this helps.

Cullin

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adrian A
Sent: Monday, July 25, 2005 4:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Call forwarding

Is there an easy way to allow the users to go to a webpage or dial an
extension and enter a phone number that their extension can be
forwarded to?
I'm using SER+Asterisk so doing this in sip.conf for example would not
work since all users are registered to SER.  Currently in
extensions.conf I have:
exten = s,2,Dial(SIP/[EMAIL PROTECTED],20)
Is there a way to check that the user at ${ARG1} has setup forwarding
and retrieve the forwarding destination?
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extensions-vmauth.conf
Description: Binary data


forward-get.agi
Description: Binary data


forward-set.agi
Description: Binary data


voicemail-auth.agi
Description: Binary data


extensions.conf
Description: Binary data


extensions-forward.conf
Description: Binary data
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RE: [Asterisk-Users] call forwarding, most basic case

[Mike Hillerbrand]

Try this
http://www.voip-info.org/tiki-index.php?page=Asterisk+Tips+follow+me

I used and it works well. Rather than segregate calls based on caller ID, it
carries the caller's ID through to the forwarded phone (cell phone, or
other?), but inserts a 0 before the number, that way you know it is an *
related call. If you don't answer (don't like the caller) or can't answer,
the call goes to voice mail.

Mike.


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Yair Hakak
Sent: Saturday, July 02, 2005 3:06 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] call forwarding, most basic case


hello all,

i need some help and after trying the wiki i'm even more confused than i
was.

 i'm trying to set up call forwarding and running into problems...
 i want the most basic call forwarding imaginable.

1. caller dials extension (say, 154)
2. dialplan is updated to forward caller's extension (based on
CALLERIDNUM) to voicemail, instead of ringing his endpoint.
3. caller is disconnected.

as you can see, i don't want any *21 or #21, and then the number, i
dont even want the caller to be able to pick the number to forward to,
the simplest case possible, and a different extension (155) to turn
the forwarding off (for now, then i'll put them in a menu together or
something.)

so, i know i need an extension like this:

exten =154,1, Answer
exten = 154,2,DBput(CFIM/${CALLERIDNUM}=Voicemail(u{CALLERIDNUM})
exten =153,3, Hangup

but line 2 is giving me fits, and the documentation is a bit thin. i'm
confused about the families in the database - do i have to create
them, or are they aready there?

of course, if i'm barking up the wrong tree and there's a much simpler
way to do this please tell me.

thanks,
 yair
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RE: [Asterisk-Users] call forwarding, most basic case

[Mike Hillerbrand]

By user do you mean the caller (initiator of the call) or the recipient? If
you mean that user is the call recipient, it is very easy. The caller's call
comes to you with its Caller ID--if you want the call to go to VM, then
don't answer the call. I use this for forwarding to other PSTN lines (cell,
remote offices, etc..), although I would guess the same thing applies to SIP
phones. The dial plan variables are only necessary if you want to pass
caller ID from the originating caller through to the forwarded number. If
you don't use the variable then the caller ID you would see would be that
from the Asterisk configuration and not from the actual caller.

The 0 inserted into the number is helpful if you have calls forwarded
simultaneously to your cell phone (or other) so that you can see by the zero
that it is a forwarded call rather than a direct call to your PSTN number (I
guess you could also use this with internal calls to distinguish calls that
are forwarded from different extension numbers). If it is a forwarded call
then by not answering it, it would go to Asterisk VM. If a direct call, it
would go to whatever aswering funtion is set up on your cell phone (or other
PSTN phone). [Please reply through the mailing list]. Mike.

-Original Message-
From: Yair Hakak [mailto:[EMAIL PROTECTED]
Sent: Saturday, July 02, 2005 5:05 PM
To: Mike Hillerbrand
Subject: Re: [Asterisk-Users] call forwarding, most basic case


hi,
 thanks for your answer, but i'm not sure i understand. this dialplan says
1. call the extension
2. set a variable with the callerIDNum
3. dial out to the follow me number with a 0 prepended to the callerID
4. switch the callerID back to the original
5. go to voicemail

how does the user turn this on and off? that's what i'm trying to do
in my case. i want the user to be able to switch between asterisk
calling his extension and asterisk sending the call directly to
voicemail.

-yair

On 7/2/05, Mike Hillerbrand [EMAIL PROTECTED] wrote:
 Try this
 http://www.voip-info.org/tiki-index.php?page=Asterisk+Tips+follow+me

 I used and it works well. Rather than segregate calls based on caller ID,
it
 carries the caller's ID through to the forwarded phone (cell phone, or
 other?), but inserts a 0 before the number, that way you know it is an *
 related call. If you don't answer (don't like the caller) or can't answer,
 the call goes to voice mail.

 Mike.


 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] Behalf Of Yair Hakak
 Sent: Saturday, July 02, 2005 3:06 PM
 To: asterisk-users@lists.digium.com
 Subject: [Asterisk-Users] call forwarding, most basic case


 hello all,

 i need some help and after trying the wiki i'm even more confused than i
 was.

  i'm trying to set up call forwarding and running into problems...
  i want the most basic call forwarding imaginable.

