Re: [asterisk-users] Call Transfer Fails - Not a Valid Extension

[Scott Griepentrog]

​The file /var/log/asterisk/full will contain helpful log messages that
show how Asterisk is internally handling the call.  It may be necessary to
increase the verbosity of the log to get more details however.

>From the linux command line, you can follow these steps to get a copy of
the relevant messages:

# asterisk -rx "core set verbose 5"

# cat /var/log/asterisk/full > mylogfile

(perform a transfer that fails with the message now, then press CTRL-C to
cancel the above command)

The mylogfile will have the log entries necessary to understand what
happened, although it may also require an understanding of the FreePBX
dialplan to interpret it.  If you can post your log file (recommend using a
pastebin rather than emailing the whole thing) it should be ​fairly easy to
spot the problem and advise you how to fix it.


On Sun, Sep 7, 2014 at 10:55 PM, Phil Ledon  wrote:

>  We have a plain vanilla installation of AsteriskNOW using Digium D40/50
> phones. All transfers are failing from any source to any extension with the
> message “that is not a valid extension”. Does anyone have any ideas about
> where to begin looking for the source of that error?
>
>
>
> *Phil Ledon*
>
>
>
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Re: [asterisk-users] Call transfer problem.

[Igor Zamocky]

You have to use "attendant" transfer, not "blind".

- A calls B
- B answers on "line 1" (button 1)
- B has to use "line 2" (push button 2) to call C, C sees call coming from
B, the same does asterisk
- while having "line 2" active, he pushes button "transfer" followed by
button "line 1"
- A speaks with C


On Mon, Feb 24, 2014 at 7:45 PM, Mike Diehl  wrote:

> I'm sorry, I should have mentioned that he's doing a "phone-based"
> transfer, not an "asterisk-based" transfer.
>
> Mike.
>
> On Mon, Feb 24, 2014 at 1:30 PM, Don Kelly  wrote:
> > Does he complete the call as a "supervised" transfer--waits for the
> called
> > party to answer before completing the transfer?
> >
> >   --Don
> >
> >
> > -Original Message-
> > From: asterisk-users-boun...@lists.digium.com
> > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Diehl
> > Sent: Monday, February 24, 2014 12:24 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: [asterisk-users] Call transfer problem.
> >
> > Hi all,
> >
> > I have a user who is having trouble transferring calls, using a
> Grandstream
> > GXP2xxx.
> >
> > Here's the use case that I've seen:
> >
> > I call the user from phone A and he answers on phone B.
> >
> > Then, he hits the transfer button on his phone and dials an extension
> that
> > is reachable by him, but not by me, based on administrative policy.
> >
> > However, the Asterisk logs indicate that the new call is being initiated
> by
> > phone A, not phone B!  Thus the call transfer fails.
> >
> > I have other users, with other phones, that are able to transfer just
> fine.
> > What could be different with this particular user?
> >
> > Any ideas?
> >
> > Mike.
> >
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Re: [asterisk-users] Call transfer problem.

[Mike Diehl]

I'm sorry, I should have mentioned that he's doing a "phone-based"
transfer, not an "asterisk-based" transfer.

Mike.

On Mon, Feb 24, 2014 at 1:30 PM, Don Kelly  wrote:
> Does he complete the call as a "supervised" transfer--waits for the called
> party to answer before completing the transfer?
>
>   --Don
>
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Diehl
> Sent: Monday, February 24, 2014 12:24 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] Call transfer problem.
>
> Hi all,
>
> I have a user who is having trouble transferring calls, using a Grandstream
> GXP2xxx.
>
> Here's the use case that I've seen:
>
> I call the user from phone A and he answers on phone B.
>
> Then, he hits the transfer button on his phone and dials an extension that
> is reachable by him, but not by me, based on administrative policy.
>
> However, the Asterisk logs indicate that the new call is being initiated by
> phone A, not phone B!  Thus the call transfer fails.
>
> I have other users, with other phones, that are able to transfer just fine.
> What could be different with this particular user?
>
> Any ideas?
>
> Mike.
>
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Re: [asterisk-users] Call transfer problem.

[Don Kelly]

Does he complete the call as a "supervised" transfer--waits for the called
party to answer before completing the transfer?

  --Don


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Diehl
Sent: Monday, February 24, 2014 12:24 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Call transfer problem.

Hi all,

I have a user who is having trouble transferring calls, using a Grandstream
GXP2xxx.

Here's the use case that I've seen:

I call the user from phone A and he answers on phone B.

Then, he hits the transfer button on his phone and dials an extension that
is reachable by him, but not by me, based on administrative policy.

However, the Asterisk logs indicate that the new call is being initiated by
phone A, not phone B!  Thus the call transfer fails.

I have other users, with other phones, that are able to transfer just fine.
What could be different with this particular user?

Any ideas?

Mike.

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Re: [asterisk-users] Call Transfer question

[qasimak...@gmail.com]

Hi faheem,

You can do this:

ACTION: Redirect
Channel: 
Context: 
Exten: 
Priority: 

Regards,
Qasim


On Thu, May 16, 2013 at 3:13 PM, Muhammad Faheem wrote:

> Hi,
> is possible that two sip extensions: user-1 and user-2 are connected and I
> want to transfer the call from user-1 to a third user "user-3".
> I know it is possible through feature keys mapping in features.conf, but I
> want to do this through AMI or Asterisk CLI Commands?
>
> Please suggest if possible?
>
> Thank you!
> Muhammad Faheem
>
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Re: [asterisk-users] Call Transfer not working

[Rizwan Hisham]

Thanks everyone. I was using the Tt flag but in the wrong place in the dial
application.

Cheers

On Mon, Apr 9, 2012 at 4:54 PM, Takehiro Matsushima <
takehiro.dream...@gmail.com> wrote:

> Thank you so much.
>
> OK, I understood that to transfer the call "t" is usually used, is it
> right?
> And I should write so in my last mail.
>
> "t" and "T" are described with same sentences in official wiki...
>
> Regards,
> Takehiro Matsushima
>
>
>
> 2012/4/9 Chris Bagnall :
> > On 9/4/12 3:04 am, Takehiro Matsushima wrote:
> >>
> >> // I don't know what's difference "t" and "T".
> >
> >
> > T allows the caller to transfer. t allows the called user to transfer.
> >
> > You very rarely want "Tt" - since I doubt you want an incoming caller to
> be
> > able to transfer their call all over the place. You usually want "t" on
> > incoming calls and "T" on outgoing calls.
> >
> > Kind regards,
> >
> > Chris
> > --
> > This email is made from 100% recycled electrons
> >
> >
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Re: [asterisk-users] Call Transfer not working

[Takehiro Matsushima]

Thank you so much.

OK, I understood that to transfer the call "t" is usually used, is it right?
And I should write so in my last mail.

"t" and "T" are described with same sentences in official wiki...

Regards,
Takehiro Matsushima



2012/4/9 Chris Bagnall :
> On 9/4/12 3:04 am, Takehiro Matsushima wrote:
>>
>> // I don't know what's difference "t" and "T".
>
>
> T allows the caller to transfer. t allows the called user to transfer.
>
> You very rarely want "Tt" - since I doubt you want an incoming caller to be
> able to transfer their call all over the place. You usually want "t" on
> incoming calls and "T" on outgoing calls.
>
> Kind regards,
>
> Chris
> --
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>
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Re: [asterisk-users] Call Transfer not working

[Chris Bagnall]

On 9/4/12 3:04 am, Takehiro Matsushima wrote:

// I don't know what's difference "t" and "T".


T allows the caller to transfer. t allows the called user to transfer.

You very rarely want "Tt" - since I doubt you want an incoming caller to 
be able to transfer their call all over the place. You usually want "t" 
on incoming calls and "T" on outgoing calls.


Kind regards,

Chris
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Re: [asterisk-users] Call Transfer not working

[Takehiro Matsushima]

Hi.

Maybe you forgotten specify to allow the transferring a call.
Try with "tT" options in "Dial()" in extensions.conf.

// I don't know what's difference "t" and "T".

-- 
Takehiro Matsushima



2012/4/7 Rizwan Hisham :
> Hi All,
> I am using asterisk 1.8.11 on centos 5. I have realtime sip peers with dtmf
> setting rfc2833 and inband. I have also enabled blind and attended transfer
> features in features.conf but still call transfers dont work. I have setup
> transfer feature in past but i dont think i am missing anything this time. I
> just dont have any clue why its not working. I have tried using ATAs and
> softphones but cant make it to work. Can anyone help? Am I missing anything?
>
> features show output:
> ===
> Builtin Feature           Default Current
> ---           --- ---
> Pickup                    *8      *8
> Blind Transfer            #       #1
> Attended Transfer                 *2
> One Touch Monitor
> Disconnect Call           *       *
> Park Call
> One Touch MixMonitor
> ==
> --
> 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 transfer

[Gergo Csibra]

Tuesday, February 16, 2010, 12:55:12 PM, cool wrote:

> call comes it should be received by extenion 2000, n if person wants to
> talk to Sales, receptionist should put the caller on hold than connect
> to Sales i.e exten 2001, while on hold the caller should hear music on
> hold,now sale exten can take his call n talk to it.same with Accounts
> ext 2002.
...
> what to next

To have call transfer in your asterisk setup, YOU need to read some
documentation. Start here: http://www.voip-info.org

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 Gergomailto:csi...@gmail.com


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Re: [asterisk-users] call transfer

[Brian]

On Tue, 2010-02-16 at 17:25 +0530, cool dude wrote:
> call transfer
> 
> call transfer from reception to other extensions.
> 
> Question: Details of Extensions
> 
> Reception - 2000
> Sales - 2001
> Accounts - 2002
> 
> any call comes it should be received by extenion 2000, n if person
> wants to talk to Sales, receptionist should put the caller on hold
> than connect to Sales i.e exten 2001, while on hold the caller should
> hear music on hold,now sale exten can take his call n talk to it.same
> with Accounts ext 2002.
> 
> 
> 
> vi /etc/asterisk/sip.conf
> 
> [general]
> port = 5060
> bindaddr = 0.0.0.0
> context = others
> 
> 
> [2000]
> type=friend
> context=reception
> secret=1234
> host=dynamic
> 
> [2001]
> type=friend
> context=sales
> secret=1234
> host=dynamic
> 
> 
> [2002]
> type=friend
> context=accounts
> secret=1234
> host=dynamic
> ~ 
> ##
> 
> vi /etc/asterisk/extension.conf
> 
> [from-zaptel]
> exten => s,1,wait(2)
> exten => s,n,Dial(SIP/2000,20)
> 
> 
> what to next
> 
> 
> __
What have you tried? Which links/mans have you read to set up music on
hold? Are any of them wrong at all? 


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Re: [asterisk-users] call transfer using DTMF

[Brad Finberg]

"
Yes,
In the features.conf under featuremap you need the blindtransfer un-commented
blindxfer => ## 
Then in your extensions.conf you need to have at least a capital T
exten => example,1,Dial(ZAP/4/12345,,T)
Then during the call you can press ## and asterisk will say transfer.
Then dial in the extension you want to transfer too.

Thank you,
Brad Finberg


- Original Message -
From: Michael 
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Cc:
Date: Tuesday, July 14 2009 11:07 PM
Subject: [asterisk-users] call transfer using DTMF
Is there a way to transfer a call, while in the middle of the call, using 
DTMF?

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Re: [asterisk-users] call transfer using DTMF

[Matt Riddell]

On 15/7/09 3:07 PM, Michael wrote:
> Is there a way to transfer a call, while in the middle of the call, using
> DTMF?

Yep,  just pass the t or T options to the dial command and set it up in 
/etc/asterisk/features.conf

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Re: [asterisk-users] call transfer in CDR

[Grey Man]

On Thu, Jan 15, 2009 at 4:09 AM, Rilawich Ango  wrote:
> Hi,
>  I wonder how I can relate the CDR records for the case of call
> transfer.  I can't find their relationship in CDR.  Any can advice?
> ango
>

You may want to read this thread.

http://lists.digium.com/pipermail/asterisk-users/2008-January/204856.html

Regards,

Greyman.

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Re: [asterisk-users] Call transfer using agi

[Lenz Emilitri]

You could simply have it Dial()  to wherever it needs to go at the end of
the script.

2009/1/6 Rajkumar S 

> Hi,
>
> I have a typical call center with queues and agents added via
> AddQueueMember. One of my requirement is to implement a forgot
> password function. If a caller does not remember the password, he
> calls up an unauthenticated line and the agent manually authenticates
> him. Then the caller should have a provision to reset his password.
> The requirement is that the agent should not know the new password of
> caller.
>
> I have an agi to change password and can transfer call to agi, but I
> do not know how to transfer the call back to agent from agi.
>
> So basically how can an agi transfer a call to an extension?
>
> Thanks and regards,
>
> raj
>
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Re: [asterisk-users] Call transfer over IAX trunk

[Andrea Spadaccini]

Ciao Noah,

> What flags do you have in your Dial() statement?  If you want both
> parties to be able to transfer with the features.conf transfer, you
> need to have 'Tt' in your dial statement, like this:
> Dial(IAX2/user:[EMAIL PROTECTED]/exten,20,Tt)

Bingo. That was the problem.

Thanks a lot,

-- 
Dr. Andrea Spadaccini
Multimedia Technologies Institute - MTI S.r.l.
Web: www.x-voice.it - Tel: +39 (0) 95 7224945

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Re: [asterisk-users] Call transfer over IAX trunk

[Noah Miller]

Hi Andrea -

> I have two asterisk servers, an IAX trunk between and some SIP users 
> registered
> to each server.
>
> The scenario is this: user A, registered to PBX 1, calls user B, registered to
> PBX 2. Then A wants to transfer the call using the features.conf method (in my
> case, **), but is unable to do this.

What flags do you have in your Dial() statement?  If you want both
parties to be able to transfer with the features.conf transfer, you
need to have 'Tt' in your dial statement, like this:
Dial(IAX2/user:[EMAIL PROTECTED]/exten,20,Tt)


- Noah

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Re: [asterisk-users] Call Transfer

[randulo]

On Sat, Jun 7, 2008 at 8:24 AM, Theodore Patsiouras
<[EMAIL PROTECTED]> wrote:
> If my secretary or anyone else picks up the call when the line is transferred 
> in my ext then I have the > internal caller ID. Can I have somehow the 
> External callerID?

Look at the channel variables that contain the callerid information.
You can assign the incoming callerid to the one that makes the call to
your local extension to do what you wish.

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Re: [asterisk-users] call transfer detection in dial plan

[Atis]

On 9/13/07, Rilawich Ango <[EMAIL PROTECTED]> wrote:
> Hi all,
>   In default, we can use # to transfer the call.  I want to know how I
> can know the user presse # to transfer the call in dial plan.
> ango

Set TRANSFER_CONTEXT or GOTO_ON_BLINDXFER variable (depending on *
version) before Dial(). I just wrote more explaining mail to list.

Regards,
Atis

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Re: [asterisk-users] call transfer not working

[Rizwan Hisham]

check to see if you have dtmf=rfc2833 and canreinvite=no in sip.conf general
settings.

