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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> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
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[asterisk-users] Call forwarding in Asterisk

[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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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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[asterisk-users] Call forwarding from Phones and getting the referrer IP

[Ishfaq Malik]

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

-- 

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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[asterisk-users] Call Forwarding / Follow-Me on PRI

[Barry D. Hassler]

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.

Most carriers are restricting CALLING Number ID to be one of the numbers
allocated to the associated circuit. This makes sense from a perspective of
call-fraud prevention.

We have clients that used call forwarding or follow-me extensively, and
configured to send the ORIGINAL callerID as the Calling ID, so when the
call shows up on their cell phone, it appears to be coming from the
originator. This capability seems to be going away.

Some legacy PRI carriers were not (and perhaps continue to be) so strict
about it, and a recent client who was used to this feature, is now unhappy
that their new PRI carrier does not allow any callingID other than one
associated with the PRI.

A workaround or RNIE (Redirecting Number Information Element) has been
recommended as an alternative, but that does not appear to be standardized,
nor implemented in asterisk. The only PBX vendors that appear to support
this in even a limited sense are Cisco and Shoretel.

I'm curious if others have encountered this same situation (I'm sure you
have), or any other pertinent thoughts.

Thanks in advance!


-- 
Barry D. Hassler
President, HCST

http://www.hcst.com/
937-427-9000
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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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[asterisk-users] Call Forwarding

[dotnetdub]

Hi Guys,

Seeing an issue with 1.6.2.17.2 and also 1.6.2.14

When we do call forwarding if the call coming in to be forwarded
asterisk sends the invite out to our ITSP as
username@anonymous.invalid instead of username@domain.

When call comes in with CLI and is forwarded it sends it as
username@domain to our ITSP.

Is this a bug or is there something I need to turn on or off? All the
ITSP's we use authenticate on username and domain.

Thanks
Brian.

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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

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

[Alessio]

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.

[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

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

 --
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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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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[asterisk-users] call forwarding

[salaheddine elharit]

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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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 :)

-- 
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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)

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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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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[asterisk-users] call forwarding callerID

[Gerard]

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)

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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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[asterisk-users] Call Forwarding

[Dan Journo]

Hi,

I'm currently programming an interface for my Asterisk service.

I've noticed an issue if someone sets up call forwarding on their sip phone.
Asterisk receives a 302 Moved Temporarily message, and forwards the call 
successfully.

However, the CDR is not correct.

If I set up call forwarding to a mobile on extension 201, and then place a call 
from extension 202, the call gets diverted.
I answer the call and talk for 30 seconds, then I hang up.

The CDR shows two calls:-

2010-08-27 13:38:24 - 202 - 201 - Answered - Billsec is 30
2010-08-27 13:38:24 - 202 - 5551234 - Answered - Billsec is 0

5551234 is the mobile number.
The second CDR entry should read 30 seconds, and the first should read 0 (or 30)

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

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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

-- 
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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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[asterisk-users] Call Forwarding to Voicemail

[Beebob007]

Hi, the following configuration:

The number 0 will be forwarded to the Ring-Group 25 in which the numbers 
are 71 and 73. If you call the 0 so the office is ringing at the 71 and 
73 .

At the terminal stations are Snom 320.
In the evening the 71 to make call forwarding via web interface to the 
99 (voice mail).


Problem:
When I ring the phonenumber 0 after call forwarding .the 73 ringing 
and the Voice Mail (99) didn?t take the call ..
But when I call the 71 after call forwarding all works fine and the 
Voicemail take the Call



How can the Number 99 prioritization so that they take the Call in any 
case ... even if the 73 are still ringing?

Or there other options?

Regards
Beebob

in Deutsch: (hoffe das lesen hier welche die Deutsch sprechen...da das 
oben nur Googletranslate war)



Hallo, folgende Konfiguration:

Die Rufnummer 0 wird auf die Ring Group 25 weitergeleitet in der sich 
die Rufnummern 71 und 73 befinden. Ruft man also die 0 an über Amt 
klingelt es bei der 71 und 73..

An den Endstellen befinden sich Snom 320.
Abends soll die 71 per Weboberfläche eine Umleitung auf die 99 
(Voicemail) machen.


Problem:
Wenn ich die 0 nach erfolgter Rufumleitung anrufe klingelt die 73 munter 
vor sich hin und die Voicemail (99)geht nicht ran..
Rufe ich aber die 71 über Amt an funktioniert die Rufumleitung auf die 
99 wunderbar.



Wie kann ich die 99 bevorrechtigen, so dass sie auf jeden Fall 
rangeht...auch wenn die 73 noch klingelt bzw gibt es noch andere 
Möglichkeiten?