 1. caller dials extension (say, 154)
 2. dialplan is updated to forward caller's extension (based on
 CALLERIDNUM) to voicemail, instead of ringing his endpoint.
 3. caller is disconnected.

 as you can see, i don't want any *21 or #21, and then the number, i
 dont even want the caller to be able to pick the number to forward to,
 the simplest case possible, and a different extension (155) to turn
 the forwarding off (for now, then i'll put them in a menu together or
 something.)

 so, i know i need an extension like this:

 exten =154,1, Answer
 exten = 154,2,DBput(CFIM/${CALLERIDNUM}=Voicemail(u{CALLERIDNUM})
 exten =153,3, Hangup

 but line 2 is giving me fits, and the documentation is a bit thin. i'm
 confused about the families in the database - do i have to create
 them, or are they aready there?

 of course, if i'm barking up the wrong tree and there's a much simpler
 way to do this please tell me.

 thanks,
  yair
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Re: [Asterisk-Users] call forwarding, most basic case

[Ferdy Riphagen]

Yair,


One option is like this:

1) User dials ext. 154 to activate call forward (to voicemail)
2) User dials ext. 155 to de-activate call forward
3) Macro to check incoming calls for database entry's
4) The local extention must use that macro (or other way of screening)

1)
exten = 154,1,Answer
exten = 154,2,Set(DB(CFIM/${CALLERIDNUM})=${CALLERIDNUM})
exten = 154,3,Hangup

2)
exten = 155,1,Answer
exten = 155,2,DBdel(CFIM/${CALLERIDNUM})
exten = 155,3,Hangup

3)
[macro-test]
exten = s,1,Set(CFIM=${DB(CFIM/${ARG1})})
exten = s,2,GotoIf(${CFIM} = CFIM/${ARG1}?1|1)
exten = s,3,Dial(${ARG2}|${ARG3}|${ARG5})

exten = 1,1,VoiceMail(u${CFIM})

4)
exten = 202,1,Macro(test|${EXTEN}|SIP/202|15||tr)


Regards,

/* Ferdy */

http://asterisk.nsec.nl
info(AT)nsec(DOT)nl



Yair Hakak wrote:
 hello all,
 
 i need some help and after trying the wiki i'm even more confused than i was.
 
  i'm trying to set up call forwarding and running into problems...
  i want the most basic call forwarding imaginable.
 
 1. caller dials extension (say, 154)
 2. dialplan is updated to forward caller's extension (based on
 CALLERIDNUM) to voicemail, instead of ringing his endpoint.
 3. caller is disconnected.
 
 as you can see, i don't want any *21 or #21, and then the number, i
 dont even want the caller to be able to pick the number to forward to,
 the simplest case possible, and a different extension (155) to turn
 the forwarding off (for now, then i'll put them in a menu together or
 something.)
 
 so, i know i need an extension like this:
 
 exten =154,1, Answer 
 exten = 154,2,DBput(CFIM/${CALLERIDNUM}=Voicemail(u{CALLERIDNUM}) 
 exten =153,3, Hangup
 
 but line 2 is giving me fits, and the documentation is a bit thin. i'm
 confused about the families in the database - do i have to create
 them, or are they aready there?
 
 of course, if i'm barking up the wrong tree and there's a much simpler
 way to do this please tell me.
 
 thanks,
  yair
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Re: [Asterisk-Users] call forwarding, most basic case

[Yair Hakak]

hello Mike,
 we are talking about very different things here. please look at my
original mail again. I want the call recipient to be able to toggle on
and off do not disturb. I don't want the phone to ring at all.

thanks,
 yair


On 7/3/05, Mike Hillerbrand [EMAIL PROTECTED] wrote:
 By user do you mean the caller (initiator of the call) or the recipient? If
 you mean that user is the call recipient, it is very easy. The caller's call
 comes to you with its Caller ID--if you want the call to go to VM, then
 don't answer the call. I use this for forwarding to other PSTN lines (cell,
 remote offices, etc..), although I would guess the same thing applies to SIP
 phones. The dial plan variables are only necessary if you want to pass
 caller ID from the originating caller through to the forwarded number. If
 you don't use the variable then the caller ID you would see would be that
 from the Asterisk configuration and not from the actual caller.
 
 The 0 inserted into the number is helpful if you have calls forwarded
 simultaneously to your cell phone (or other) so that you can see by the zero
 that it is a forwarded call rather than a direct call to your PSTN number (I
 guess you could also use this with internal calls to distinguish calls that
 are forwarded from different extension numbers). If it is a forwarded call
 then by not answering it, it would go to Asterisk VM. If a direct call, it
 would go to whatever aswering funtion is set up on your cell phone (or other
 PSTN phone). [Please reply through the mailing list]. Mike.
 
 -Original Message-
 From: Yair Hakak [mailto:[EMAIL PROTECTED]
 Sent: Saturday, July 02, 2005 5:05 PM
 To: Mike Hillerbrand
 Subject: Re: [Asterisk-Users] call forwarding, most basic case
 
 
 hi,
  thanks for your answer, but i'm not sure i understand. this dialplan says
 1. call the extension
 2. set a variable with the callerIDNum
 3. dial out to the follow me number with a 0 prepended to the callerID
 4. switch the callerID back to the original
 5. go to voicemail
 
 how does the user turn this on and off? that's what i'm trying to do
 in my case. i want the user to be able to switch between asterisk
 calling his extension and asterisk sending the call directly to
 voicemail.
 