On 7/4/07, satish patel <[EMAIL PROTECTED]> wrote:


Dear all

  I have install asterisk 1.2.x and it is working fine my
setup is like

[*]---[Mediant2k][Avaya]

 Now i want to transfer call in internal extension i have read more
document on www.voip-info.com but it is now so much clear so if u have any
sample configuration file and doucment plz suggest me i have configure
feature.conf and extention.conf for this task

regards


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Re: [asterisk-users] Call transfer in asterisk

[satish patel]

Dear all
   
  i have read that document but dont understand about function i 
have include featuremap in extension.conf 
   
  [mysip]
  include => featuremap
   
  and reload extention.conf i got this error
   
  *CLI> extensions reload
Jul  2 19:23:04 WARNING[16320]: pbx.c:6444 ast_context_verify_includes: Context 
'mysip' tries includes nonexistent context 'featuremap'
*CLI>

   
  also i have chenged in feature.conf 
   
  [featuremap]
blindxfer => #1 ; Blind transfer
disconnect => *0; Disconnect
automon => *1   ; One Touch Record
atxfer => *2; Attended transfer
   
   
  why my inculde function not working properly 
   
  


Lee Jenkins <[EMAIL PROTECTED]> wrote:
  satish patel wrote:
> dear all
> 
> I am new in asterisk and i have now setup asterik for 
> 40 phone now i want to configure call transfer between phone so how it 
> is possible and what configuration part in asterisk will perfomed for 
> this task give me suggestion for my solution
> 
> Regards
> 
> Satish Patel
> 

And this:
http://www.voip-info.org/wiki-Asterisk+config+features.conf


Warm Regards,

Lee




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Re: [asterisk-users] Call transfer in asterisk

[Lee Jenkins]

satish patel wrote:
> dear all
> 
>  I am new in asterisk and i have now setup asterik for 
> 40 phone now i want to configure call transfer between phone so how it 
> is possible and what configuration part in asterisk will perfomed for 
> this task give me suggestion for my  solution
> 
> Regards
> 
> Satish Patel
> 

And this:
http://www.voip-info.org/wiki-Asterisk+config+features.conf


Warm Regards,

Lee




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Re: [asterisk-users] Call transfer in asterisk

[Dominik Zalewski]

On Monday 02 July 2007 01:45:44 pm satish patel wrote:
> dear all
>
>  I am new in asterisk and i have now setup asterik for 40
> phone now i want to configure call transfer between phone so how it is
> possible and what configuration part in asterisk will perfomed for this
> task give me suggestion for my  solution
>
> Regards
>
> Satish Patel
>
>
> -
> Yahoo! oneSearch: Finally,  mobile search that gives answers, not web
> links.

http://www.voip-info.org/wiki/index.php?page=PBX+CallTransfer

-- 
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t- +2 02 336 0003
w- http://www.open-craft.com

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Re: [asterisk-users] Call transfer feature

[Lee Jenkins]

satish patel wrote:
> Dear ALL
> 
>I want to transfer call from one phone 2 another 
> phone so this is asterisk feature or SIP Phone feature or endpoint 
> feature how can i transfer phone call from to another phone
> 
> 
> Rgd
> 
> Satish patel
> 

Check out this page:

http://www.voip-info.org/wiki-Asterisk+config+features.conf


-- 

Warm Regards,

Lee




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RE: [asterisk-users] CALL TRANSFER

[Damon Estep]

Be careful, if you set both T and t you might be allowing the wrong
party to transfer the call! In MOST cases you would want T or t, not T
and t, although there are some cases where you might want both.

 

 

 



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of omar
parihuana
Sent: Friday, December 01, 2006 5:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] CALL TRANSFER

 

Thanks!!!

 

I forget Tt option! (too basis!!)

 

On 12/1/06, Damon Estep <[EMAIL PROTECTED]> wrote: 

Your dial string must have either the t or T option set.

 



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] ] On Behalf Of omar
parihuana
Sent: Friday, December 01, 2006 9:10 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion 
Subject: [asterisk-users] CALL TRANSFER

 

Hi Guys,

 

I'm implementing my Asterisk step by step, so far the communications
between softphones, hardphones with Gateways, voice mail, are working
fine. Rightnow I would like to enable Call tranfer (like Traditional
PBX) in Blind Transfer and AttendXFER, I'm reading features.conf in
accordance to voip-info.org <http://voip-info.org/>  but the transfer
doesn't work!  Please if you can provide me some examples will be very
appreciate. 

 

Rgds.

-- 
Omar E.P.T
-
Certified Networking Professionals make better Connections!

http://omarept.blogspot.com/

  Usysnet Corp 
Open Source Solutions
www.usysnet.com.pe <http://www.usysnet.com.pe/>  


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http://omarept.blogspot.com/

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Re: [asterisk-users] CALL TRANSFER

[omar parihuana]

Thanks!!!

I forget Tt option! (too basis!!)


On 12/1/06, Damon Estep <[EMAIL PROTECTED]> wrote:


 Your dial string must have either the t or T option set.


  --

*From:* [EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED] *On Behalf Of *omar parihuana
*Sent:* Friday, December 01, 2006 9:10 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] CALL TRANSFER



Hi Guys,



I'm implementing my Asterisk step by step, so far the communications
between softphones, hardphones with Gateways, voice mail, are working fine.
Rightnow I would like to enable Call tranfer (like Traditional PBX) in Blind
Transfer and AttendXFER, I'm reading features.conf in accordance to
voip-info.org but the transfer doesn't work!  Please if you can provide me
some examples will be very appreciate.



Rgds.

--
Omar E.P.T
-
Certified Networking Professionals make better Connections!

http://omarept.blogspot.com/

  Usysnet Corp
Open Source Solutions
www.usysnet.com.pe

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http://omarept.blogspot.com/

 Usysnet Corp
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RE: [asterisk-users] CALL TRANSFER

[Damon Estep]

Your dial string must have either the t or T option set.

 



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of omar
parihuana
Sent: Friday, December 01, 2006 9:10 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] CALL TRANSFER

 

Hi Guys,

 

I'm implementing my Asterisk step by step, so far the communications
between softphones, hardphones with Gateways, voice mail, are working
fine. Rightnow I would like to enable Call tranfer (like Traditional
PBX) in Blind Transfer and AttendXFER, I'm reading features.conf in
accordance to voip-info.org but the transfer doesn't work!  Please if
you can provide me some examples will be very appreciate.

 

Rgds.

-- 
Omar E.P.T
-
Certified Networking Professionals make better Connections!

http://omarept.blogspot.com/

  Usysnet Corp 
Open Source Solutions
www.usysnet.com.pe 

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Re: [asterisk-users] Call transfer issues

[Kevin Smith]

My guess is I stumped everyone ;)

Anyway, I rolled back asterisk to 1.2.9.1 (same for libpri and zaptel 
back one release) and transfers were working again. Now I'm still quite 
new to asterisks, I know enough to hold my own, but not enough to know 
the full inter workings of it. But here is my thought:


Caller A calls in and talks to Employee B. B wants to transfer to C. 
Asterisk sets up the bridge between B and C. B completes the transfer. 
Now A and C are connected but there is no audio stream. If C or A puts 
the other on hold, and then resumes the call, audio is restored.


By that I would say placing them on hold clears a flag or updates one to 
connect the audio stream? Or am I way off on this assumption? Also if 
this sounds like a possible bug, what information do I need to include, 
or is good to include, when submitting bugs?


Thanks,
Kevin

Kevin Smith wrote:

Hey everyone,

Currently we are running IP601's with 1.6.6 SIP firmware on Asterisk 
1.2.10. It has been reported to me when doing an attended transfer the 
audio drops out. I ran a few different tests and here is what I noticed.


1. Blind transfers work with no problem.
2. Attended transfers were you transfer the call before the person 
picks up works.
3. If the person the call is being transferred to answers and then the 
transfer completes, the audio drops.


I noticed in the CLI the following (I replaced the number with XXX's)

-- Attempting native bridge of SIP/989XXX-b76167c8 and 
SIP/989XXX-08f956b8

 == Parsing '/etc/asterisk/manager.conf': Found
   -- Stopped music on hold on Zap/2-1
 == Spawn extension (Mercury-Directory-Dialer, 989XXX, 8) exited 
non-zero on 'SIP/989XXX-b76167c8'
   -- Executing Hangup("SIP/989XXX-b76167c8", "") in new 
stack
 == Spawn extension (Mercury-Directory-Dialer, h, 1) exited non-zero 
on 'SIP/989XXX-b76167c8'
   -- Incoming call: Got SIP response 500 "Internal Server Error" back 
from 64.7.177.103


Now what I noticed is that once the transfer is done, I'm still 
connected the the person that called me to do an attended transfer. 
However, if I hang up the phone, the call drops. If I place the call 
on hold and take them off hold, audio is resumed and everything works 
normally.


Here is the conf information

exten => s,1,SetCallerID(${ARG1})
exten => s,n,Set(DST_EXT_NUM=${ARG2})
exten => s,n,gotoif,$[${ARG2}=989XX]?TIME:GOON ;Add test if 
hours is the basis for voice mail


exten => s,n(GOON),AGI(VoiceMail.php)   ;Test for phone status
exten => s,n,Noop(Testing Completed||Code:${ON_PHONE}${IN_QUEUE})
exten => s,n,Dial(SIP/${ARG2},25)

...VoiceMail choice

exten => h,1,HangUp()

Where I have VoiceMail choice it takes the variables from the AGI 
script and decides which voice message to play. But the problem is 
happening before that occurs so I don't think it has anything to do 
with the problem.


Any ideas to what could be the cause or how to correct it? SIP version 
or does the new asterisk build have any new features enabled by 
default that the older build would not? Any suggestions or thoughts 
would be greatly helpful.


Kevin
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Re: [Asterisk-Users] call transfer to external phone number

[Keith Geffert]

SIP transfers happen out of band, so the context is the sip phone's
context noted in sip.conf.

For Inbound and outbound (ie Dial application), the context is the entry
point in the dial plan.  If you need features.conf transfers to work in
a specific context you need to set the __TRANSFER_CONTEXT variable
before the Dial application so asterisk knows what context to look for
extensions.

The relevant wiki page:

http://www.voip-info.org/wiki/view/Asterisk+config+features.conf

example:

exten => 101,1,set(__TRANSFER_CONTEXT=vm-internal)
exten => 101,n,Macro(superdial,SIP/vm-ext1&SIP/outsip/9995551212,15,tr,
,pstn,2,${CALLERIDNAME},${CALLERIDNUM},pstn,[EMAIL PROTECTED])

So .. in this instance, when we outdial the cellphone (9995551212) with
the 't' option, we support transfers.  If we don't set the transfer
context as above when the # key is hit.  Asterisk is looking in the
[inbound] context because that is where extension 101 was dialed from.
But ... exten 101 doesn't want those available extensions, they want the
same set of extensions they have at their sip phone so they can transfer
to voicemail and so on.

Since our outbound pattern dials to SIP/outsip also exist in
[vm-internal] .. calls can be transferred out to PSTN numbers.


in any case.. this is how I got it to work. :)



Cosmin Prund wrote:
>> From: [EMAIL PROTECTED] [mailto:asterisk-users-
>> [EMAIL PROTECTED] On Behalf Of C F
>> Sent: Monday, April 03, 2006 3:49 PM
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: Re: [Asterisk-Users] call transfer to external phone number
>>
>> Yes, as long as the context that the phone transfering has an exten
>> declared for that number.
>>
> 
> Does Asterisk make any distinction between an "internal" number and an
> "external" number? I'm inclined to think it might be some kind of "timeout"
> issue. And I've got the proof:
> 
>>From my Aastra 9xxx (don't know the number and I'm to lasy to go look) I can
> transfer a call to any "extension", including the lng extension required
> for dialing an external number (ie: #0X). Unfortunatelly that's the
> ONLY phone I can do that from! I can't do it from XLite softphone and I
> can't do it from analog phones connected to a Linksys PAP2.
> 
> For the phones that are unable to transfer to external numbers I've got
> "alias" extensions defined (basic, 3 digit extensions).
> 
>> On 4/3/06, Giuseppe <[EMAIL PROTECTED]> wrote:
>>> Hi!
>>> Is it possible to transfer a call to an external phone instead of
>>> transferring the call to internal phone?
>>> (I'm sorry for my bad english, I hope you understand)
>>> When, during a call, I digit "#123", the call is transferred to internal
>>> extension "123",
>>> but if I digit "#external_phone_number", it tells me that it's
>> impossible.
>>> Any idea?
>>>
>>> Thanks a lot!
>>>
>>> Giuseppe
>>>
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Re: [Asterisk-Users] call transfer to external phone number

[C F]

Asteirsk has got no clue what's internal and what's not, it's the
context that decide what numbers are available for a user.
In your case more info is needed to troubleshoot it.

On 4/3/06, Cosmin Prund <[EMAIL PROTECTED]> wrote:
> > From: [EMAIL PROTECTED] [mailto:asterisk-users-
> > [EMAIL PROTECTED] On Behalf Of C F
> > Sent: Monday, April 03, 2006 3:49 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [Asterisk-Users] call transfer to external phone number
> >
> > Yes, as long as the context that the phone transfering has an exten
> > declared for that number.
> >
>
> Does Asterisk make any distinction between an "internal" number and an
> "external" number? I'm inclined to think it might be some kind of "timeout"
> issue. And I've got the proof:
>
> >From my Aastra 9xxx (don't know the number and I'm to lasy to go look) I can
> transfer a call to any "extension", including the lng extension required
> for dialing an external number (ie: #0X). Unfortunatelly that's the
> ONLY phone I can do that from! I can't do it from XLite softphone and I
> can't do it from analog phones connected to a Linksys PAP2.
>
> For the phones that are unable to transfer to external numbers I've got
> "alias" extensions defined (basic, 3 digit extensions).
>
> > On 4/3/06, Giuseppe <[EMAIL PROTECTED]> wrote:
> > > Hi!
> > > Is it possible to transfer a call to an external phone instead of
> > > transferring the call to internal phone?
> > > (I'm sorry for my bad english, I hope you understand)
> > > When, during a call, I digit "#123", the call is transferred to internal
> > > extension "123",
> > > but if I digit "#external_phone_number", it tells me that it's
> > impossible.
> > > Any idea?
> > >
> > > Thanks a lot!
> > >
> > > Giuseppe
> > >
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RE: [Asterisk-Users] call transfer to external phone number

[Cosmin Prund]

> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of C F
> Sent: Monday, April 03, 2006 3:49 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] call transfer to external phone number
> 
> Yes, as long as the context that the phone transfering has an exten
> declared for that number.
> 

Does Asterisk make any distinction between an "internal" number and an
"external" number? I'm inclined to think it might be some kind of "timeout"
issue. And I've got the proof:

>From my Aastra 9xxx (don't know the number and I'm to lasy to go look) I can
transfer a call to any "extension", including the lng extension required
for dialing an external number (ie: #0X). Unfortunatelly that's the
ONLY phone I can do that from! I can't do it from XLite softphone and I
can't do it from analog phones connected to a Linksys PAP2.

For the phones that are unable to transfer to external numbers I've got
"alias" extensions defined (basic, 3 digit extensions).

> On 4/3/06, Giuseppe <[EMAIL PROTECTED]> wrote:
> > Hi!
> > Is it possible to transfer a call to an external phone instead of
> > transferring the call to internal phone?
> > (I'm sorry for my bad english, I hope you understand)
> > When, during a call, I digit "#123", the call is transferred to internal
> > extension "123",
> > but if I digit "#external_phone_number", it tells me that it's
> impossible.
> > Any idea?
> >
> > Thanks a lot!
> >
> > Giuseppe
> >
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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 transfer to external phone number

[C F]

Yes, as long as the context that the phone transfering has an exten
declared for that number.