Gruß
Beebob
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[asterisk-users] Call forwarding, callerID, and e911

[John A. Sullivan III]

We were able to solve the below problem.  I'll post it in case someone
encounters the same issue.  No need to respond or even read unless you
see a better way.  Thanks - John

We have manually set callerID on our outbound lines to reflect the
appropriate DID both for e911 and to be polite to folks we call, e.g.:

exten = _1NXXNXX,1,Set(CALLERID(num)=5197546340)
exten = _1NXXNXX,n,Goto(outbound-US,${EXTEN},1)

This is working perfectly fine (numbers changed to protect the
innocent!) until someone forwards their phone.  When someone calls in
for them, the forwarded call becomes an outbound call and we are
overwriting the callerID rather than showing the original callerID.

Is there some way that I'm missing to distinguish between an outbound
call and a forwarded outbound call? Is there a better way to do what we
are doing? Thanks - John

We solved this with the following logic assuming outside callerIDs were
at least 7 digits long and internal extensions were less than 7 digits
(numbers changed to protect the real numbers):

exten = _1NXXNXX,1,ExecIf($[${CALLERID(num)} - 100  
0]?Set(CALLERID(num)=6715728792))


-- 
John A. Sullivan III
Open Source Development Corporation
+1 207-985-7880
jsulli...@opensourcedevel.com

http://www.spiritualoutreach.com
Making Christianity intelligible to secular society


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[asterisk-users] Call Forwarding Lopp Prevention

[Paradise Dove]

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

when both extensions are Busy a large number of stale calls will be
made in the system!
how can i prevent this mess in my system?

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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

-- 
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[asterisk-users] call forwarding

[gilbert saunders]

hi 
  im starting off with asterisk and i need to know how to do call forarding...
  in out old telephony system we used to press *21*number# and all the calls 
would be forwarded to that number and we used #21# to undivert is that possible 
in asterisk and how do i do it 
   
  i have attached my extensions.conf file and would appreciate it if you could 
help me out with some code if its possible to make such a diversion work
   
  urgent
   
  thank you in advance

   
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[asterisk-users] Call forwarding-in india

[sandeep]

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

Regards,
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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?

-- 
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[asterisk-users] Call forwarding (from PHONE configuration) with PRI

[Barry D. Hassler]

Hi folks.

My client is wanting to use call forwarding configured on their phones
(Linksys SPA942), with a PRI from their provider. When we configure call
forwarding, we invariably get a The number you have dialed is not in
service message from the providers.

Examining the detailed dial plan debugging as well as the PRI debugging,
the number is dialed correctly. The only difference noticed between a
forwarded call versus a normal outbound call is that an extended
attribute of Forwarded Unconditionally.

The (slightly edited -- I've edited the phone numbers to protect the
innocent) trace is below. Is this an asterisk issue, or an ILEC issue? I
have a ticket open with the ILEC as well, but they tend to be less than
helpful (their response has been to turn on their call forwarding option).

Any thoughts appreciated!


[ 02 01 01 1e ]

   -- Got SIP response 302 Moved Temporarily back from 192.168.1.55
mail*CLI

   -- Now forwarding Zap/1-1 to
'Local/[EMAIL PROTECTED]' (thanks to SIP/2105-08217980)
   -- Executing Macro(Local/[EMAIL PROTECTED],2,
to-pstn-standard|1937000) in new stack
   -- Executing NoOp(Local/[EMAIL PROTECTED],2,
CALLERID(num)= 937001) in new stack
   -- Executing NoOp(Local/[EMAIL PROTECTED],2,
MACRO_EXTEN= 1937000) in new stack
   -- Executing GotoIf(Local/[EMAIL PROTECTED],2,
0?:7) in new stack
   -- Goto (macro-to-pstn-standard,s,7)
   -- Executing NoOp(Local/[EMAIL PROTECTED],2, Call
Forwarded) in new stack
   -- Executing Set(Local/[EMAIL PROTECTED],2,
CALLERID(name)=CLIENT NAME) in new stack
   -- Executing Set(Local/[EMAIL PROTECTED],2,
CALLERID(num)=513001) in new stack
   -- Executing Dial(Local/[EMAIL PROTECTED],2,
Zap/g1/1937000||r) in new stack
   -- Requested transfer capability: 0x00 - SPEECH
mail*CLI


[04 03 80 90 a2]
Bearer Capability (len= 5) [ Ext: 1  Q.931 Std: 0  Info transfer

capability: Speech (0)

 Ext: 1  Trans mode/rate: 64kbps,

circuit-mode (16)

 Ext: 1  User information layer 1: u-Law

(34)

[18 03 a9 83 82]
Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0, Exclusive

Dchan: 0

   ChanSel: Reserved
  Ext: 1  Coding: 0   Number Specified   Channel

Type: 3

  Ext: 1  Channel: 2 ]
[1e 02 80 83]
Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)