 -yair
 
 On 7/2/05, Mike Hillerbrand [EMAIL PROTECTED] wrote:
  Try this
  http://www.voip-info.org/tiki-index.php?page=Asterisk+Tips+follow+me
 
  I used and it works well. Rather than segregate calls based on caller ID,
 it
  carries the caller's ID through to the forwarded phone (cell phone, or
  other?), but inserts a 0 before the number, that way you know it is an *
  related call. If you don't answer (don't like the caller) or can't answer,
  the call goes to voice mail.
 
  Mike.
 
 
  -Original Message-
  From: [EMAIL PROTECTED]
  [mailto:[EMAIL PROTECTED] Behalf Of Yair Hakak
  Sent: Saturday, July 02, 2005 3:06 PM
  To: asterisk-users@lists.digium.com
  Subject: [Asterisk-Users] call forwarding, most basic case
 
 
  hello all,
 
  i need some help and after trying the wiki i'm even more confused than i
  was.
 
   i'm trying to set up call forwarding and running into problems...
   i want the most basic call forwarding imaginable.
 
  1. caller dials extension (say, 154)
  2. dialplan is updated to forward caller's extension (based on
  CALLERIDNUM) to voicemail, instead of ringing his endpoint.
  3. caller is disconnected.
 
  as you can see, i don't want any *21 or #21, and then the number, i
  dont even want the caller to be able to pick the number to forward to,
  the simplest case possible, and a different extension (155) to turn
  the forwarding off (for now, then i'll put them in a menu together or
  something.)
 
  so, i know i need an extension like this:
 
  exten =154,1, Answer
  exten = 154,2,DBput(CFIM/${CALLERIDNUM}=Voicemail(u{CALLERIDNUM})
  exten =153,3, Hangup
 
  but line 2 is giving me fits, and the documentation is a bit thin. i'm
  confused about the families in the database - do i have to create
  them, or are they aready there?
 
  of course, if i'm barking up the wrong tree and there's a much simpler
  way to do this please tell me.
 
  thanks,
   yair
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Re: [Asterisk-Users] Call forwarding...

[Peter Bowyer]

In 18/05/05, Mark Benson [EMAIL PROTECTED] wrote:
 
 I'm trying to setup a call forwarding rule so that when an extention
 doesn't answer the call is forwarded to my mobile.
 
 I'm using voiptalk.org for incoming and outgoing calls and SIP phones
 for extentions (so all IP based - no real phone lines).
 
 I tried this (from voip-info.org wiki)...
 
 exten = 1234,1,dial(sip/1234,20)
 exten = 1234,2,playback(pls-wait-connect-call)
 exten = 1234,3,Setvar(NewCaller=${CALLERIDNUM})
 exten = 1234,4,SetCIDNum(0${CALLERIDNUM})
 exten = 1234,5,dial(${TRUNK}c/9871234321,20,r)
 exten = 1234,6,SetCIDNum(${NewCaller})
 exten = 1234,7,voicemail2([EMAIL PROTECTED])
 exten = 1234,101,voicemail2([EMAIL PROTECTED])
 exten = 1234,102,hangup
 
 Mine looks like this...
 
 exten = 08700688nnn,1,Dial(SIP/operator,1,t)
 exten = 08700688nnn,2,playback(pls-wait-connect-call)
 exten = 08700688nnn,3,Setvar(NewCaller=${CALLERIDNUM})
 exten = 08700688nnn,4,SetCIDNum(0${CALLERIDNUM})
 exten = 08700688nnn,5,Dial(${TRUNK}/07961106nnn,20,r)
 exten = 08700688nnn,6,SetCIDNum(${NewCaller})
 exten = 08700688nnn,7,Voicemail(u100)
 exten = 08700688nnn,8,Hangup()
 exten = 08700688nnn,101,Voicemail(b100)
 exten = 08700688nnn,102,Hangup()
 
 (where nnn is a real number)
 The sip channel is set to time out quickly for testing.
 And I don't appear to have the pls-wait-connect-call audio file - but
 that isn't an issue for the time being...
 The IAX2/0870n is the extention/device that calls go out on via
 voiptalk... (my call provider)...
 If I include the c/ in the TRUNK line I get...
 
   -- Executing Dial(IAX2/[EMAIL PROTECTED]:4569-1,
 c/07961106nnn|20|r) in new stack
 May 18 10:37:40 WARNING[24416]: channel.c:1957 ast_request: No channel
 type registered for 'c'
 May 18 10:37:40 NOTICE[24416]: app_dial.c:927 dial_exec_full: Unable to
 create channel of type 'c' (cause 66)

Have you set the TRUNK variable in the [globals] section of
extensions.conf? Looks like you didn't.

Peter

-- 
Peter Bowyer
Email: [EMAIL PROTECTED]
Tel: +44 1296 768003
VoIP: sip:[EMAIL PROTECTED]
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Re: [Asterisk-Users] Call forwarding...

[Mark Benson]

Er... set the trunk variable to what? I thought it was a built in 
variable...