On 4/3/06, Giuseppe <[EMAIL PROTECTED]> wrote:
> Hi!
> Is it possible to transfer a call to an external phone instead of
> transferring the call to internal phone?
> (I'm sorry for my bad english, I hope you understand)
> When, during a call, I digit "#123", the call is transferred to internal
> extension "123",
> but if I digit "#external_phone_number", it tells me that it's impossible.
> Any idea?
>
> Thanks a lot!
>
> Giuseppe
>
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Re: [Asterisk-Users] call transfer to external phone number

[Dovid Bender]

 Hi!
> Is it possible to transfer a call to an external
> phone instead of
> transferring the call to internal phone?
> (I'm sorry for my bad english, I hope you
> understand)
> When, during a call, I digit "#123", the call is
> transferred to internal 
> extension "123",
> but if I digit "#external_phone_number", it tells me
> that it's impossible.
> Any idea?
> 
> Thanks a lot!
> 
> Giuseppe

I know that with polycom I was able to do this. Not by
using the # sign but by hitting the transfer button
and then entering the persons number and pressing
transfer. Please post your extensions.conf


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Re: [Asterisk-Users] Call Transfer

[Mojo with Horan & Company, LLC]

I would use asterisk's built in blind or attended transfer features. 
This way the system is based around dtmf and the users aren't tied to a 
specific model of phone to accomodate future upgrades.


In order to do this I would recommend editing features.conf so blindxfer 
= ** instead of *.  A single * for transfer makes it real difficult to 
use banking and other IVRs that ask you to press #.


Then, in each necessary Dial cmd of your dialplan, make sure there is a 
t or a T in the options to enable either the called or calling users to 
initiate the transfer.


More details about the Dial command: 
http://www.voip-info.org/wiki-Asterisk+cmd+Dial




Moj

Dave Morrow wrote:
Can anyone point me in the right direction.  My users (all using Sipura 
SPA-841 phones) need the ability to transfer a call to another number.  
How can I setup a dial plan to do this?



David A. Morrow
Technical Systems Lead
Autodata Solutions Company
[EMAIL PROTECTED] 
_http://www.autodatasolutions.com_

NEW !!! Tel: (519) 963-3020
Fax: (519) 451-6615 


< Lead, follow or get out of the way! >

This message has originated from Autodata Solutions. The attached 
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--
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Office Manger, Horan & Company, LLC
(907) 747- x112
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Re: [Asterisk-Users] call transfer

[Michael Sampson]

I got this to work by editing the line
exten => s,14,Dial(${OUT_${ARG1}}/${OUTNUM})
to say
exten => s,14,Dial(${OUT_${ARG1}}/${OUTNUM},,Tt)
in extensions.conf

Do you know of anyway to set it up through AMP, so it works with all calls?

Michael Sampson
Information Systems Manager
Customer Contact Services
[EMAIL PROTECTED]
952-936-4000



Michael Sampson wrote:

I'm not sure how this is suppose to work. But I want to be able to 
call people from a SIP phone and transfer them into a conference room. 
If I call another extension that is a SIP phone I can hit # and then 
enter the conference room number. If I call from the PSTN to the SIP 
extension phone I can transfer by hitting # too. But if I call from 
the SIP phone extension to a PSTN number it doesn't do anything when I 
hit the #. I'm using [EMAIL PROTECTED] and under general settings I have 
"tTrwW" for Asterisk Dial Command Settings.

Can you call through a Zap trunk from a SIP phone and do a call transfer?


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Re: [Asterisk-Users] Call transfer with voicemail password

[Giovanni Miano]

http://www.voip-info.org/wiki/view/Asterisk+authenticate+using+voicemail+passwords

Cheers

2005/12/1, Joe Pukepail <[EMAIL PROTECTED]>:
> Look into the findme feature, there is a patch on the bug tracker to add
> this feature.  I believe that someone shows how to do it in the dial plan.
> I plan on implementing this, but haven't gotten around to it yet.
>
>
>
> On 11/30/05, Benjamin Lenard <[EMAIL PROTECTED]> wrote:
> > Hi,
> >
> > I'm trying to have an extension ring my SIP phone then try my cell
> > phone.  I can transfer the call fine to the cell but I want it to ask
> > for a pin , voicemail pin, before transferring the call.
> > This is so if my cell's voicemail answers , the call doesn't transfer
> > to it.
> >
> > Any ideas?
> >
> > Thanks,
> > Ben
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>


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Re: [Asterisk-Users] Call transfer with voicemail password

[Joe Pukepail]

Look into the findme feature, there is a patch on the bug tracker to add this feature.  I believe that someone shows how to do it in the dial plan.   I plan on implementing this, but haven't gotten around to it yet. 

On 11/30/05, Benjamin Lenard <[EMAIL PROTECTED]> wrote:
Hi,I'm trying to have an extension ring my SIP phone then try my cellphone.  I can transfer the call fine to the cell but I want it to ask
for a pin , voicemail pin, before transferring the call.This is so if my cell's voicemail answers , the call doesn't transferto it.Any ideas?Thanks,Ben___
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Re: [Asterisk-Users] call transfer and pick chan_h323

[Lenz]

AFAIK there were some known issues preventing call transfer from H323  
terminals, at least with Innovaphone ones.

Yours
l.



On Fri, 18 Nov 2005 17:45:39 +0100, Santosh Rao  
<[EMAIL PROTECTED]> wrote:



Hello list,
   We have asterisk v1.2.0 CVS head and ooh323 in place. calls  
can be made and recieved to and from extensions.
How to implement call transfer and call pickup. when using asterisk  
1.0.x dtmf=inband registers and sends dtmf but with asterisk 1.2 and  
ooh323 it does not.. is this a known issue ? While google heard tht  
there was a issue with chan_h323.so would not send inband so tried to  
install chan_0h323.so but but.. asterisk refuses to start with  
chan_oh323 it says  "Unregistered channel type 'Modem'"
my basic requirements are h323 , call pickup and call transfer? below  
attached are the configurations files tht we are using currently ...


thanking for all your support ..



Extensions.conf:-
[testing]
exten => _7.,1,Pickup({66}:[EMAIL PROTECTED])
exten => 666,1,Dial(H323/192.168.1.194,100,Ttr)
exten => 667,1,Dial(H323/192.168.1.195,100,Ttr)
exten => 668,1,Dial(H323/192.168.1.196,100,Ttr)
exten => 669,1,Dial(H323/192.168.1.192,100,Ttr)

H323.conf:-
[general]
port = 1720
bindaddr = 0.0.0.0 ; this SHALL contain a single, valid IP address for  
this machine

disallow=all
allow=ulaw
allow=alaw
;dtmfmode=auto
dtmfmode=inband

gatekeeper = DISABLE
context=testing

[vivek]
type=friend
host=192.168.1.194
context=testing
Callgroup=1
pickupgroup=1-9,13

[santosh]
type=friend
host=192.168.1.195
context=testing
Callgroup=1
pickupgroup=1-9,13

[binu]
type=friend
host=192.168.1.196
context=testing
Callgroup=1
pickupgroup=1-9,13

[test1]
type=friend
host=192.168.1.192
context=testing
Callgroup=1
pickupgroup=1-9,13

Features.conf:-

[general]
parkext => 700 ; What ext. to dial to park
parkpos => 701-720 ; What extensions to park calls on
context => parkedcalls ; Which context parked calls are in
pickupex = *8

[featuremap]
blindxfer => #1 ; Blind transfer
atxfer => *2 ; Attended transfer





  "I haven't lost my mind; it's backed up on
   tape somewhere."

Santosh Rao
Trikon Electronics Pvt Ltd






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Re: [Asterisk-Users] Call Transfer problems-am I missing something?

[alex]

Hi,

Thanks for the clarification.  I had seen that the two options 
existed, but the docs for the dial() command didn't state the 
difference.
On Sun, Oct 30, 2005 at 08:23:32PM -0500, David Bandel wrote:
> On 10/30/05, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
> > Hi All,
> >
> > Recently got call-transfer somewhat working on my asterisk-1.0.9
> > install, and came across an interesting problem.  I have an account on a
> > VOIP Provider (voipbuster using iax to be exact) and use a line like
> > this in extensions.conf to have it handle all outgoing calls beginning
> > with 1:
> > exten => _1NN,1,Dial(voipbuster/00${EXTEN},t)
> > When I call someone and press # on the phone ( I've tried this with
> > various softphones and a regular phone connected to a linksys pap2)
> > Nothing happens.However, if the called party presses # they get the
> > extension prompt, and can then transfer me to an other extension.  Does
> > anyone know why the calling party can't initiate the transfer? am I
> > missing something?
> 
> Yes.  The ,t  in the Dial() options is for callee, the T is for
> caller.  ,tT is for both.
> 
> Ciao,
> 
> David A. Bandel
> --
> Focus on the dream, not the competition.
> - Nemesis Air Racing Team motto
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Re: [Asterisk-Users] Call Transfer problems-am I missing something?

[Eric \"ManxPower\" Wieling]

David Bandel wrote:

On 10/30/05, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:


Hi All,

Recently got call-transfer somewhat working on my asterisk-1.0.9
install, and came across an interesting problem.  I have an account on a
VOIP Provider (voipbuster using iax to be exact) and use a line like
this in extensions.conf to have it handle all outgoing calls beginning
with 1:
exten => _1NN,1,Dial(voipbuster/00${EXTEN},t)
When I call someone and press # on the phone ( I've tried this with
various softphones and a regular phone connected to a linksys pap2)
Nothing happens.However, if the called party presses # they get the
extension prompt, and can then transfer me to an other extension.  Does
anyone know why the calling party can't initiate the transfer? am I
missing something?



Yes.  The ,t  in the Dial() options is for callee, the T is for
caller.  ,tT is for both.


As is documented in "show application dial" in the Asterisk CLI.
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Re: [Asterisk-Users] Call Transfer problems-am I missing something?

[David Bandel]

On 10/30/05, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
> Hi All,
>
> Recently got call-transfer somewhat working on my asterisk-1.0.9
> install, and came across an interesting problem.  I have an account on a
> VOIP Provider (voipbuster using iax to be exact) and use a line like
> this in extensions.conf to have it handle all outgoing calls beginning
> with 1:
> exten => _1NN,1,Dial(voipbuster/00${EXTEN},t)
> When I call someone and press # on the phone ( I've tried this with
> various softphones and a regular phone connected to a linksys pap2)
> Nothing happens.However, if the called party presses # they get the
> extension prompt, and can then transfer me to an other extension.  Does
> anyone know why the calling party can't initiate the transfer? am I
> missing something?

Yes.  The ,t  in the Dial() options is for callee, the T is for
caller.  ,tT is for both.

Ciao,

David A. Bandel
--
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Re: [Asterisk-Users] Call Transfer

[Rhonda Herron]

He is what happens from the time the extension is selected from the time 
the digital receptionist answers until  I hangup. I watched the logs as 
I was pushing all sorts of transfer button possibilities and nothing. It 
just stayed at 'ooh, voice format changed to 4' Which, while humorous 
tells me nothing except that my phone is not able to communicate with 
the sever at all from the time the call is put through until the call is 
done.


Oct 20 15:33:45 VERBOSE[2909]: -- Executing 
Dial("IAX2/[EMAIL PROTECTED]/4", "IAX2/7878|15|tr") in new stack

Oct 20 15:33:45 DEBUG[2909]: SIMPLE DIAL (NO URL)
Oct 20 15:33:45 VERBOSE[2909]: -- Called 7878
Oct 20 15:33:45 VERBOSE[2909]: -- Call accepted by xxx.xxx.xxx.xxx 
(format ulaw)

Oct 20 15:33:45 VERBOSE[2909]: -- Format for call is ulaw
Oct 20 15:33:45 VERBOSE[2909]: -- IAX2/7878/8 is ringing
Oct 20 15:33:50 VERBOSE[2909]: -- IAX2/7878/8 answered 
IAX2/[EMAIL PROTECTED]/4
Oct 20 15:33:50 VERBOSE[2909]: -- Attempting native bridge of 
IAX2/[EMAIL PROTECTED]/4 and IAX2/7878/8

Oct 20 15:33:50 DEBUG[2909]: Ooh, voice format changed to 4

Here is the extension  config for 7878:
exten => 7878,1,Macro(exten-vm,[EMAIL PROTECTED],7878)

And this is the config for aah_1 ( our digital receptionist)
[aa_1]
include => aa_1-custom
exten => 1,1,Goto(,s,1);
exten => fax,1,Goto(ext-fax,in_fax,1);
exten => h,1,Hangup();
exten => i,1,Playback(invalid);
exten => i,2,Goto(s,7);
include => ext-local
include => app-messagecenter
include => app-directory
exten => s,1,GotoIf($[${DIALSTATUS} = ANSWER]?4);
exten => s,2,Answer();
exten => s,3,Wait(1);
exten => s,4,SetVar(DIR-CONTEXT=default);
exten => s,5,DigitTimeout(3); Basic
exten => s,6,ResponseTimeout(7);
exten => s,7,Background(custom/aa_1);

Thanks, once again,
-R


Rhonda Herron wrote:

Yes, I can dial *97 for VM and check messages. When I select # during  
a call it does nothing though. I tried inband for DTMF but that didnt 
work. Am going to run debug mode ( first I have to figure out how :) ) 
and I will let you know what I find out.


Thanks so far,
R

Tom Vile wrote:

Blind transfer should work fine #.  Can you dial into Voicemail and 
enter your password succesfully?


On 10/20/05, *BJ Weschke* < [EMAIL PROTECTED] 
> wrote:


 I'm not sure the txfer functionality is in the 1.0.X branch. I'm
pretty sure you will need HEAD or the 1.2 betas.

On 10/20/05, *Rhonda Herron* < [EMAIL PROTECTED]
> wrote:

It is set to rfc2833.

Tom Vile wrote:


maybe its not setting the DTMF tones properly.  What do you


have setup


for the phone and extensions.  Usually its rfc2833 but could


be inband.



On 10/20/05, *Rhonda Herron* < [EMAIL PROTECTED]





>> wrote:


I have the phone specific directions to transfer calls,


but I


tried your
suggestion. No go. I have 3 of the Eezee phones and  call


transfer


doesn't  work on any of them, so I really don't think it


is hardware


related. I think the problem may be with my feature.conf


which had no


reference to blindxfer or atxfer. I added them so my


feature.conf now


looks like this:

transferdigittimeout => 3  ; Number of seconds to


wait between


digits when transfering a call
;courtesytone = beep; Sound file to play to


the parked


caller
 ; when someone dials a


parked call


xfersound = beep   ; to indicate an attended


transfer is


complete
xferfailsound = beeperr; to indicate a failed


transfer


;adsipark = yes ; if you want ADSI parking
announcements
;pickupexten = *8   ; Configure the pickup


extension.


Default is *8
;featuredigittimeout = 500  ; Max time (ms) between


digits for


 ; feature


activation.  Default is 500



[featuremap]
blindxfer => #; Blind transfer
disconnect => *0   ; Disconnect
;automon => *1  ; One Touch Record
atxfer => *2


I rebooted my * server but still no go. Are


there  dependencies  I am


not aware of? Should [featuremap] be referenced elsewhere


as well?