0: 0   Location: User (0)

  Ext: 1  Progress Description: Calling

equipment is non-ISDN. (3) ]

[28 19 b1 22 46 57 44 20 46 52 4f 4d 20 42 4c 55 45 20 4c 4f 4f 50 20

4c 4c 43 22]

Display (len=25) Charset: 31 [ FWD FROM x ]
[6c 0c 49 80 35 31 33 32 30 34 32 31 30 30]
Calling Number (len=14) [ Ext: 0  TON: Subscriber Number (4)  NPI:

Private Numbering Plan (9)

  Presentation: Presentation permitted, user

number not screened (0) '513001' ]

[70 0c c9 31 39 33 37 34 32 37 39 30 30 30]
Called Number (len=14) [ Ext: 1  TON: Subscriber Number (4)  NPI:

Private Numbering Plan (9) '1937000' ]

[74 07 49 01 8f 32 31 30 35]
Redirecting Number (len= 9) [ Ext: 0  TON: Subscriber Number (4)  NPI:

Private Numbering Plan (9)

  Ext: 0 Presentation: Presentation

permitted, user number passed network screening (1)

  Ext: 1 Reason: Forwarded unconditionally

(15) '2105' ]
   -- Called g1/1937000
   -- Local/[EMAIL PROTECTED],1 is ringing
mail*CLI


--
Barry D. Hassler
President, HCST

http://www.hcst.net/
937-427-9000
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[asterisk-users] Call Forwarding

[Rob Schall]

An odd question...

I have asterisk running just basic sip phones and sending/receiving
calls using ZAP. The phones are polycom 501s.

When a user presses the Forward soft key and puts an external number
(a cell phone), and then someone from the inside (another extension) to
the phone which has the forward on... I get this odd and loud
humming/buzz noise in place of what the ringer normally would be. Once
the call completes, its fine. If you dial from the outside into the SIP
phone, the forward happens just fine.

Any thoughts?

Rob

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[asterisk-users] Call forwarding

[Dominik Zalewski]

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

[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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[asterisk-users] Call forwarding....

[Leeno Jose .P.A]

Hi Friends,

I have installed Asterisk Version 1.2.13 and its working fine with 
*X-Lite* soft phone. Now I want to forward call to another local 
phone(Hardware) if nobody is attending the call at dialed extension. 
What should i do???



Thanks in advance...


--
Leeno Jose.P.A
System Administrator



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[asterisk-users] Call Forwarding Using Asterisk

[jk]

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

[Zeeshan Zakaria]

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?

[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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[asterisk-users] Call forwarding

[Vladimir Dvorak]

Hello to all asterisk users, 

I have a problem with call forwarding.

My extensions.conf:

[outbound]
exten =  _*22*XXX,1,Set(DB(CFIM/${CALLERID(num)})=${EXTEN:4})
exten =  _*22*,1,DBdel(CFIM/${CALLERID(num)})

Have three stations, 301, 302 and 303. When dial on 301 following
number:

*22*302

it should redirect all calls targeted to 301 to number 302. But it
doesn`t work.

If anyone of you has experience with call forwarding, your help will be
appreciated. Thank you very much.

Vladimir 

Here is SIP output:

---

-- SIP read from 192.168.0.10:5060:
ACK sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bK-4b96baec
From: 301 sip:[EMAIL PROTECTED];tag=c3d74f8b3bb05e94o0
To: sip:[EMAIL PROTECTED];tag=as3e772365
Call-ID: [EMAIL PROTECTED]
CSeq: 101 ACK
Max-Forwards: 70
Contact: 301 sip:[EMAIL PROTECTED]:5060
User-Agent: Sipura/SPA2002-3.1.2(a)
Content-Length: 0


--- (10 headers 0 lines)---

-- SIP read from 192.168.0.10:5060:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bK-dd489610
From: 301 sip:[EMAIL PROTECTED];tag=c3d74f8b3bb05e94o0
To: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
Max-Forwards: 70
Proxy-Authorization: Digest
username=301,realm=asterisk,nonce=46339822,uri=sip:[EMAIL 
PROTECTED],algorithm=MD5,response=3284e4149a3abe9e0c4c454af19aa7b5
Contact: 301 sip:[EMAIL PROTECTED]:5060
Expires: 240
User-Agent: Sipura/SPA2002-3.1.2(a)
Content-Length: 422
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp

v=0
o=- 437796 437796 IN IP4 192.168.0.10
s=-
c=IN IP4 192.168.0.10
t=0 0
m=audio 16406 RTP/AVP 0 2 4 8 18 96 97 98 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