Peter Bowyer wrote:
Have you set the TRUNK variable in the [globals] section of
extensions.conf? Looks like you didn't.
Peter
 

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Re: [Asterisk-Users] Call forwarding...

[Peter Bowyer]

On 18/05/05, Mark Benson [EMAIL PROTECTED] wrote:
 Er... set the trunk variable to what? I thought it was a built in
 variable...

No, it's not. Looking at your dialplan extract, you need to set TRUNK
to the name of the trunk to place the outgoing call on.

eg

TRUNK=IAX/voiptalk

You might need to mess around to get the dialstring to end up in the
right format for the provider you're using, also. Or imbed it directly
in the dialplan.

Peter
-- 
Peter Bowyer
Email: [EMAIL PROTECTED]
Tel: +44 1296 768003
VoIP: sip:[EMAIL PROTECTED]
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Re: [Asterisk-Users] Call forwarding...

[Mark Benson]

I have been able to get it working by explicitly setting the dial command...
So should the trunk variable be the divice to dial out on?
Mark Benson wrote:
Er... set the trunk variable to what? I thought it was a built in 
variable...

Peter Bowyer wrote:
Have you set the TRUNK variable in the [globals] section of
extensions.conf? Looks like you didn't.
Peter
 

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Re: [Asterisk-Users] Call forwarding...

[Mark Benson]

Thanks,
Staring to see where I was going wrong. Now I know the explicit dial 
string (as you say I tried that in the dial plan and it worked) I can 
mess around with the trunk variable.

Cheers!
Peter Bowyer wrote:
On 18/05/05, Mark Benson [EMAIL PROTECTED] wrote:
 

Er... set the trunk variable to what? I thought it was a built in
variable...
   

No, it's not. Looking at your dialplan extract, you need to set TRUNK
to the name of the trunk to place the outgoing call on.
eg
TRUNK=IAX/voiptalk
You might need to mess around to get the dialstring to end up in the
right format for the provider you're using, also. Or imbed it directly
in the dialplan.
Peter
 

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RE: [Asterisk-Users] Call forwarding...

[Kanuri, Seshu (Company IT)]

Try changing SetCIDNum SetCallerID and use to SetCIDName as under:

Ex:
---
exten = s, 1, SetCallerID(${CALLERIDNUM})
exten = s, 2, SetCIDName(${CALLERIDNAME})
exten = s, 3, Dial(${ARG2}/${ARG1},${RINGSECS})
exten = s, 4, Voicemail(u${ARG1})
exten = s, 5, Hangup
exten = s, 101, Voicemail(b${ARG1})
exten = s, 102, Hangup
 
Seshu Kanuri


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark
Benson
Sent: Wednesday, May 18, 2005 6:26 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Call forwarding...

Sorry for posting this again, but it seems to have become attached to
another thread. Guess I replied to another message instead of starting a
new one...

Hi,

I'm trying to setup a call forwarding rule so that when an extention
doesn't answer the call is forwarded to my mobile.

I'm using voiptalk.org for incoming and outgoing calls and SIP phones
for extentions (so all IP based - no real phone lines).

I tried this (from voip-info.org wiki)...

exten = 1234,1,dial(sip/1234,20)
exten = 1234,2,playback(pls-wait-connect-call)
exten = 1234,3,Setvar(NewCaller=${CALLERIDNUM})
exten = 1234,4,SetCIDNum(0${CALLERIDNUM}) exten =
1234,5,dial(${TRUNK}c/9871234321,20,r)
exten = 1234,6,SetCIDNum(${NewCaller})
exten = 1234,7,voicemail2([EMAIL PROTECTED]) exten =
1234,101,voicemail2([EMAIL PROTECTED])
exten = 1234,102,hangup

Mine looks like this...

exten = 08700688nnn,1,Dial(SIP/operator,1,t)
exten = 08700688nnn,2,playback(pls-wait-connect-call)
exten = 08700688nnn,3,Setvar(NewCaller=${CALLERIDNUM})
exten = 08700688nnn,4,SetCIDNum(0${CALLERIDNUM})
exten = 08700688nnn,5,Dial(${TRUNK}/07961106nnn,20,r)
exten = 08700688nnn,6,SetCIDNum(${NewCaller})
exten = 08700688nnn,7,Voicemail(u100)
exten = 08700688nnn,8,Hangup()
exten = 08700688nnn,101,Voicemail(b100) exten =
08700688nnn,102,Hangup()

(where nnn is a real number)
The sip channel is set to time out quickly for testing.
And I don't appear to have the pls-wait-connect-call audio file - but
that isn't an issue for the time being...
The IAX2/0870n is the extention/device that calls go out on via
voiptalk... (my call provider)...
If I include the c/ in the TRUNK line I get...

   -- Executing Dial(IAX2/[EMAIL PROTECTED]:4569-1,
c/07961106nnn|20|r) in new stack
May 18 10:37:40 WARNING[24416]: channel.c:1957 ast_request: No channel
type registered for 'c'
May 18 10:37:40 NOTICE[24416]: app_dial.c:927 dial_exec_full: Unable to
create channel of type 'c' (cause 66)

Asterisk shows this from the moment the sip channel is considered not to
have answered (1 sec)...