I am
working with * CVS 1.0.9 and have found an article on


wiki that


support
for call transfer was added in 1.2.  Are there other


places I need to


hack for this functionality?

Thanks,
-R

Tom Vile wrote:

> try # and then dial the extension.
>
> On 10/20/05, *Rhonda Herron* < [EMAIL PROTECTED]

Re: [Asterisk-Users] Call Transfer

[Rhonda Herron]

Yes, I can dial *97 for VM and check messages. When I select # during  a 
call it does nothing though. I tried inband for DTMF but that didnt 
work. Am going to run debug mode ( first I have to figure out how :) ) 
and I will let you know what I find out.


Thanks so far,
R

Tom Vile wrote:

Blind transfer should work fine #.  Can you dial into Voicemail and 
enter your password succesfully?


On 10/20/05, *BJ Weschke* < [EMAIL PROTECTED] 
> wrote:


 I'm not sure the txfer functionality is in the 1.0.X branch. I'm
pretty sure you will need HEAD or the 1.2 betas.

On 10/20/05, *Rhonda Herron* < [EMAIL PROTECTED]
> wrote:

It is set to rfc2833.

Tom Vile wrote:


maybe its not setting the DTMF tones properly.  What do you

have setup

for the phone and extensions.  Usually its rfc2833 but could

be inband.


On 10/20/05, *Rhonda Herron* < [EMAIL PROTECTED]



>> wrote:


I have the phone specific directions to transfer calls,

but I

tried your
suggestion. No go. I have 3 of the Eezee phones and  call

transfer

doesn't  work on any of them, so I really don't think it

is hardware

related. I think the problem may be with my feature.conf

which had no

reference to blindxfer or atxfer. I added them so my

feature.conf now

looks like this:

transferdigittimeout => 3  ; Number of seconds to

wait between

digits when transfering a call
;courtesytone = beep; Sound file to play to

the parked

caller
 ; when someone dials a

parked call

xfersound = beep   ; to indicate an attended

transfer is

complete
xferfailsound = beeperr; to indicate a failed

transfer

;adsipark = yes ; if you want ADSI parking
announcements
;pickupexten = *8   ; Configure the pickup

extension.

Default is *8
;featuredigittimeout = 500  ; Max time (ms) between

digits for

 ; feature

activation.  Default is 500


[featuremap]
blindxfer => #; Blind transfer
disconnect => *0   ; Disconnect
;automon => *1  ; One Touch Record
atxfer => *2


I rebooted my * server but still no go. Are

there  dependencies  I am

not aware of? Should [featuremap] be referenced elsewhere

as well?

I am
working with * CVS 1.0.9 and have found an article on

wiki that

support
for call transfer was added in 1.2.  Are there other

places I need to

hack for this functionality?

Thanks,
-R

Tom Vile wrote:

> try # and then dial the extension.
>
> On 10/20/05, *Rhonda Herron* < [EMAIL PROTECTED]



>
> 
 mailto:[EMAIL PROTECTED] wrote:

>
> Hello,
>
> I have my [EMAIL PROTECTED] working beautifully for basic call
function. So now I
> am testing extended functions for my office users

and am

hitting a
> wall.
> I simply need to be able to put a call on hold and

forward

it to any
> another internal extension. I have an Eezee AT-320

IAX2

phone and
> according to the directions, I  simply select Hold

> enter ext>

> hit Fwd.
> However when I press the button all I do is annoy

the caller

with
> loud
> button punching sounds. Does something need to be

configured

in * to
> allow call transfer to work? I am using an inbound

trunk from

> Teliax- no
> cards, just a T1 direct to my * server.  I have

found transfer

> functions
> for zapatel- but as I said I am just using the T1

and have

no zapatel
> trunks/configurations.  I have also seen a lot of
information for call
> forwarding but that sets up a permanent forward

function to a

> specific
> extension. I just want to be able to say "One

moment, Mike can

> help you
> with that, let me transfer you" and then be able to

do it.

Since this
> happens with all my AT-320 phones, I don't think it

is hardware

> related
> and there is no mention of call transfer

configuration for

the phone
> itself.
>
> Thanks
>
> -R
> ___
> --Bandwidth and Colocation sponsored by

Easynews.com 

Re: [Asterisk-Users] Call Transfer

[Tom Vile]

Blind transfer should work fine #.  Can you dial into Voicemail and enter your password succesfully?On 10/20/05, BJ Weschke <
[EMAIL PROTECTED]> wrote: I'm not sure the txfer functionality is in the 1.0.X
 branch. I'm pretty sure you will need HEAD or the 1.2 betas.On 10/20/05, Rhonda Herron <

[EMAIL PROTECTED]> wrote:
It is set to rfc2833.Tom Vile wrote:
> maybe its not setting the DTMF tones properly.  What do you have setup> for the phone and extensions.  Usually its rfc2833 but could be inband.>> On 10/20/05, *Rhonda Herron* <

[EMAIL PROTECTED]> [EMAIL PROTECTED]>> wrote:>>> I have the phone specific directions to transfer calls, but I
> tried your
> suggestion. No go. I have 3 of the Eezee phones and  call transfer> doesn't  work on any of them, so I really don't think it is hardware> related. I think the problem may be with my feature.conf

 which had no> reference to blindxfer or atxfer. I added them so my feature.conf now> looks like this:>> transferdigittimeout => 3  ; Number of seconds to wait between> digits when transfering a call
> ;courtesytone =
beep;
Sound file to play to the parked> caller>  ; when someone dials a parked call>
xfersound =
beep  
; to indicate an attended transfer is
> complete>
xferfailsound =
beeperr; to indicate a
failed transfer> ;adsipark =
yes
; if you want ADSI parking> announcements>
;pickupexten =
*8  
; Configure the pickup extension.
> Default is *8> ;featuredigittimeout = 500  ; Max time (ms) between digits for>  ; feature activation.  Default is 500>> [featuremap]
> blindxfer =>
#;
Blind transfer> disconnect =>
*0  
; Disconnect> ;automon =>
*1  ;
One Touch Record> atxfer => *2>
>> I rebooted my * server but still no go. Are there  dependencies  I am> not aware of? Should [featuremap] be referenced elsewhere as well?> I am> working with * CVS 1.0.9

 and have found an article on wiki that> support> for call transfer was added in 1.2.  Are there other places I need to> hack for this functionality?>> Thanks,> -R
>> Tom Vile wrote:>> > try # and then dial the extension.> >> > On 10/20/05, *Rhonda Herron* <
[EMAIL PROTECTED]
> [EMAIL PROTECTED]>> > 
[EMAIL PROTECTED] 
[EMAIL PROTECTED]>>> wrote:> >> > Hello,> >> > I have my [EMAIL PROTECTED] working beautifully for basic call> function. So now I> > am testing extended functions for my office users and am
> hitting a> > wall.> > I simply need to be able to put a call on hold and forward> it to any> > another internal extension. I have an Eezee AT-320 IAX2
> phone and>
> according to the directions,
I  simply select Hold > enter ext>> > hit Fwd.> > However when I press the button all I do is annoy the caller
> with> > loud> > button punching sounds. Does something need to be configured> in * to> > allow call transfer to work? I am using an inbound trunk from
> > Teliax- no>
> cards, just a T1 direct to my *
server.  I have found transfer> > functions> > for zapatel- but as I said I am just using the T1 and have
> no zapatel> > trunks/configurations.  I have also seen a lot of> information for call> > forwarding but that sets up a permanent forward function to a> > specific
> > extension. I just want to be able to say "One moment, Mike can> > help you> > with that, let me transfer you" and then be able to do it.> Since this
> > happens with all my AT-320 phones, I don't think it is hardware> > related> > and there is no mention of call transfer configuration for> the phone
> > itself.
> >> > Thanks> >> > -R> > ___> > --Bandwidth and Colocation sponsored by 

Easynews.com> > > <
http://Easynews.com> --> >> > Asterisk-Users mailing list
> > Asterisk-Users@lists.digium.com> 
Asterisk-Users@lists.digium.com>
> > Asterisk-Users@lists.digium.com> 
Asterisk-Users@lists.digium.com
>>> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > <
http://lists.digium.com/mailman/listinfo/asterisk-users> >> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >> >> >> >
> > --> > Tom Vile> > Baldwin Technology Solutions, Inc> > Consulting - Web Design - VoIP Telephony> > 
www.baldwintechsolutions.com
> 

Re: [Asterisk-Users] Call Transfer

[BJ Weschke]

 I'm not sure the txfer functionality is in the 1.0.X branch. I'm pretty sure you will need HEAD or the 1.2 betas.On 10/20/05, Rhonda Herron <
[EMAIL PROTECTED]> wrote:It is set to rfc2833.Tom Vile wrote:
> maybe its not setting the DTMF tones properly.  What do you have setup> for the phone and extensions.  Usually its rfc2833 but could be inband.>> On 10/20/05, *Rhonda Herron* <
[EMAIL PROTECTED]> [EMAIL PROTECTED]>> wrote:>>> I have the phone specific directions to transfer calls, but I> tried your
> suggestion. No go. I have 3 of the Eezee phones and  call transfer> doesn't  work on any of them, so I really don't think it is hardware> related. I think the problem may be with my feature.conf
 which had no> reference to blindxfer or atxfer. I added them so my feature.conf now> looks like this:>> transferdigittimeout => 3  ; Number of seconds to wait between> digits when transfering a call
> ;courtesytone = beep; Sound file to play to the parked> caller>  ; when someone dials a parked call> xfersound = beep   ; to indicate an attended transfer is
> complete> xferfailsound = beeperr; to indicate a failed transfer> ;adsipark = yes ; if you want ADSI parking> announcements> ;pickupexten = *8   ; Configure the pickup extension.
> Default is *8> ;featuredigittimeout = 500  ; Max time (ms) between digits for>  ; feature activation.  Default is 500>> [featuremap]
> blindxfer => #; Blind transfer> disconnect => *0   ; Disconnect> ;automon => *1  ; One Touch Record> atxfer => *2>
>> I rebooted my * server but still no go. Are there  dependencies  I am> not aware of? Should [featuremap] be referenced elsewhere as well?> I am> working with * CVS 1.0.9
 and have found an article on wiki that> support> for call transfer was added in 1.2.  Are there other places I need to> hack for this functionality?>> Thanks,> -R
>> Tom Vile wrote:>> > try # and then dial the extension.> >> > On 10/20/05, *Rhonda Herron* <[EMAIL PROTECTED]
> [EMAIL PROTECTED]>> > [EMAIL PROTECTED] 
[EMAIL PROTECTED]>>> wrote:> >> > Hello,> >> > I have my [EMAIL PROTECTED] working beautifully for basic call> function. So now I> > am testing extended functions for my office users and am
> hitting a> > wall.> > I simply need to be able to put a call on hold and forward> it to any> > another internal extension. I have an Eezee AT-320 IAX2
> phone and> > according to the directions, I  simply select Hold > enter ext>> > hit Fwd.> > However when I press the button all I do is annoy the caller
> with> > loud> > button punching sounds. Does something need to be configured> in * to> > allow call transfer to work? I am using an inbound trunk from
> > Teliax- no> > cards, just a T1 direct to my * server.  I have found transfer> > functions> > for zapatel- but as I said I am just using the T1 and have
> no zapatel> > trunks/configurations.  I have also seen a lot of> information for call> > forwarding but that sets up a permanent forward function to a> > specific
> > extension. I just want to be able to say "One moment, Mike can> > help you> > with that, let me transfer you" and then be able to do it.> Since this
> > happens with all my AT-320 phones, I don't think it is hardware> > related> > and there is no mention of call transfer configuration for> the phone> > itself.
> >> > Thanks> >> > -R> > ___> > --Bandwidth and Colocation sponsored by 
Easynews.com> > >  --> >> > Asterisk-Users mailing list
> > Asterisk-Users@lists.digium.com> Asterisk-Users@lists.digium.com>
> > Asterisk-Users@lists.digium.com> Asterisk-Users@lists.digium.com
>>> > http://lists.digium.com/mailman/listinfo/asterisk-users> > <
http://lists.digium.com/mailman/listinfo/asterisk-users> >> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users> >> >> >> >
> > --> > Tom Vile> > Baldwin Technology Solutions, Inc> > Consulting - Web Design - VoIP Telephony> > www.baldwintechsolutions.com
>  <> http://www.baldwintechsolutions.com>
> > Phone: 518-631-2855 x205> > Phone: 845-652-4578 x205> > Phone: 978-203-384

Re: [Asterisk-Users] Call Transfer

[Rhonda Herron]

It is set to rfc2833.

Tom Vile wrote:

maybe its not setting the DTMF tones properly.  What do you have setup 
for the phone and extensions.  Usually its rfc2833 but could be inband.


On 10/20/05, *Rhonda Herron* <[EMAIL PROTECTED] 
> wrote:



I have the phone specific directions to transfer calls, but I
tried your
suggestion. No go. I have 3 of the Eezee phones and  call transfer
doesn't  work on any of them, so I really don't think it is hardware
related. I think the problem may be with my feature.conf which had no
reference to blindxfer or atxfer. I added them so my feature.conf now
looks like this:

transferdigittimeout => 3  ; Number of seconds to wait between
digits when transfering a call
;courtesytone = beep; Sound file to play to the parked
caller
 ; when someone dials a parked call
xfersound = beep   ; to indicate an attended transfer is
complete
xferfailsound = beeperr; to indicate a failed transfer
;adsipark = yes ; if you want ADSI parking
announcements
;pickupexten = *8   ; Configure the pickup extension.
Default is *8
;featuredigittimeout = 500  ; Max time (ms) between digits for
 ; feature activation.  Default is 500

[featuremap]
blindxfer => #; Blind transfer
disconnect => *0   ; Disconnect
;automon => *1  ; One Touch Record
atxfer => *2


I rebooted my * server but still no go. Are there  dependencies  I am
not aware of? Should [featuremap] be referenced elsewhere as well?
I am
working with * CVS 1.0.9 and have found an article on wiki that
support
for call transfer was added in 1.2.  Are there other places I need to
hack for this functionality?