--- (15 headers 19 lines)---
Using INVITE request as basis request - [EMAIL PROTECTED]
Sending to 192.168.0.10 : 5060 (NAT)
Found user '301'
Found RTP audio format 0
Found RTP audio format 2
Found RTP audio format 4
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 96
Found RTP audio format 97
Found RTP audio format 98
Found RTP audio format 100
Found RTP audio format 101
Peer audio RTP is at port 192.168.0.10:16406
Found description format PCMU
Found description format G726-32
Found description format G723
Found description format PCMA
Found description format G729a
Found description format G726-40
Found description format G726-24
Found description format G726-16
Found description format NSE
Found description format telephone-event
Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x51d (g723|ulaw|alaw|
g726|g729|ilbc)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Looking for *22*302 in outbound (domain 192.168.0.1)
list_route: hop: sip:[EMAIL PROTECTED]:5060
Transmitting (NAT) to 192.168.0.10:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.0.10:5060;branch=z9hG4bK-dd489610;received=192.168.0.10
From: 301 sip:[EMAIL PROTECTED];tag=c3d74f8b3bb05e94o0
To: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:[EMAIL PROTECTED]
Content-Length: 0


---
-- Executing Set(SIP/301-503d, DB(CFIM/301)=302) in new stack
-- Executing Hangup(SIP/301-503d, ) in new stack
Reliably Transmitting (NAT) to 192.168.0.10:5060:
SIP/2.0 603 Declined
Via: SIP/2.0/UDP
192.168.0.10:5060;branch=z9hG4bK-dd489610;received=192.168.0.10
From: 301 sip:[EMAIL PROTECTED];tag=c3d74f8b3bb05e94o0
To: sip:[EMAIL PROTECTED];tag=as0c8d34d4
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:[EMAIL PROTECTED]
Content-Length: 0


---
Retransmitting #1 (NAT) to 192.168.0.10:5060:
SIP/2.0 603 Declined
Via: SIP/2.0/UDP
192.168.0.10:5060;branch=z9hG4bK-dd489610;received=192.168.0.10
From: 301 sip:[EMAIL PROTECTED];tag=c3d74f8b3bb05e94o0
To: sip:[EMAIL PROTECTED];tag=as0c8d34d4
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:[EMAIL PROTECTED]
Content-Length: 0



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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

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---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.


-- 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 ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

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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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[asterisk-users] Call Forwarding in SIP.conf

[broadbandvoice]

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.

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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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 http://lists.digium.com/mailman/listinfo/asterisk-users











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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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 http://lists.digium.com/mailman/listinfo/asterisk-users












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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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[asterisk-users] call forwarding to mobile phone

[Lito Lampitoc]

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

[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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[asterisk-users] call forwarding

[Ever Zalazar]

Hi people. I want to know about call forwarding. I 
dial *72, and a message say me to dial the extension , I did, then the message 
said is forward is UNCONDITIONLA . But when I call , it doesn't work the 
forwarding.
Who can help me please.

Best Regards

Ever
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[Asterisk-Users] Call forwarding for particular extension when line 1 is busy

[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



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[Asterisk-Users] Call forwarding for particular extension

[nr k]

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



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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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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
 
 
 
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[Asterisk-Users] Call Forwarding with Account Code.. can it be done?

[Matt]

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]

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

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[Asterisk-Users] Call Forwarding

[Abdul Lateef]

Hi all,

I have one external VoIP terminator, I need to forward
all calls to that terminator i did some configuration
in sip.conf but i am confiused what will be the
configuration in extentions.conf to forward all calls
to that terminator.

  sip.conf

[general]
register =
450102:201079:[EMAIL PROTECTED]:5060/450102

i found that 450102 user successfully registered on
terminator.

Now i want to register Grandstreem using 450102 user
on Asterisk Server and using this want to forward call
using the same username to the terminator.

[user]
type=friend
username=450102
secret=201079
fromuser=450102
authuser=450102
context=allcall
allow=g729


extentions.conf

[allcall]
exten = 

Please advice me how i can run this configuration.



Yours,
Abdul Lateef
Computer Programmer
HATIF COM
Mob: +974 - 5405022
Tel: +974 - 4883068
ICQ: 276994704
YM!: abdul_zu
Fax: +974 - 4883063
Doha Qatar
http://www.hatif.com




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[Asterisk-Users] Call Forwarding

[Dave Morrow]

Title: Call Forwarding






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 ;

 == Spawn extension (default, *21*800, 4) exited non-zero on 'SIP/8001-9be7'



David A. Morrow

Technical Systems Lead

Autodata Solutions Company

[EMAIL PROTECTED]

http://www.autodata.net

Tel: (519) 951-6079

Fax: (519) 451-6615 


 Poor planning on your part does not necessarily constitute an emergency on my part! 


This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]


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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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