   -- Nobody picked up in 1000 ms
   -- Executing Playback(IAX2/[EMAIL PROTECTED]:4569-1,
pls-wait-connect-call) in new stack
May 18 10:20:26 WARNING[24416]: file.c:486 ast_openstream_full: File
pls-wait-connect-call does not exist in any format May 18 10:20:26
WARNING[24416]: file.c:790 ast_streamfile: Unable to open
pls-wait-connect-call (format ilbc): No such file or directory May 18
10:20:26 WARNING[24416]: app_playback.c:83 playback_exec: 
ast_streamfile failed on IAX2/[EMAIL PROTECTED]:4569-1 for
pls-wait-connect-call
   -- Executing SetVar(IAX2/[EMAIL PROTECTED]:4569-1,
NewCaller=01202843nnn) in new stack
   -- Executing SetCIDNum(IAX2/[EMAIL PROTECTED]:4569-1,
001202843nnn) in new stack
   -- Executing Dial(IAX2/[EMAIL PROTECTED]:4569-1,
/07961106nnn|20|r) in new stack
May 18 10:20:26 WARNING[24416]: channel.c:1957 ast_request: No channel
type registered for ''
May 18 10:20:26 NOTICE[24416]: app_dial.c:927 dial_exec_full: Unable to
create channel of type '' (cause 66)  == Everyone is busy/congested at
this time (1:0/0/1)
   -- Executing SetCIDNum(IAX2/[EMAIL PROTECTED]:4569-1,
01202843nnn) in new stack
   -- Executing VoiceMail(IAX2/[EMAIL PROTECTED]:4569-1,
u100) in new stack
   -- Playing '/var/spool/asterisk/voicemail/default/100/unavail' 
(language 'en')

Again - I'm not worried about the audio file warning - I can fix that
later... I guess this is the important bit...

   -- Executing Dial(IAX2/[EMAIL PROTECTED]:4569-1,
/07961106nnn|20|r) in new stack
May 18 10:20:26 WARNING[24416]: channel.c:1957 ast_request: No channel
type registered for ''
May 18 10:20:26 NOTICE[24416]: app_dial.c:927 dial_exec_full: Unable to
create channel of type '' (cause 66)  == Everyone is busy/congested at
this time (1:0/0/1)

The call then drops into voicemail...

I've tried various permuations but still no call is made to the mobile
number. Any ideas?

Cheers,

Mark

I should mention that I have tried using the call forward function of
the sip phones, but a) this means configuring the phones and some are
remote and behind firewalls and b) It doesn't work... 

 
NOTICE: If received in error, please destroy and notify sender.  Sender does 
not waive confidentiality or privilege, and use is prohibited. 
 
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Re: [Asterisk-Users] Call forwarding...

[Peter Bowyer]

On 18/05/05, Mark Benson [EMAIL PROTECTED] wrote:
 I have been able to get it working by explicitly setting the dial command...
 
 So should the trunk variable be the divice to dial out on?

Yes. But I wouldn't worry if you can't get it to work - the use of a
variable substitution in that case is simply a convenience in the
dialplan - you don't need to do it that way if you don't want.

Peter

-- 
Peter Bowyer
Email: [EMAIL PROTECTED]
Tel: +44 1296 768003
VoIP: sip:[EMAIL PROTECTED]
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Re: [Asterisk-Users] Call Forwarding / Redirect with PRI

[Peter Svensson]

On Tue, 17 May 2005, Lenwood S. Sawyer, III wrote:

 I have a PRI from Bellsouth going to my asterisk box with a Digium 
 Wildcard TE110P.  I would like to be able to use call forwarding without 
 having to use two channels.  Is it possible to use call redirect with a 
 PRI.  Does the BRIstuff package help at all?

What flavour of switch are you connected to? 

For Lucent 5ESS 2 B Channel Transfer is implemented in libpri. I think
it will be used automatically if the conditions are suitable (no dtmf
detection going on etc).

BRIstuff provides ECT and CD which are used by EuroISDN. These are called 
explicitly from the dialplan, not automatically like the 2BCT above.

Peter


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Re: [Asterisk-Users] Call Forwarding / Redirect with PRI

[Lenwood S. Sawyer, III]

Switch is NI2.
Peter Svensson wrote:
On Tue, 17 May 2005, Lenwood S. Sawyer, III wrote:

I have a PRI from Bellsouth going to my asterisk box with a Digium 
Wildcard TE110P.  I would like to be able to use call forwarding without 
having to use two channels.  Is it possible to use call redirect with a 
PRI.  Does the BRIstuff package help at all?

What flavour of switch are you connected to? 

For Lucent 5ESS 2 B Channel Transfer is implemented in libpri. I think
it will be used automatically if the conditions are suitable (no dtmf
detection going on etc).
BRIstuff provides ECT and CD which are used by EuroISDN. These are called 
explicitly from the dialplan, not automatically like the 2BCT above.

Peter
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Re: [Asterisk-Users] Call Forwarding to Cell Phone, Pager, etc

[Andrew Kohlsmith]

On March 2, 2005 02:56 pm, Nitesh Divecha wrote:
 For example, if exten 202 is away, he will set his call blasting priority
 like first ring the exten for 10 sec if not answered then ring cell number
 for 10 sec again if not answered, then ring home and etc.

This is indeed possible.  You need to break down the problem so you can see 
what you have to do in the dialplan.