Thanks,
-R

Tom Vile wrote:

> try # and then dial the extension.
>
> On 10/20/05, *Rhonda Herron* <[EMAIL PROTECTED]

> >> wrote:
>
> Hello,
>
> I have my [EMAIL PROTECTED] working beautifully for basic call
function. So now I
> am testing extended functions for my office users and am
hitting a
> wall.
> I simply need to be able to put a call on hold and forward
it to any
> another internal extension. I have an Eezee AT-320 IAX2
phone and
> according to the directions, I  simply select Hold > enter ext>
> hit Fwd.
> However when I press the button all I do is annoy the caller
with
> loud
> button punching sounds. Does something need to be configured
in * to
> allow call transfer to work? I am using an inbound trunk from
> Teliax- no
> cards, just a T1 direct to my * server.  I have found transfer
> functions
> for zapatel- but as I said I am just using the T1 and have
no zapatel
> trunks/configurations.  I have also seen a lot of
information for call
> forwarding but that sets up a permanent forward function to a
> specific
> extension. I just want to be able to say "One moment, Mike can
> help you
> with that, let me transfer you" and then be able to do it.
Since this
> happens with all my AT-320 phones, I don't think it is hardware
> related
> and there is no mention of call transfer configuration for
the phone
> itself.
>
> Thanks
>
> -R
> ___
> --Bandwidth and Colocation sponsored by Easynews.com

>  --
>
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com

> >
> http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
>
> --
> Tom Vile
> Baldwin Technology Solutions, Inc
> Consulting - Web Design - VoIP Telephony
> www.baldwintechsolutions.com
 <
http://www.baldwintechsolutions.com>
> Phone: 518-631-2855 x205
> Phone: 845-652-4578 x205
> Phone: 978-203-3848 x205
> Fax: 518-631-2856
>
>
>
>___
>--Bandwidth and Colocation sponsored by Easynews.com

Re: [Asterisk-Users] Call Transfer

[Tom Vile]

maybe its not setting the DTMF tones properly.  What do you have
setup for the phone and extensions.  Usually its rfc2833 but could
be inband.On 10/20/05, Rhonda Herron <[EMAIL PROTECTED]> wrote:
I have the phone specific directions to transfer calls, but I tried yoursuggestion. No go. I have 3 of the Eezee phones and  call transferdoesn't  work on any of them, so I really don't think it is hardware
related. I think the problem may be with my feature.conf which had noreference to blindxfer or atxfer. I added them so my feature.conf nowlooks like this:transferdigittimeout => 3  ; Number of seconds to wait between
digits when transfering a call
;courtesytone =
beep;
Sound file to play to the parked caller
; when someone dials a parked call xfersound =
beep  
; to indicate an attended transfer iscomplete xferfailsound = beeperr; to indicate a failed transfer
;adsipark =
yes
; if you want ADSI parking announcements ;pickupexten =
*8  
; Configure the pickup extension.Default is *8 ;featuredigittimeout = 500  ; Max time (ms) between digits for
; feature activation.  Default is 500 [featuremap]
blindxfer =>
#;
Blind transfer disconnect => *0   ; Disconnect
;automon =>
*1  ;
One Touch Record atxfer => *2I rebooted my * server but still no go. Are there  dependencies  I amnot aware of? Should [featuremap] be referenced elsewhere as well? I amworking with * CVS 1.0.9
 and have found an article on wiki that supportfor call transfer was added in 1.2.  Are there other places I need tohack for this functionality?Thanks,-RTom Vile wrote:> try # and then dial the extension.
>> On 10/20/05, *Rhonda Herron* <[EMAIL PROTECTED]> [EMAIL PROTECTED]>> wrote:>> Hello,
>> I have my [EMAIL PROTECTED] working beautifully for basic call function. So now I> am testing extended functions for my office users and am hitting a> wall.> I simply need to be able to put a call on hold and forward it to any
> another internal extension. I have an Eezee AT-320 IAX2 phone and> according to the directions, I  simply select Hold > enter ext>> hit Fwd.> However when I press the button all I do is annoy the caller with
> loud> button punching sounds. Does something need to be configured in * to> allow call transfer to work? I am using an inbound trunk from> Teliax- no> cards, just a T1 direct to my * server.  I have found transfer
> functions> for zapatel- but as I said I am just using the T1 and have no zapatel> trunks/configurations.  I have also seen a lot of information for call> forwarding but that sets up a permanent forward function to a
> specific> extension. I just want to be able to say "One moment, Mike can> help you> with that, let me transfer you" and then be able to do it. Since this> happens with all my AT-320 phones, I don't think it is hardware
> related> and there is no mention of call transfer configuration for the phone> itself.>> Thanks>> -R> ___
> --Bandwidth and Colocation sponsored by Easynews.com>  -->> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com> Asterisk-Users@lists.digium.com>> 
http://lists.digium.com/mailman/listinfo/asterisk-users> > To UNSUBSCRIBE or update options visit:>http://lists.digium.com/mailman/listinfo/asterisk-users>>
>>> --> Tom Vile> Baldwin Technology Solutions, Inc> Consulting - Web Design - VoIP Telephony> www.baldwintechsolutions.com <
http://www.baldwintechsolutions.com>> Phone: 518-631-2855 x205> Phone: 845-652-4578 x205> Phone: 978-203-3848 x205> Fax: 518-631-2856
>>>>___>--Bandwidth and Colocation sponsored by 
Easynews.com -->>Asterisk-Users mailing list>Asterisk-Users@lists.digium.com>
http://lists.digium.com/mailman/listinfo/asterisk-users>To UNSUBSCRIBE or update options visit:>   http://lists.digium.com/mailman/listinfo/asterisk-users
>___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list
Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:   
http://lists.digium.com/mailman/listinfo/asterisk-users-- Tom VileBaldwin Technology Solutions, IncConsulting - Web Design - VoIP Telephony
www.baldwintechsolutions.comPhone: 518-631-2855 x205Phone: 845-652-4578 x205Phone: 978-203-3848 x205Fax: 518-631-2856
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Asterisk-Users@lists.digium.com
h

Re: [Asterisk-Users] Call Transfer

[Rhonda Herron]

I have the phone specific directions to transfer calls, but I tried your 
suggestion. No go. I have 3 of the Eezee phones and  call transfer 
doesn't  work on any of them, so I really don't think it is hardware 
related. I think the problem may be with my feature.conf which had no 
reference to blindxfer or atxfer. I added them so my feature.conf now 
looks like this:


transferdigittimeout => 3  ; Number of seconds to wait between 
digits when transfering a call

;courtesytone = beep; Sound file to play to the parked caller
; when someone dials a parked call
xfersound = beep   ; to indicate an attended transfer is 
complete

xferfailsound = beeperr; to indicate a failed transfer
;adsipark = yes ; if you want ADSI parking announcements
;pickupexten = *8   ; Configure the pickup extension.  
Default is *8

;featuredigittimeout = 500  ; Max time (ms) between digits for
; feature activation.  Default is 500

[featuremap]
blindxfer => #; Blind transfer
disconnect => *0   ; Disconnect
;automon => *1  ; One Touch Record
atxfer => *2


I rebooted my * server but still no go. Are there  dependencies  I am 
not aware of? Should [featuremap] be referenced elsewhere as well? I am 
working with * CVS 1.0.9 and have found an article on wiki that support 
for call transfer was added in 1.2.  Are there other places I need to 
hack for this functionality?


Thanks,
-R

Tom Vile wrote:


try # and then dial the extension.

On 10/20/05, *Rhonda Herron* <[EMAIL PROTECTED] 
> wrote:


Hello,

I have my [EMAIL PROTECTED] working beautifully for basic call function. So 
now I
am testing extended functions for my office users and am hitting a
wall.
I simply need to be able to put a call on hold and forward it to any
another internal extension. I have an Eezee AT-320 IAX2 phone and
according to the directions, I  simply select Hold > enter ext>
hit Fwd.
However when I press the button all I do is annoy the caller with
loud
button punching sounds. Does something need to be configured in * to
allow call transfer to work? I am using an inbound trunk from
Teliax- no
cards, just a T1 direct to my * server.  I have found transfer
functions
for zapatel- but as I said I am just using the T1 and have no zapatel
trunks/configurations.  I have also seen a lot of information for call
forwarding but that sets up a permanent forward function to a
specific
extension. I just want to be able to say "One moment, Mike can
help you
with that, let me transfer you" and then be able to do it. Since this
happens with all my AT-320 phones, I don't think it is hardware
related
and there is no mention of call transfer configuration for the phone
itself.

Thanks

-R
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--
Tom Vile
Baldwin Technology Solutions, Inc
Consulting - Web Design - VoIP Telephony
www.baldwintechsolutions.com 
Phone: 518-631-2855 x205
Phone: 845-652-4578 x205
Phone: 978-203-3848 x205
Fax: 518-631-2856



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Re: [Asterisk-Users] Call Transfer

[Tom Vile]

try # and then dial the extension.On 10/20/05, Rhonda Herron <[EMAIL PROTECTED]> wrote:
Hello,I have my [EMAIL PROTECTED] working beautifully for basic call function. So now Iam testing extended functions for my office users and am hitting a wall.I simply need to be able to put a call on hold and forward it to any
another internal extension. I have an Eezee AT-320 IAX2 phone andaccording to the directions, I  simply select Hold > enter ext> hit Fwd.However when I press the button all I do is annoy the caller with loud
button punching sounds. Does something need to be configured in * toallow call transfer to work? I am using an inbound trunk from Teliax- nocards, just a T1 direct to my * server.  I have found transfer functions
for zapatel- but as I said I am just using the T1 and have no zapateltrunks/configurations.  I have also seen a lot of information for callforwarding but that sets up a permanent forward function to a specific
extension. I just want to be able to say "One moment, Mike can help youwith that, let me transfer you" and then be able to do it. Since thishappens with all my AT-320 phones, I don't think it is hardware related
and there is no mention of call transfer configuration for the phoneitself.Thanks-R___--Bandwidth and Colocation sponsored by 
Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users
-- Tom VileBaldwin Technology Solutions, IncConsulting - Web Design - VoIP Telephonywww.baldwintechsolutions.comPhone: 518-631-2855 x205Phone: 845-652-4578 x205
Phone: 978-203-3848 x205Fax: 518-631-2856
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Re: [Asterisk-Users] Call transfer - atxfer

[Eric \"ManxPower\" Wieling]

Andrew Nowrot wrote:

Hi,

Thank for the Email

I'm using 1.0.9 so probably I'm will not have this feature. In which
version of Asterisk the DTMF Attended Transfer is supported, in 1.2
Beta?


CVS-HEAD and 1.2Beta1 and later.
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Re: [Asterisk-Users] Call transfer - atxfer

[Andrew Nowrot]

Hi,

Thank for the Email

I'm using 1.0.9 so probably I'm will not have this feature. In which
version of Asterisk the DTMF Attended Transfer is supported, in 1.2
Beta?

Best wishes

Andrew
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Re: [Asterisk-Users] Call transfer - atxfer

[Eric \"ManxPower\" Wieling]

Are you using 1.0.x?  DTMF Attended Transfer is not supported in 1.0.x. 
 Unless you have a brain dead phone, you should be able to use SIP 
attended transfer in 1.0.x.  (that would be the transfer key on the phone)


Andrew Nowrot wrote:

Hi,

I try to set up attended transfer in my Asterisk Box . My
features.conf look like this:

[general]
parkext => 100
parkpos => 1-5
context => parkedcalls
parkingtime => 100
transferdigittimeout => 3l
courtesytone = beep
xfersound = beep
xferfailsound = invalid
featuredigittimeout = 500
;adsipark = yes
pickupexten = *8
[featuremap]
atxfer => *2
blindxfer => #
disconnect => *0
automon => *1

and when I press *2 console says something like this:

Oct 17 15:52:23 DEBUG[20916]: rtp.c:190 send_dtmf: Sending dtmf: 42
(*), at 10.2.20.65
Oct 17 15:52:23 DEBUG[20916]: channel.c:2762 ast_channel_bridge: Got
AST_BRIDGE_DTMF_CHANNEL_1 on c1 (SIP/rafal-89b1)
Oct 17 15:52:23 DEBUG[20916]: channel.c:2798 ast_channel_bridge:
Bridge stops bridging channels SIP/andrzej-0265 and SIP/rafal-89b1
Oct 17 15:52:23 DEBUG[20916]: res_features.c:600 ast_bridge_call: Read
from SIP/rafal-89b1 (1,42)
-- Attempting native bridge of SIP/andrzej-0265 and SIP/rafal-89b1
Oct 17 15:52:23 DEBUG[20916]: rtp.c:190 send_dtmf: Sending dtmf: 50
(2), at 10.2.20.65
Oct 17 15:52:23 DEBUG[20916]: channel.c:2762 ast_channel_bridge: Got
AST_BRIDGE_DTMF_CHANNEL_1 on c1 (SIP/rafal-89b1)
Oct 17 15:52:23 DEBUG[20916]: channel.c:2798 ast_channel_bridge:
Bridge stops bridging channels SIP/andrzej-0265 and SIP/rafal-89b1
Oct 17 15:52:23 DEBUG[20916]: res_features.c:600 ast_bridge_call: Read
from SIP/rafal-89b1 (1,50)
-- Attempting native bridge of SIP/andrzej-0265 and SIP/rafal-89b1

Does anyone know what's going on? What should I do to make attended
transfer works well?

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RE: [Asterisk-Users] call transfer

[Anton Krall]

This is configured on your features.conf file. In there you can see what
keys to use to do blind and attended transfers, be sure those lines are not
commented out. 

|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|[EMAIL PROTECTED]
|Sent: Lunes, 01 de Agosto de 2005 01:07 p.m.
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: [Asterisk-Users] call transfer
|
|
|
|Hi!
|
|I have searched answer how can I transfer calls with 
|asterisk,with no result.
|Can you advice me and show some example file how can I use SIP 
|phone to transfer calls by hitting # and get the "Transfer" 
|prompt and enter an extension I want to transfer to?
|
|Thanks for your answers
|
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RE: [Asterisk-Users] call transfer

[Cullin J. Wible]

You must use the 't' 'T' options in the Dial() command when placing calls to
and from the device.

We had extensions that were combinations of SIP and IAX devices and didn't
want/need this behavior on all of our devices so we setup our extensions
with something as follows:

Exten => 1000,1,Dial(Local/IAX-1000/[EMAIL PROTECTED]&Local/SIP-1000/[EMAIL 
PROTECTED], 60,
r)

[devices]
Exten => SIP-1000,1,Dial(SIP-XYZ, 60, tr)
Exten => IAX-1000,1,Dial(IAX-ABC, 60, r)


That will ring both devices using different dial statements for each.

Cullin

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Monday, August 01, 2005 2:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] call transfer



Hi!

I have searched answer how can I transfer calls with asterisk,with no
result.
Can you advice me and show some example file how can I use SIP phone to
transfer calls by hitting # and get the "Transfer" prompt and enter an
extension I want to transfer to?

Thanks for your answers




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Re: [Asterisk-Users] Call Transfer using SIP clients

[Brian Capouch]

Frank Schoep wrote:



The X-Lite softphone does indeed have a "Transfer" and "Hold" in the 
interface, but the functionality of those buttons appears to have been 
disabled when the client is connected and registered on the Asterisk server. 
Pressing the on-screen buttons doesn't have any effect while either having a 
call or while idle.




I'm pretty sure it's disabled on purpose on the "free" Linux version of 
the phone.  I remember reading that somewhere on their site once upon a 
time.


B.
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Re: [SPAM:***** SpamScore] Re: [Asterisk-Users] Call Transfer using SIP clients

[Frank Schoep]

On Monday 04 July 2005 16:47, Elwin Andriol wrote:
> I don't know if this will be of any help to you, but at least I can
> confirm problems with transfering calls with SIP agents. A little while
> ago we were having big problems getting transfers using DTMF to work.
>
> In that particular situation we were using a mix of only "hard" SIP
> devices (BT101's and Elmeg/Snom190's) and ZAP channels. We tried both
> the stable version of asterisk and the CVS HEAD, but without results
> (but negative). In the end, we solved the problem by not using DTMF
> transfers at all, but by using the transfer capabilities of the SIP
> devices themselves (transfer for and hold buttons). These buttons did
> not appear to work (correctly) with the stable asterisk version we
> initially used (1.0.7), but with the CVS HEAD (> 29-MAY-2005) they
> appear to work just fine.
>
> I'm not familiar with "soft" SIP agents, so I don't know if the ones you
> use have such build-in transfer capabilities as their hardware
> counterparts like the BT101's and Snom190's have. I they do, you might
> wan't to give it a try. This is of course rather a workaround than a
> solution to your problem.
>
> E. Andriol

The X-Lite softphone does indeed have a "Transfer" and "Hold" in the 
interface, but the functionality of those buttons appears to have been 
disabled when the client is connected and registered on the Asterisk server. 
Pressing the on-screen buttons doesn't have any effect while either having a 
call or while idle.

Related to client-side transferring: I set the "canreinvite" option in the SIP 
configuration to "no" because both clients are behind a NAT / firewall and I 
read in the documentation that you'd want to disable the "canreinvite" option 
in those situations. I haven't had any trouble because of this, as I stated 
earlier calling and talking is working without hitches.

I haven't had the chance to try hardware phones yet, the testing I'm doing at 
the moment involves softphones only. Now that I think of it, I'll try to 
setup other applications again which might send DTMFs in a different form 
compared to X-Lite.

In the meantime thanks in advance to everyone involved in this thread now and 
in the future.

Sincerely,

Frank
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Re: [SPAM:***** SpamScore] RE: [Asterisk-Users] Call Transfer using SIP clients

[Frank Schoep]

On Tuesday 05 July 2005 05:57, Tulika Pradhan wrote:
> call transfer works for me fine without any additions in features.conf
> by simply using Dial(SIP/${EXTEN},20,tT)
> and pressing #
> this works both from caller as well as callee.
>
> tulika

Could you provide me with some more information so I can check where the 
differences in our setups are? It would really help to see how you 
implemented your extensions and SIP configuration. Could you describe the 
following regarding your Asterisk installation:

- Asterisk version
- The SIP clients you use
- Excerpt of "extensions.conf", which definitions and contexts do you include
- Excerpt of "features.conf", which lines (if any) are in there
- (Maybe) an excerpt of "sip.conf", how are the SIP peers configured

I hope you find the time to post these bits and pieces as it will make it 
easier for me to debug the situation. I've already tried numerous settings 
and combinations of options, but haven't had any luck yet. Thanks in advance 
for your precious time.

If anyone else has some ideas regarding my question, feel free to jump in - 
the more the merrier.

Sincerely,

Frank
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RE: [Asterisk-Users] Call Transfer using SIP clients

[Tulika Pradhan]

call transfer works for me fine without any additions in features.conf
by simply using Dial(SIP/${EXTEN},20,tT)
and pressing #
this works both from caller as well as callee.

tulika


From: Frank Schoep <[EMAIL PROTECTED]>
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion 


To: Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users] Call Transfer using SIP clients
Date: Mon, 4 Jul 2005 16:11:13 +0200

Hello all,

First of all, let me apologize about the length of this message, but I 
suppose

it was necessary to include the details.

I've spent quite some time already trying to get the call transfer function 
to
work on my Asterisk installation. Let me first describe the general 
situation

of the setup I am using, so you might be able to pinpoint the cause of the
problem.

I'm currently using Asterisk CVS as of July 4th 2005. The only means of
communication at the moment is the XTen X-Lite SIP Client, I already added
the following entries to my "sip.conf" configuration file:

[frank]
canreinvite=no
type=friend
secret=frank
username=frank
nat=yes
host=dynamic

[test]
canreinvite=no
type=friend
secret=test
username=test
nat=yes
host=dynamic

The SIP setup is working without a problem, the X-Lite application 
correctly

registers the users and I can set up calls between them. I've also tested
queues and they work without a problem, too. Next up is my extensions
configuration, of which the interesting section now looks like this:

[default]
include => general ; longshot, added out of desparation
include => parkedcalls ; longshot, added out of desparation
include => featuremap ; longshot, added out of desparation

exten => 800,1,Answer
exten => 800,2,Dial(SIP/frank,20,tT)
exten => 800,3 Hangup

exten => 802,1,Answer
exten => 802,2,Dial(SIP/test,20,tT)
exten => 802,3 Hangup

Notice the inclusion of several contexts that should or would have to be
defined in the features configuration. My features.conf looks something 
like

this, I trimmed the 'general' section for brevity:

[general]
; (trimmed) default options

[featuremap]
blindxfer => #1 ; Blind transfer
disconnect => *0 ; Disconnect
automon => *1 ; One Touch Record
atxfer => *2 ; Attended transfer

My testing scenario starts as follows:
- log in both X-Lite SIP clients
- from the 'test' phone, call extension 800
- on X-Lite client 'frank' accept the call
- talk to eachother

At this point I want to transfer to call to another extension, also defined 
in

"sip.conf" but unlisted here. The problem is that nothing happens when I
press the "#1" or "*2" keys in the 'frank' X-Lite client. I also tested 
these

key combinations on the 'test' X-Lite client during the call, but that also
had not effect.

I searched the web and the mailing list archive for a solution, and if I
recall correctly, someone stated that call transfer is only available for
calls originating from the PSTN. Is this correct, also in regard of the
current version of Asterisk? Has anyone got an idea how to get call 
transfer

to work?

One thing I tried was to change the DTMF settings in the clients, so they 
are

sent in-band, but this also didn't help. Should I revert this option?

Thanks in advance for your time and patience.

Sincerely,

Frank Schoep
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Re: [Asterisk-Users] Call Transfer using SIP clients

[Elwin Andriol]

Frank Schoep wrote:


Hello all,

First of all, let me apologize about the length of this message, but I suppose 
it was necessary to include the details.


I've spent quite some time already trying to get the call transfer function to 
work on my Asterisk installation. Let me first describe the general situation 
of the setup I am using, so you might be able to pinpoint the cause of the 
problem.


I'm currently using Asterisk CVS as of July 4th 2005. The only means of 
communication at the moment is the XTen X-Lite SIP Client, I already added 
the following entries to my "sip.conf" configuration file:


[frank]
canreinvite=no
type=friend
secret=frank
username=frank
nat=yes
host=dynamic

[test]
canreinvite=no
type=friend
secret=test
username=test
nat=yes
host=dynamic

The SIP setup is working without a problem, the X-Lite application correctly 
registers the users and I can set up calls between them. I've also tested 
queues and they work without a problem, too. Next up is my extensions 
configuration, of which the interesting section now looks like this:


[default]
include => general ; longshot, added out of desparation
include => parkedcalls ; longshot, added out of desparation
include => featuremap ; longshot, added out of desparation

exten => 800,1,Answer
exten => 800,2,Dial(SIP/frank,20,tT)
exten => 800,3 Hangup

exten => 802,1,Answer
exten => 802,2,Dial(SIP/test,20,tT)
exten => 802,3 Hangup

Notice the inclusion of several contexts that should or would have to be 
defined in the features configuration. My features.conf looks something like 
this, I trimmed the 'general' section for brevity:


[general]
; (trimmed) default options

[featuremap]
blindxfer => #1 ; Blind transfer
disconnect => *0 ; Disconnect
automon => *1 ; One Touch Record
atxfer => *2 ; Attended transfer

My testing scenario starts as follows:
- log in both X-Lite SIP clients
- from the 'test' phone, call extension 800
- on X-Lite client 'frank' accept the call
- talk to eachother

At this point I want to transfer to call to another extension, also defined in 
"sip.conf" but unlisted here. The problem is that nothing happens when I 
press the "#1" or "*2" keys in the 'frank' X-Lite client. I also tested these 
key combinations on the 'test' X-Lite client during the call, but that also 
had not effect.


I searched the web and the mailing list archive for a solution, and if I 
recall correctly, someone stated that call transfer is only available for 
calls originating from the PSTN. Is this correct, also in regard of the 
current version of Asterisk? Has anyone got an idea how to get call transfer 
to work?


One thing I tried was to change the DTMF settings in the clients, so they are 
sent in-band, but this also didn't help. Should I revert this option?


Thanks in advance for your time and patience.

Sincerely,

Frank Schoep
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I don't know if this will be of any help to you, but at least I can 
confirm problems with transfering calls with SIP agents. A little while 
ago we were having big problems getting transfers using DTMF to work.


In that particular situation we were using a mix of only "hard" SIP 
devices (BT101's and Elmeg/Snom190's) and ZAP channels. We tried both 
the stable version of asterisk and the CVS HEAD, but without results 
(but negative). In the end, we solved the problem by not using DTMF 
transfers at all, but by using the transfer capabilities of the SIP 
devices themselves (transfer for and hold buttons). These buttons did 
not appear to work (correctly) with the stable asterisk version we 
initially used (1.0.7), but with the CVS HEAD (> 29-MAY-2005) they 
appear to work just fine.


I'm not familiar with "soft" SIP agents, so I don't know if the ones you 
use have such build-in transfer capabilities as their hardware 
counterparts like the BT101's and Snom190's have. I they do, you might 
wan't to give it a try. This is of course rather a workaround than a 
solution to your problem.


E. Andriol

--
---
HeuvelTop ICT Diensten v.o.f.
---
"There are management solutions to technical problems,
but no technical solutions to management problems"
---

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Re: [Asterisk-Users] Call Transfer Problem

[Kevin P. Fleming]

Adam Robins wrote:


The double-star now works great.  If I press it while on a call, I go
into transfer mode.  The problem is that the # still works as well!
Shouldn't the atzfer specification turn off the #?


Blind transfers are on '#' by default, so you may need to move them to 
another sequence as well.

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Re: [Asterisk-Users] Call transfer

[Henry Devito]

Thanks, Don't know how I could have missed that,  This works on incoming 
calls to the station and calls from station to station.  How do I make it 
work if I dial out over a zap channel and then want to transfer to another 
extension the # doesn't do anything except generate tone on the line.  N0 
transfer,
- Original Message - 
From: "Eric Wieling aka ManxPower" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Sent: Thursday, April 28, 2005 12:26 PM
Subject: Re: [Asterisk-Users] Call transfer


Henry Devito wrote:
I just bought one of these zyxel wireless phones, of course there is no 
transfer key.  Is there a patch for the stable 1.0.7 that I can implement 
# or any other key or combination to initiate a transfer?

I looked briefly through the wiki and archived lists and didn't see much.
"show application dial"  Pay special attention to the t/T options. Those 
options are for devices that are too stupid or brain dead to have a 
transfer key that works.
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Re: [Asterisk-Users] Call transfer

[Eric Wieling aka ManxPower]

Henry Devito wrote:
I just bought one of these zyxel wireless phones, of course there is no 
transfer key.  Is there a patch for the stable 1.0.7 that I can 
implement # or any other key or combination to initiate a transfer?

I looked briefly through the wiki and archived lists and didn't see much.
"show application dial"  Pay special attention to the t/T options. 
Those options are for devices that are too stupid or brain dead to have 
a transfer key that works.
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Re: [Asterisk-Users] Call Transfer Features

[C F]

Use ## it's in features.conf


On Tue, 22 Mar 2005 07:42:27 -0700, Damon Estep
<[EMAIL PROTECTED]> wrote:
> Looking for a liitle help if anyone has dealt with this;
> 
> The options on dial and queue of t (allow called party to transfer call)
> and T (allow calling aprty to transfer call) seem to work fine (as long
> as you do not confuse them with the same t and T that indicate
> timeout!).
> 
> The problem I am having is the use of the # key to do so. Many times a
> caller will palce a call to an IVR that requires the use of the # key to
> access a feature on the remote IVR, but * intercepts the # and offers
> the transfer prompt (as expected).
> 
> The solution seems to be to change the feature to use a different key,
> although I do not know how. I am sure I could find it on the wiki, but
> it seems to be down this morning?
> 
> Any suggestinos on a transfer key sequence that does not interfere with
> external IVRs that often?
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Re: [Asterisk-Users] Call transfer

[Robert Webb]

On Tue, 8 Mar 2005 14:17:23 -0300
 "Alejandro G" <[EMAIL PROTECTED]> wrote:

I have 2 asterisk box in different locations. When I 
received a call in one
location and want to transfer it to an extension in the 
other location the
external call is hanged up when the person who is 
transfering the call hangs
up. The sequence is like this:

1. Call is received and attended by person 1 in 
extension 3000 in location 1
2. Person 1 press flash and dials extension 4000 in 
location 2
3. Person 2 in extension 4000 i location 2 pick up the 
call and talk to
person 1
4. Person 1 hangs up and the external call is hanged up.

Is anything wrong?
Thanks.
Alejandro Ghergherian

Post your conf file sections that are relavant from 
extensions, sip or iax That way we can see how you are 
handling things and be able to see what might be wrong.

Also, bring up a cli and monitor what is happening during 
the transfer. Then copy and paste that to the list.

Robert
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Re: [Asterisk-Users] Call transfer questions

[Paul Zimm]

also tried the following without luck
[featuremap]
blindxfer => #1; Blind transfer
disconnect => *0   ; Disconnect
automon => *1  ; One Touch Record
atxfer => *2
it still seems to want to accept only # as transfer
I am running Asterisk CVS-v1-0-03/07/05-06:50:06
You are running V1.0.x stable of asterisk. Tthe attended transfer feature
is only available in CVS-HEAD, which at some point (June ?) will become 
1.1.x stable
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Re: [Asterisk-Users] Call transfer questions

[Jer]

At 05:44 AM 3/7/2005, you wrote:
Dear all
I am trying to work out how make call trasfer work the way I want is
I am the called party I want to transfer a call so I press # and enter the 
ext but then it disconnects me
this is a blind transfer
how do I make it so its not a blind transfer so i can talk to the person 
before i transfer the call...and go backl to the orig caller if the 
transfered to ext doesnt answer
also can the caller hear MOH while I am talking to person I am transfering 
the call to

what would I need to do this
just point me in the right direction and i'll go read some more...
I using so far is T in dial()
Thanks
sorry for the noob question
also tried the following without luck
[featuremap]
blindxfer => #1; Blind transfer
disconnect => *0   ; Disconnect
automon => *1  ; One Touch Record
atxfer => *2
it still seems to want to accept only # as transfer
I am running Asterisk CVS-v1-0-03/07/05-06:50:06



Jer
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RE: [Asterisk-Users] call transfer to conference call

[mattf]

You need to send a Manager command(Redirect Action) to the asterisk server.
(BYou can do this by connecting to the manager API through any kind of
(Btelnet-type connector in any number of programming languages: Perl, C, PHP,
(Betc..
(B
(BTake a look at the WIKI for more info:
(B
(Bhttp://www.voip-info.org/wiki-Asterisk+manager+API
(B
(BThe redirect of the existing conversation to the conference room(meetme)
(Bwould look something like this:
(B
(BAction: Redirect
(BChannel: Zap/73-1
(BExtraChannel: SIP/199testphone-1f3c
(BExten: 8600029
(BContext: default
(BPriority: 1
(B
(Bwhere 8600029 is the meetme room.
(B
(BHope that helps,
(B
(BMATT---
(B
(B
(B-Original Message-
(BFrom: Kuniyoshi Murata [mailto:[EMAIL PROTECTED]
(BSent: Monday, January 03, 2005 9:26 PM
(BTo: Asterisk Users Mailing List - Non-Commercial Discussion
(BSubject: [Asterisk-Users] call transfer to conference call
(B
(B
(BHi, 
(B
(BI have following setup already. 
(B
(BPSTN call via zap channel is working, Xlite via sip channel is working, and
(Bconference call is working. 
(B
(BAnd here is what I want to do. 
(B
(BA. My friends are making conference call in a conference room and all
(Bclients are Xlite.
(BB. I make a call from my Xlite to PSTN phone of the guest speaker I want to
(Binvite to the conference call
(BC. I will transfer my call with the guest to the conference call that is
(Balready taking place (that is A) 
(B
(BOf course I can do A and B, but to do C what kind of command, operation,
(Band/or script are needed? 
(B
(BAny comments, directions and references are welcome. 
(B
(BThanks in advance
(BKuni 
(B
(B-- 
(BKuniyoshi Murata.iChat/AIM:macwebcaster
(BEnglish-Japanese Interpreter mailto:[EMAIL PROTECTED]
(BMacintosh Webcast Specialisthttp://www.macwebcaster.com
(B___
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Re: [Asterisk-Users] Call Transfer Problem