Basically you want to Dial() his extension for 10 seconds and failing that 
Dial() something else (and failing that Dial() something else, and so on and 
so forth, eventually going to VoiceMail())

Everything you want (including setting timeouts for how long to ring) is in 
the Dial() application, and Asterisk has pretty good documentation on the 
Dial command, both on the wiki and also in the show application dial 
command output in the Asterisk CLI.

If you are still having trouble with this after trying out a few things, post 
here again, showing us specifically what you've tried and we'll help some 
more.

-A.
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RE: [Asterisk-Users] Call Forwarding to Cell Phone, Pager, etc

[Adam Robins]

Yes.
http://lists.digium.com/pipermail/asterisk-users/2005-February/087538.ht
ml

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nitesh
Divecha
Sent: Wednesday, March 02, 2005 2:56 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Call Forwarding to Cell Phone, Pager, etc

Hello all,

Was just wondering if Asterisk can do Call forwarding to cell phones,
pagers, home phone, etc.

For example, if exten 202 is away, he will set his call blasting
priority like first ring the exten for 10 sec if not answered then ring
cell number for 10 sec again if not answered, then ring home and etc.

Like a call blasting priority...

Any help would be appreciated.

Neel



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Re: [Asterisk-Users] Call Forwarding to Cell Phone, Pager, etc

[Richard J. Sears]

Hi Neel - 

We did it by dialing several devices at once, then calling the cell and
then pulling the call back if no one answered the cell to allow the
local vm to handle vm functions. Here is how we did it:



[macro-stdexten_cell]
; ARG1 = Greeting
; ARG2 = Extension(s) to dial
; ARG3 = Cell Phone Number to dial
;
exten = s,1,Playback(${ARG1})
exten = s,2,Dial(${ARG2},15,rtm)
exten = s,3,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Dial(IAX2/[EMAIL PROTECTED]/${ARG3},15,rtm)
exten = s-NOANSWER,2,Voicemail(u${MACRO_EXTEN})
exten = s-CHANUNAVAIL,1,Voicemail(b${MACRO_EXTEN}) ;if chan 
unavail (sip phone not regisitered?)
;exten = s-NOANSWER,3,Goto(default,s,1)
exten = s-BUSY,1,Voicemail(b${MACRO_EXTEN})
;exten = s-BUSY,2,Goto(default,s,1)
exten = s-.,1,Goto(s-NOANSWER,1)
exten = a,1,VoicemailMain(${MACRO_EXTEN})


Then here is the extension entry that made it happen:

exten = 
${RJSWORK},1,Macro(stdexten_cell,sears_welcome,SIP/${RJSWORK}SIP/${RJSDESK}SIP/${RJSLAPTOP}SIP/${RJSHOME},${RJSCELL})

All the variables are just the extensions numbers.

Hope this helps





On Wed, 2 Mar 2005 11:56:14 -0800
Nitesh Divecha [EMAIL PROTECTED] wrote:

 Hello all,
 
 Was just wondering if Asterisk can do Call forwarding to cell phones,
 pagers, home phone, etc.
 
 For example, if exten 202 is away, he will set his call blasting priority
 like first ring the exten for 10 sec if not answered then ring cell number
 for 10 sec again if not answered, then ring home and etc.
 
 Like a call blasting priority...
 
 Any help would be appreciated.
 
 Neel
 
 
 
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Re: [Asterisk-Users] Call forwarding

[William Waites]

Wow. This list is high traffic Just to add to the noise, here's
some of my extensions.conf that implements what you are talking about.
In particular, the macro featureexten takes an argument that is the
same as the context the user uses for outbound dialing. The result
being that whatever number the user puts in the CFIM database is
dialed in the same way that it would be if the user dialed it directly
from their telephone. It is used like this:

exten = 5551212,1,Macro(featureexten,usercontext,SIP/whatever,${EXTEN})

Normally [usercontext] will include = [features]

Of course beware that this is only accessible from lines that have their
callerid forced to something reasonable, else you can steal lines...

[features]
; Unconditional Call Forward
exten = _*21X.,1,Answer()
exten = _*21X.,2,DBput(CFIM/${CALLERIDNUM}=${EXTEN:3})
exten = _*21X.,3,Playback(unconditional)
exten = _*21X.,4,Playback(call-forwarding)
exten = _*21X.,5,SayDigits(${EXTEN:3})
exten = _*21X.,6,Hangup()
exten = #21,1,Answer()
exten = #21,2,DBdel(CFIM/${CALLERIDNUM})
exten = #21,3,Playback(unconditional)
exten = #21,4,Playback(call-forwarding)
exten = #21,5,Playback(disabled)
exten = #21,6,Hangup()

; Call Forward on Busy or Unavailable
exten = _*61X.,1,Answer()
exten = _*61X.,2,DBput(CFBS/${CALLERIDNUM}=${EXTEN:3})
exten = _*61X.,3,Playback(call-forwarding)
exten = _*61X.,4,Playback(on-busy)
exten = _*61X.,5,SayDigits(${EXTEN:3})
exten = _*61X.,6,Hangup
exten = #61,1,Answer()
exten = #61,2,DBdel(CFBS/${CALLERIDNUM})
exten = #61,3,Playback(call-forwarding)
exten = #61,4,Playback(on-busy)
exten = #61,5,Playback(disabled)
exten = #61,6,Hangup()