[Michael Bielicki]

you need the x or X option to your Dial command. "show application
dial" is your friend ...

cheers

Michael


On Mon, 11 Oct 2004 08:37:36 -0500 (CDT), [EMAIL PROTECTED]
<[EMAIL PROTECTED]> wrote:
> On Fri, 8 Oct 2004, Michael Nolan wrote:
> 
> Hi !
> 
> I have checked my asterisk. It contains this patch or thBis patch is
> already compiled into it. can you plz guide me as to how i can make use
> of it ? I have pressed '#' but it doesnot give me any dial tone. Are there
> any special changes that need to be done in extensions.conf to make it
> work ? plz help me in this regard.
> 
> Usman.
> 
> > This patch works a treat for us:
> >
> > http://bugs.digium.com/bug_view_page.php?bug_id=0002460
> >
> > Makes all # transfers attended, but the transfer button on the phones
> > can still be used for blind transfers from our SIP phones.
> >
> > Cheers,
> >
> > Michael
> >
> >
> > On Fri, 8 Oct 2004 01:56:53 -0500 (CDT), [EMAIL PROTECTED]
> > <[EMAIL PROTECTED]> wrote:
> > > Hi Users,
> > >
> > > I am having a prblem using attended call transfer with asterisk. Actually
> > > my sip phone does not seem to support it. Can i use attended call transfer
> > > using the dial plan ... ??? means can somehow messing up with
> > > extesnions.conf I can get attended call transfer ? And yes also is there
> > > any way I can do it with AGI scripting ? Any AGI similar examples will be
> > > a lot of help. Thanks !
> > >
> > > Usman.
> > >
> > > ___
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> > >http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> >
> >
> >
> 
> 
> 
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-- 
Michael Bielicki
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Re: [Asterisk-Users] Call Transfer Problem

[usman]

On Fri, 8 Oct 2004, Michael Nolan wrote:

Hi ! 

I have checked my asterisk. It contains this patch or thBis patch is 
already compiled into it. can you plz guide me as to how i can make use 
of it ? I have pressed '#' but it doesnot give me any dial tone. Are there 
any special changes that need to be done in extensions.conf to make it 
work ? plz help me in this regard.

Usman.

> This patch works a treat for us:
> 
> http://bugs.digium.com/bug_view_page.php?bug_id=0002460
> 
> Makes all # transfers attended, but the transfer button on the phones
> can still be used for blind transfers from our SIP phones.
> 
> Cheers,
> 
> Michael
> 
> 
> On Fri, 8 Oct 2004 01:56:53 -0500 (CDT), [EMAIL PROTECTED]
> <[EMAIL PROTECTED]> wrote:
> > Hi Users,
> > 
> > I am having a prblem using attended call transfer with asterisk. Actually
> > my sip phone does not seem to support it. Can i use attended call transfer
> > using the dial plan ... ??? means can somehow messing up with
> > extesnions.conf I can get attended call transfer ? And yes also is there
> > any way I can do it with AGI scripting ? Any AGI similar examples will be
> > a lot of help. Thanks !
> > 
> > Usman.
> > 
> > ___
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> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> > 
> 
> 
> 

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Re: [Asterisk-Users] Call Transfer Problems with Grandstream Budgetone 100 Phone

[Steve Totaro]

 

  - Original Message - 
  From: 
  James Dutton 
  To: [EMAIL PROTECTED] 
  
  Sent: Thursday, August 05, 2004 6:27 
  AM
  Subject: [Asterisk-Users] Call Transfer 
  Problems with Grandstream Budgetone 100 Phone
  
  I have two 
  Grandstream Budgetone 100 phones connected to my local asterisk 
  server.
   
  I am able to 
  receive incoming calls, and place outgoing calls, but have two 
  problems...
   
  1) I cannot 
  transfer calls between the two phones. Pressing transfer takes me to a dial 
  tone, I key in the internal number then press # or transfer, and the original 
  call is cut off and the other internal phone does not 
ring.
   
  try pressing send 
  instead of # or transfer
   
  2) I cannot hear 
  an outgoing ringing tone when placing the call.
   
  add r to your dial 
  statement in extensions.conf
   
  I would be 
  extremely grateful to anyone out who has experience of these phones and can 
  help.
   
  Regards
   
  James 
  Dutton

RE: [Asterisk-Users] Call Transfer Problems with Grandstream Budgetone 100 Phone

[Sergio Serrano]

Title: Mensaje



Push 
send after you number,
 
srsergio
 
 -Mensaje original-De: 
[EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] En nombre de James 
DuttonEnviado el: jueves, 05 de agosto de 2004 12:28Para: 
[EMAIL PROTECTED]Asunto: [Asterisk-Users] Call Transfer 
Problems with Grandstream Budgetone 100 Phone

  I have two 
  Grandstream Budgetone 100 phones connected to my local asterisk 
  server.
   
  I am able to 
  receive incoming calls, and place outgoing calls, but have two 
  problems...
   
  1) I cannot 
  transfer calls between the two phones. Pressing transfer takes me to a dial 
  tone, I key in the internal number then press # or transfer, and the original 
  call is cut off and the other internal phone does not 
ring.
   
  2) I cannot hear 
  an outgoing ringing tone when placing the call.
   
  I would be 
  extremely grateful to anyone out who has experience of these phones and can 
  help.
   
  Regards
   
  James 
  Dutton

Re: [Asterisk-Users] call transfer with consultation

[Andrew Kohlsmith]

> For example, when an input call comes through X100P,
> my Zap/3 extension rings. I pickup Zap/3 and I want to
> transfer the call to Zap/4, but before to establish
> the call between X100P and Zap/4 I need to request
> Zap/4 for answering the call.

Currently not possible, although here is a workaround since you are using Zap 
interfaces:

Call comes in and you answer it.
Hook flash (briefly hang up and pick up the phone again) -- caller is on hold 
and you can dial the extension you want to transfer it to.
Talk to the extension
Hook flash again, and now you, the extension and the caller are in a 3-way 
call.
Hang up -- the call is now transfered.

Hope this helps.

Regards,
Andrew
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Re: [Asterisk-Users] Call transfer

[Paul Liew]

Hi Mick,

It's going to be hard for anybody here on the list to help you, unless you
are more specific, ie, what you did exactly to get a crash, and console
output (with verbose set) debugs, logs (under /var/log/asterisk) and some
configuration files. We'll be in a better position to help you then without
trying to be mind readers.

Paul
- Original Message - 
From: <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, November 17, 2003 6:49 PM
Subject: RE: [Asterisk-Users] Call transfer


>
> WARNING[1242952640]: File app_dial.c, Line 318 (wait_for_answer): Unable
> to forward voice
>
> This is what I get
>
> And a crash
>
>
>
>
>
>
>
> Regards Mick
>
>
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> [EMAIL PROTECTED]
> Sent: Monday, 17 November 2003 5:14 PM
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] Call transfer
>
>
> On Mon, 17 Nov 2003 [EMAIL PROTECTED] wrote:
>
> > Does anyone know how to make
> >
> > Calls auto transfer to a mobile if no one answers ??
>
> suppose your mobile number is +923008508070
>
> exten => 15,1,Dial(IAX/farfon|30) ; try for 30 seconds on IAX exten =>
> 15,2,Dial(Zap/1/03008508070|45) ; then try for 45 on my cell
>
>
> - wasim
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RE: [Asterisk-Users] Call transfer

[mick]

WARNING[1242952640]: File app_dial.c, Line 318 (wait_for_answer): Unable
to forward voice

This is what I get

And a crash







Regards Mick



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Monday, 17 November 2003 5:14 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Call transfer


On Mon, 17 Nov 2003 [EMAIL PROTECTED] wrote:

> Does anyone know how to make
> 
> Calls auto transfer to a mobile if no one answers ??

suppose your mobile number is +923008508070

exten => 15,1,Dial(IAX/farfon|30) ; try for 30 seconds on IAX exten =>
15,2,Dial(Zap/1/03008508070|45) ; then try for 45 on my cell


- wasim
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Re: [Asterisk-Users] Call transfer

[wasim]

On Mon, 17 Nov 2003 [EMAIL PROTECTED] wrote:

> Does anyone know how to make
> 
> Calls auto transfer to a mobile if no one answers ??

suppose your mobile number is +923008508070

exten => 15,1,Dial(IAX/farfon|30) ; try for 30 seconds on IAX
exten => 15,2,Dial(Zap/1/03008508070|45) ; then try for 45 on my cell


- wasim
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Re: [Asterisk-Users] Call transfer ATA186

[Mark Spencer]

It might be possible to implement BYE also style transfer which is totally
deprecated in SIP but appears to be (at least as of a few months ago) what
the ATA's used.  If you want to add a bug to the bug tracker (including
SIP debug) from your attmpted call, I can take a look at it.

Mark

On Tue, 19 Aug 2003, Brian Capouch wrote:

> CW_ASN wrote:
> > I use 3Party using flash key and dialing the extension. When the other ATA
> > answer the call, I press flash again.
> > I test Call Transfer using # key (#ext#). If you know another way to do
> > that, please let me know.
> >
>
> I'm tearing my hair out trying to exercise a variation on this theme.
> I'm mad from trying, so there may be some realy easy thing that is
> escaping me here.
>
> What I want to do is answer a call, put the caller on hold, dial up
> another extension and speak briefly with the person who answers (e.g. "I
> have Mr. Faltzernaust on the line") and then bow out and leave the
> caller and the callee to talk.
>
> When I try the 3party the whole thing goes to s**t when I hang up, and
> in a transfer I don't get that opportunity to announce the caller.
>
> Is there an easy way to do what I want to do here?
>
> Thx.
>
> B.
>
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Re: [Asterisk-Users] Call transfer ATA186

[John Todd]

At 12:07 AM -0700 8/19/03, John Todd wrote:
At 1:42 AM -0500 8/19/03, Brian Capouch wrote:
CW_ASN wrote:
I use 3Party using flash key and dialing the extension. When the other ATA
answer the call, I press flash again.
I test Call Transfer using # key (#ext#). If you know another way to do
that, please let me know.
I'm tearing my hair out trying to exercise a variation on this 
theme. I'm mad from trying, so there may be some realy easy 
thing that is escaping me here.

What I want to do is answer a call, put the caller on hold, dial up 
another extension and speak briefly with the person who answers 
(e.g. "I have Mr. Faltzernaust on the line") and then bow out and 
leave the caller and the callee to talk.

When I try the 3party the whole thing goes to s**t when I hang up, 
and in a transfer I don't get that opportunity to announce the 
caller.

Is there an easy way to do what I want to do here?

Thx.

B.


If it's any solace to you, there is no way I know of that one can do 
supervised call transfer (what you describe above) on an ATA-186. 
The "#" key trick (appending a "t" on your Dial statements) allows 
you to transfer, but it's an unsupervised transfer.  That trickery 
is done 100% in Asterisk, so you may be able to hack the source code 
to get it working with some different technique.

The workaround is:

 - call party #1, establish call
 - hit flash, dial new number, hit # (the # is locally interpreted 
by the ATA as "finished dialing" character)
 - talk to the third party; tell them a call is about to come in.
 - have the third party hang up
 - hit flash again, bringing you back to party #1
 - hit "#" and type in the number of third party, type "#" (this 
time, interpreted by Asterisk as "finished dialing" character)
 - hang up

Sucks, doesn't it?

JT
Duh.  I shouldn't work so late.  All this time I had simply been 
forgetting to hang up on people.  :)  I suppose this makes sense; 
it's half of a three-way call, except after you talk to the third 
party, you just hang up.  The ATA then redials the third party and 
connects the two together.  The RTP stream no longer goes through the 
ATA after the redial, if you're curious.

http://www.cisco.com/en/US/products/hw/gatecont/ps514/products_administration_guide_chapter09186a0080150e5a.html#1015900

JT
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Re: [Asterisk-Users] Call transfer ATA186

[Dan]

Hi John,

- Original Message - 
From: "John Todd" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, August 19, 2003 10:07 AM
Subject: Re: [Asterisk-Users] Call transfer ATA186



> If it's any solace to you, there is no way I know of that one can do
> supervised call transfer (what you describe above) on an ATA-186.
> The "#" key trick (appending a "t" on your Dial statements) allows
> you to transfer, but it's an unsupervised transfer.  That trickery is
> done 100% in Asterisk, so you may be able to hack the source code to
> get it working with some different technique.

It works for me now using Flash key. When updated to the latest CVS and
setting the call parameters in ATA config the unattended transfer works on
ATA too now.

>
> The workaround is:
>
>   - call party #1, establish call
>   - hit flash, dial new number, hit # (the # is locally interpreted by
> the ATA as "finished dialing" character)
>   - talk to the third party; tell them a call is about to come in.
>   - have the third party hang up
>   - hit flash again, bringing you back to party #1
>   - hit "#" and type in the number of third party, type "#" (this
> time, interpreted by Asterisk as "finished dialing" character)
>   - hang up
>
> Sucks, doesn't it?

For me the workaround was to define a dial loop in extensions.conf when a
phone is busy (call waiting must be activated).
Then do like that.
- answer the call on ATA
- pres Flash key
- call the final destination
- wait to answer and inform it about the call
- close the phone
- fastbusy on final destination
- close the final destination
- final destination is automatically called by Asterisk from the initial
caller.
- establish the call between them.

All you need is to keep dialing the final destination (in a loop) if busy,
till it rings.

It seems that with the latest CVS update and the correct parameters in ATA
config page this works without the need to do this trick.
Just do like that:
- answer the call on ATA
- pres Flash key
- call the final destination
- wait to answer and inform it about the call
- close the phone
- MOH is stopped and the call is automatically establish between them .
 Tested several times with different configurations:
7960->ATA->ATA
X-Lite->7960->ATA
7960->X-Lite->ATA
ATA->7960->ATA
ATA->ATA->7960

and it works for me now.

BR,
Dan

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Re: [Asterisk-Users] Call transfer ATA186

[John Todd]

At 1:42 AM -0500 8/19/03, Brian Capouch wrote:
From: Brian Capouch <[EMAIL PROTECTED]>
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Call transfer ATA186
Reply-To: [EMAIL PROTECTED]
Date: Tue, 19 Aug 2003 01:42:53 -0500
CW_ASN wrote:
I use 3Party using flash key and dialing the extension. When the other ATA
answer the call, I press flash again.
I test Call Transfer using # key (#ext#). If you know another way to do
that, please let me know.
I'm tearing my hair out trying to exercise a variation on this 
theme. I'm mad from trying, so there may be some realy easy 
thing that is escaping me here.

What I want to do is answer a call, put the caller on hold, dial up 
another extension and speak briefly with the person who answers 
(e.g. "I have Mr. Faltzernaust on the line") and then bow out and 
leave the caller and the callee to talk.

When I try the 3party the whole thing goes to s**t when I hang up, 
and in a transfer I don't get that opportunity to announce the 
caller.