; Hide Caller-ID
exten = _*67X.,1,SetCIDNum()
exten = _*67X.,2,SetCIDName(UNKNOWN)
exten = _*67X.,3,Goto(${EXTEN:3},1)

; Last Number
exten = *69,1,Answer()
exten = *69,2,DBget(temp=LCN/${CALLERIDNUM})
exten = *69,3,Playback(last-num-to-call)
exten = *69,4,SayDigits(${temp})
exten = *69,5,Hangup
exten = *69,103,Playback(im-sorry)
exten = *69,104,Playback(num-not-in-db)
exten = *69,105,Hangup

; Call-Centre
exten = *66,1,AgentLogin(${CALLERIDNUM})

; Voicemail
exten = *98,1,VoiceMailMain(${CALLERIDNUM})

[macro-featureexten]
;
; Standard extension macro (with call forwarding):
; ${ARG1} - forwarding context/dialplan
; ${ARG2} - Device(s) to ring
; ${ARG3} - voicemailbox
;
exten = s,1,DBPut(LCN/${MACRO_EXTEN}=${CALLERIDNUM})
exten = s,2,DBget(temp=CFIM/${MACRO_EXTEN}) ; Get CFIM key, if not existing, 
goto 102
exten = s,3,Goto(${ARG1},${temp},1)  ; unconditional forward
exten = s,4,ChanisAvail({ARG2})
exten = s,5,Dial(${AVAILORIGCHAN},30)
exten = s,6,DBget(temp=CFBS/${MACRO_EXTEN}) ; Get CFBS key, if not existing, 
goto 105
exten = s,7,Goto(${ARG1},${temp},1) ; Forward on busy or unavailable

; No CFIM key
exten = s,103,Goto(s,4)

; No available channels
exten = s,105,Goto(s,107)

; No CFBS key - voicemail ?
exten = s,107,VoiceMail(u${ARG3})


On Fri, Feb 04, 2005 at 09:22:21AM -0500, Adam Robins wrote:
 I've written a macro that allows users to dynamically change their call
 forwarding destination.  The purpose is to set up a follow me process
 where a user can get calls on their cell, at home, etc., based on the
 forwarding number they enter.  Using the CFIM database, I have the setup
 portion working great.  Now, I want to actually use that information to
 forward a call.  
  
 Here is my issue:  The forwarded number saved in CFIM could be another
 extension, a local number or an LD number, each of which would be dialed
 using a different technology (internal, SIP-provider, Zap, etc.).  I
 want to avoid having to check the number and code all of the logic for
 each method - because I already have all of this set up in the dialplan
 for callers who would have dialed this forwarded number directly.
  
 What I would like to do is take the variable containing the number
 retrieved from CFIM, place it on the stack as the called number, and
 have it reenter the dial plan, similar to the WAITEXTEN command.
  
 Any ideas are appreciated!
  
 For those interested, here is the Forwarding Setup macro:
  
 ;
 ; Call forwarding Macro
 ;
 [macro-forwarding]
 exten = s,1,Answer
 exten = s,2,Wait(1)
 exten = s,3,DigitTimeout(3)
 exten = s,4,ResponseTimeout(10)
 exten = s,5,Read(fwext,fw-extension,2); ask
 extension (2 digits)
 exten = s,6,Authenticate(/etc/asterisk/authFWD)   ; only
 authorized individuals
 exten = s,7,Playback(fw-extension); repeat back
 extension
 exten = s,8,SayNumber(${fwext},f)
 exten = s,9,DBget(fwnum=CFIM/${fwext}); check if
 already forwarded
  
 ; ext is forwarded
 exten = s,10,Playback(fw-is-forwarded-to) ; play
 forwarded number from database
 exten = s,11,SayDigits(${fwnum})
 exten = s,12,Read(resp,fw-cancel-1-change-2,1); 1 to cancel
 fwd, 2 to change #
 exten = s,13,GotoIf($[${resp} = 1]?17:14) ; 1 entered,
 goto delete
 exten = s,14,GotoIf($[${resp} = 

Re: [Asterisk-Users] Call Forwarding and CDR

[Matthew Boehm]

I'm sure someone will mention ForkCDR() but that doesn't really work. I am
in the same situation where a person may dial the DID 8229698858 but the CDR
will show that persons extension rather than the actual number that was
dialed.

-Matthew

- Original Message - 
From: aza [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thursday, February 17, 2005 2:12 PM
Subject: [Asterisk-Users] Call Forwarding and CDR


 Hi,

 Does anyone have a solution to that allows an incoming call
 to be forwarded to a mobile (or other billable destination)
 and provide a CDR for the mobile call.

 Such as:

 exten - 1001,Dial(SIP/sipclientZAP/g1/023423,20)

 The CDR for this call will have 1001 as the destination
 regardless of whether the SIP or ZAP channel takes it. The
 problem I have is if the call is forwarded to the
 ZAP/g1/023423 channel it costs money and I want to find a
 way to get a CDR for this leg of the call with the 023423
 number as the destination and not 1001.

 Anyone have something similar?

 Aaron


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Re: [Asterisk-Users] Call forwarding

[Kevin P. Fleming]

Adam Robins wrote:
What I would like to do is take the variable containing the number
retrieved from CFIM, place it on the stack as the called number, and
have it reenter the dial plan, similar to the WAITEXTEN command.
Easy-peasy. That's what Local channels are for.
If you have retrieved the CFIM value into a variable called CFIM, and 
the user's phone would normally dial via a context called 
customer-dial, then:

exten = ...,...,Dial(Local/[EMAIL PROTECTED])
This will process the call exactly the same way as if the phone user had 
dialed that number from their phone.
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Re: [Asterisk-Users] Call forwarding

[Ryan Courtnage]

Hi Kevin,

On Fri, 2005-04-02 at 08:09 -0700, Kevin P. Fleming wrote:
 Adam Robins wrote:
 
  What I would like to do is take the variable containing the number
  retrieved from CFIM, place it on the stack as the called number, and
  have it reenter the dial plan, similar to the WAITEXTEN command.
 
 Easy-peasy. That's what Local channels are for.
 
 If you have retrieved the CFIM value into a variable called CFIM, and 
 the user's phone would normally dial via a context called 
 customer-dial, then:
 
 exten = ...,...,Dial(Local/[EMAIL PROTECTED])

Can multiple Local channels be safely used in a single Dial command?
ie:

exten
= ...,...,Dial(Local/[EMAIL PROTECTED],Local/[EMAIL PROTECTED],Local/[EMAIL 
PROTECTED])

Thanks
Ryan

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Re: [Asterisk-Users] Call forwarding

[Andrew Willerding]

 I've written a macro that allows users to dynamically change their call
 forwarding destination.  The purpose is to set up a follow me process
 where a user can get calls on their cell, at home, etc., based on the
 forwarding number they enter.  Using the CFIM database, I have the setup
 portion working great.  Now, I want to actually use that information to
 forward a call.  
  
 Here is my issue:  The forwarded number saved in CFIM could be another
 extension, a local number or an LD number, each of which would be dialed
 using a different technology (internal, SIP-provider, Zap, etc.).  I
 want to avoid having to check the number and code all of the logic for
 each method - because I already have all of this set up in the dialplan
 for callers who would have dialed this forwarded number directly.
  
 What I would like to do is take the variable containing the number
 retrieved from CFIM, place it on the stack as the called number, and
 have it reenter the dial plan, similar to the WAITEXTEN command.
  


I've done something similar.  I have a relatively small PBX configuration and
all extensions are SIP phones assigned a 6 digit or less extension number. 
I've built a custom macro that determines how many digits need to be dialed and
then based on the number of digits it either prefixes SIP/ or Zap/  to the
actual Dial command.  It works like a charm!

Andrew
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Re: [Asterisk-Users] Call forwarding

[Kevin P. Fleming]

Ryan Courtnage wrote:
Can multiple Local channels be safely used in a single Dial command?
ie:
exten
= ...,...,Dial(Local/[EMAIL PROTECTED],Local/[EMAIL PROTECTED],Local/[EMAIL 
PROTECTED])
Yes, using the standard  connector like you would use if you were 
dialing multiple SIP peers or any other peers.
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Re: [Asterisk-Users] Call forwarding

[Pedro]

Cool idea.

One question - let's say someone specifies their home phone number and
their cell number.  How do you take into the account if the cell VM
picks up (ie. if cell is out of coverage and VM greeting is played)?


On Fri, 04 Feb 2005 10:41:28 -0700, Kevin P. Fleming
[EMAIL PROTECTED] wrote:
 Ryan Courtnage wrote:
 
  Can multiple Local channels be safely used in a single Dial command?
  ie:
 
  exten
  = ...,...,Dial(Local/[EMAIL PROTECTED],Local/[EMAIL 
  PROTECTED],Local/[EMAIL PROTECTED])
 
 Yes, using the standard  connector like you would use if you were
 dialing multiple SIP peers or any other peers.
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RE: [Asterisk-Users] Call forwarding

[Adam Robins]

Works beautifully!  Thanks. 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin P.
Fleming
Sent: Friday, February 04, 2005 10:10 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Call forwarding

Adam Robins wrote:

 What I would like to do is take the variable containing the number 
 retrieved from CFIM, place it on the stack as the called number, and 
 have it reenter the dial plan, similar to the WAITEXTEN command.

Easy-peasy. That's what Local channels are for.

If you have retrieved the CFIM value into a variable called CFIM, and
the user's phone would normally dial via a context called
customer-dial, then:

exten = ...,...,Dial(Local/[EMAIL PROTECTED])

This will process the call exactly the same way as if the phone user had
dialed that number from their phone.
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Re: [Asterisk-Users] Call forwarding

[Ryan Courtnage]

On Fri, 2005-04-02 at 13:11 -0500, Pedro wrote:
 Cool idea.
 
 One question - let's say someone specifies their home phone number and
 their cell number.  How do you take into the account if the cell VM
 picks up (ie. if cell is out of coverage and VM greeting is played)?

AFAIK, there isn't much you can do in this scenario - other than ringing
your house for a few rings before ringing your house AND the cell.  Even
then, the cell provider's 'out of the service area' message would answer
the call.


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