Is there an easy way to do what I want to do here?

Thx.

B.


If it's any solace to you, there is no way I know of that one can do 
supervised call transfer (what you describe above) on an ATA-186. 
The "#" key trick (appending a "t" on your Dial statements) allows 
you to transfer, but it's an unsupervised transfer.  That trickery is 
done 100% in Asterisk, so you may be able to hack the source code to 
get it working with some different technique.

The workaround is:

 - call party #1, establish call
 - hit flash, dial new number, hit # (the # is locally interpreted by 
the ATA as "finished dialing" character)
 - talk to the third party; tell them a call is about to come in.
 - have the third party hang up
 - hit flash again, bringing you back to party #1
 - hit "#" and type in the number of third party, type "#" (this 
time, interpreted by Asterisk as "finished dialing" character)
 - hang up

Sucks, doesn't it?

JT
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Re: [Asterisk-Users] Call transfer ATA186

[Dan]

Hi,

You can find the standard procedure to do this on ATA here:
http://www.cisco.com/en/US/products/hw/gatecont/ps514/products_administration_guide_chapter09186a0080150e5a.html#1015891

for unattended transfer
and:
http://www.cisco.com/en/US/products/hw/gatecont/ps514/products_administration_guide_chapter09186a0080150e5a.html#1015900
for attended transfer.

They both work for me now.
You can use '#' procedure too.

BR,
Dan


- Original Message - 
From: "Brian Capouch" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, August 19, 2003 9:42 AM
Subject: Re: [Asterisk-Users] Call transfer ATA186


> CW_ASN wrote:
> > I use 3Party using flash key and dialing the extension. When the other
ATA
> > answer the call, I press flash again.
> > I test Call Transfer using # key (#ext#). If you know another way to do
> > that, please let me know.
> >
>
> I'm tearing my hair out trying to exercise a variation on this theme.
> I'm mad from trying, so there may be some realy easy thing that is
> escaping me here.
>
> What I want to do is answer a call, put the caller on hold, dial up
> another extension and speak briefly with the person who answers (e.g. "I
> have Mr. Faltzernaust on the line") and then bow out and leave the
> caller and the callee to talk.
>
> When I try the 3party the whole thing goes to s**t when I hang up, and
> in a transfer I don't get that opportunity to announce the caller.
>
> Is there an easy way to do what I want to do here?
>
> Thx.
>
> B.
>
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>

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Re: [Asterisk-Users] Call transfer ATA186

[Dan]

Hi Gus,

I have all those problems too, but all gone when update to the latest
Asterisk CVS.
Now I can use unattended transfer on ATA with '#' or Flash.

Check the following settings in ATA (I presume that SIP is used):
CallFeatures: 0x0ff80ff8
ConnectMode:0x00460400

Tell me exactly how have you tried to do the transfer (step by step).

Best regards,
Dan

- Original Message - 
From: "ASN" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, August 19, 2003 12:51 AM
Subject: [Asterisk-Users] Call transfer ATA186


Hi all:

I'm testing a new installation of *, bringing up some ATA186. In *
environment, all stuff works greats. The only thing that don't work is a
Call Transfer, but the 3Party works ok. Some time ago I read that somebody
had proven this functionality successfully. If somebody knows what I
missing, please let me know.

Thanks in advance,

Gus


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Re: [Asterisk-Users] Call transfer ATA186

[Brian Capouch]

CW_ASN wrote:
I use 3Party using flash key and dialing the extension. When the other ATA
answer the call, I press flash again.
I test Call Transfer using # key (#ext#). If you know another way to do
that, please let me know.
I'm tearing my hair out trying to exercise a variation on this theme. 
I'm mad from trying, so there may be some realy easy thing that is 
escaping me here.

What I want to do is answer a call, put the caller on hold, dial up 
another extension and speak briefly with the person who answers (e.g. "I 
have Mr. Faltzernaust on the line") and then bow out and leave the 
caller and the callee to talk.

When I try the 3party the whole thing goes to s**t when I hang up, and 
in a transfer I don't get that opportunity to announce the caller.

Is there an easy way to do what I want to do here?

Thx.

B.

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Re: [Asterisk-Users] Call transfer ATA186

[CW_ASN]

I will copy the configurations and let you know (may be some parameter is
wrong).

Thanks a lot!

Gus

- Original Message -
From: "Brian West" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, August 18, 2003 7:03 PM
Subject: Re: [Asterisk-Users] Call transfer ATA186


> Works for me.. I can press # and dial the ext and press # to transfer a
> call.
>
> www.bkw.org/~brian/ata.html for the settings I used in my ATA
>
> bkw
>
> On Mon, 18 Aug 2003, ASN wrote:
>
> > Hi all:
> >
> > I'm testing a new installation of *, bringing up some ATA186. In *
environment, all stuff works greats. The only thing that don't work is a
Call Transfer, but the 3Party works ok. Some time ago I read that somebody
had proven this functionality successfully. If somebody knows what I
missing, please let me know.
> >
> > Thanks in advance,
> >
> > Gus
> >
> >
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Re: [Asterisk-Users] Call transfer ATA186

[CW_ASN]

I use 3Party using flash key and dialing the extension. When the other ATA
answer the call, I press flash again.
I test Call Transfer using # key (#ext#). If you know another way to do
that, please let me know.

Regards,

Gus

- Original Message -
From: "Fredrik Hedberg" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, August 18, 2003 7:15 PM
Subject: Re: [Asterisk-Users] Call transfer ATA186


> How exactly does you 3Party calling work? ;)
>
> Fred
>
> ASN wrote:
>
> > Hi all:
> >
> > I'm testing a new installation of *, bringing up some ATA186. In *
> > environment, all stuff works greats. The only thing that don't work is
> > a Call Transfer, but the 3Party works ok. Some time ago I read that
> > somebody had proven this functionality successfully. If somebody knows
> > what I missing, please let me know.
> >
> > Thanks in advance,
> >
> > Gus
> >
> >
>
>
>
>
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Re: [Asterisk-Users] Call transfer ATA186

[Fredrik Hedberg]

How exactly does you 3Party calling work? ;)

Fred

ASN wrote:

Hi all:
 
I'm testing a new installation of *, bringing up some ATA186. In * 
environment, all stuff works greats. The only thing that don't work is 
a Call Transfer, but the 3Party works ok. Some time ago I read that 
somebody had proven this functionality successfully. If somebody knows 
what I missing, please let me know.
 
Thanks in advance,
 
Gus
 
 




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Re: [Asterisk-Users] Call transfer ATA186

[Brian West]

Works for me.. I can press # and dial the ext and press # to transfer a
call.

www.bkw.org/~brian/ata.html for the settings I used in my ATA

bkw

On Mon, 18 Aug 2003, ASN wrote:

> Hi all:
>
> I'm testing a new installation of *, bringing up some ATA186. In * environment, all 
> stuff works greats. The only thing that don't work is a Call Transfer, but the 
> 3Party works ok. Some time ago I read that somebody had proven this functionality 
> successfully. If somebody knows what I missing, please let me know.
>
> Thanks in advance,
>
> Gus
>
>
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Re: [Asterisk-Users] Call Transfer

[Chee Foong]

Excellent idea mate,

Now I am able to do what I wanted with Great help from Jeremy McNamara.

Thanks alot

Foong

- Original Message -
From: "Andy Powell" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, July 30, 2003 6:56 PM
Subject: Re: [Asterisk-Users] Call Transfer


> Foong
>
> Take a look at the sample.call file, modifying the settings in there and
copying the file to /var/spool/asterisk/outgoing will cause asterisk to dial
the call.. an example config is below
>
> Channel: SIP/[EMAIL PROTECTED]
> MaxRetries: 2
> RetryTime: 60
> WaitTime: 30
> Context: mysipcontext2
> Extension: 2000
> Priority: 1
>
> This will make asterisk dial exten 1000 in the context mysipcontext when
it's answered it will then call exten 2000 in mysipcontext2..
>
> All you need is a script to lookup in the database and generate the script
file for you and it's done.
>
> HTH
>
> Andy
>
>
> *** REPLY SEPARATOR  ***
>
> On 30/07/2003 at 16:30 Chee Foong wrote:
>
> >Hello Dan,
> >
> >Thanks for you reply.
> >
> >Base on you recomendation using the 'T' argument. I manage to do call
> >transfer an it works really well.
> >
> >My problem comes when my boss comes out with a superb idea where the
> >transfering process is automated without involving a human :(
> >
> >Say asterisk get 2 numbers (from database, text file, etc), one belongs
> >party A and the other belongs to party B. Asterisk will calls both
parties
> >and do the tranfer automatically. In another words, asterisk is
resposible
> >to 'press' the '#' to do the transfer. I don't this can be achieve in the
> >extension.conf not matter how you structure you dial plan.
> >
> >Perhaps, the only way is to write a apps and plug it into asterisk like
all
> >the asterisk modules such as Meetme.
> >
> >Any ideas?
> >
> >
> >Foong
> >
> >- Original Message -
> >From: "Dan" <[EMAIL PROTECTED]>
> >To: <[EMAIL PROTECTED]>
> >Sent: Wednesday, July 30, 2003 3:42 PM
> >Subject: Re: [Asterisk-Users] Call Transfer
> >
> >
> >> Hi,
> >>
> >> It works if you put the 'T' switch in the dial line.
> >>
> >> You can then transfer the call from the caller.
> >> I have tested it in the folllowing configuration and it works:
> >> Call from a Cisco 7960 to an ATA 186.
> >> Select 'Transfer" on 7960
> >> Call another extension (X-Lite)
> >> Select again transfer on 7960.
> >> The call remain between ATA and X-Lite.
> >>
> >> This is what you need?
> >>
> >> BR,
> >> Dan
> >>
> >> - Original Message -
> >> From: "Chee Foong" <[EMAIL PROTECTED]>
> >> To: <[EMAIL PROTECTED]>
> >> Sent: Wednesday, July 30, 2003 7:08 AM
> >> Subject: [Asterisk-Users] Call Transfer
> >>
> >>
> >> Hello all,
> >>
> >> I am in a situation where I need to use asterisk to call someone say
> >Party
> >> A. After the call to Party A got through, asterisk will put Party A on
> >hold,
> >> then asterisk will call Party B. If call to Party B got through,
asterisk
> >> will transfer Party A to Party B.
> >>
> >> I wonder if this features is implemented into asterisk. I have found a
> >post
> >> in asterisk mailing list:
> >> http://lists.digium.com/pipermail/asterisk-users/2003-June/013253.html
> >>
> >> but that doesn't help much.
> >>
> >> If this features is not implemented, can anyone give me some point on
how
> >to
> >> implement this in asterisk? Do I need to write an app like the Dial
apps
> >for
> >> asterisk to load at start up?
> >>
> >>
> >> thanks
> >>
> >> Foong
> >>
> >>
> >> ___
> >> Asterisk-Users mailing list
> >> [EMAIL PROTECTED]
> >> http://lists.digium.com/mailman/listinfo/asterisk-users
> >>
> >
> >___
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>
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Re: [Asterisk-Users] Call Transfer, Budgettone 100

[Brian West]

I was told in #asterisk that you just hit transfer, dial the extension,
speak to caller and press transfer once your done talking and it should do
it.   In addition you can do transfer+extension+transfer+hangup...

Thats how I was told it would work.

bkw

On Wed, 30 Jul 2003, denon wrote:

> Last I checked, SIP transfer to park doesn't work .. only way to do it is
> using T and a # transfer .. which is ugly.  Has this been fixed?
>
> -d
>
> At 10:51 AM 7/30/2003 +0200, you wrote:
> >park the call
> >
> >On Wednesday 30 July 2003 11:16 am, [EMAIL PROTECTED] wrote:
> > > hi,
> > > can someone who has used Budgettone phones tell me how to do the
> > > following:
> > >
> > > an incoming call comes in and is answered by the receptionist.
> > > she need to put the call on hold, speak to whoever the call is for,
> > > and either (after that) pass on the call, otherwise speak again to
> > > whoever was on the call and hang up ..
> > >
> > > so far i've got as far as a blind transfer by pressing transfer button
> > > and then the new extension ..
> > >
> > > cheers
> > > Dave
> > > ---
> > > Email sent using AnyEmail (http://netbula.com/anyemail/)
> > > Netbula LLC is not responsible for the content of this email
> > >
> > > ___
> > > Asterisk-Users mailing list
> > > [EMAIL PROTECTED]
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >___
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Re: [Asterisk-Users] Call Transfer

[Chee Foong]

Dan,
The time to call could be stored into database with party A and party B
phone number.

Asterisk or perhaps a script (mentions by Andy Powel in another reply) just
keep checking the database and make calls if time is < current time and the
call has not been processed yet.

In this manner, the caller can even schedule a call for tomorrow mornnig,
all he do is just insert a record in database and wait :).

Foong


- Original Message -
From: "Dan" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, July 30, 2003 7:15 PM
Subject: Re: [Asterisk-Users] Call Transfer


> Foong,
>
> > Actually, we have a client that is too lazy to do all the dialing, he
want
> a
> > system that will call him and also the person he wanted to call, just
like
> > some receptionists do theese days. The different is that asterisk is
> taking
> > over the receptionist's job
> ... then... who decide when the call must be initiated and how?
>
> Dan
>
>
> ___
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Re: [Asterisk-Users] Call Transfer

[Chee Foong]

Thanks Andy

Will try that

Thanks again.

Foong
- Original Message -
From: "Andy Powell" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, July 30, 2003 6:56 PM
Subject: Re: [Asterisk-Users] Call Transfer


> Foong
>
> Take a look at the sample.call file, modifying the settings in there and
copying the file to /var/spool/asterisk/outgoing will cause asterisk to dial
the call.. an example config is below
>
> Channel: SIP/[EMAIL PROTECTED]
> MaxRetries: 2
> RetryTime: 60
> WaitTime: 30
> Context: mysipcontext2
> Extension: 2000
> Priority: 1
>
> This will make asterisk dial exten 1000 in the context mysipcontext when
it's answered it will then call exten 2000 in mysipcontext2..
>
> All you need is a script to lookup in the database and generate the script
file for you and it's done.
>
> HTH
>
> Andy
>
>
> *** REPLY SEPARATOR  ***
>
> On 30/07/2003 at 16:30 Chee Foong wrote:
>
> >Hello Dan,
> >
> >Thanks for you reply.
> >
> >Base on you recomendation using the 'T' argument. I manage to do call
> >transfer an it works really well.
> >
> >My problem comes when my boss comes out with a superb idea where the
> >transfering process is automated without involving a human :(
> >
> >Say asterisk get 2 numbers (from database, text file, etc), one belongs
> >party A and the other belongs to party B. Asterisk will calls both
parties
> >and do the tranfer automatically. In another words, asterisk is
resposible
> >to 'press' the '#' to do the transfer. I don't this can be achieve in the
> >extension.conf not matter how you structure you dial plan.
> >
> >Perhaps, the only way is to write a apps and plug it into asterisk like
all
> >the asterisk modules such as Meetme.
> >
> >Any ideas?
> >
> >
> >Foong
> >
> >- Original Message -
> >From: "Dan" <[EMAIL PROTECTED]>
> >To: <[EMAIL PROTECTED]>
> >Sent: Wednesday, July 30, 2003 3:42 PM
> >Subject: Re: [Asterisk-Users] Call Transfer
> >
> >
> >> Hi,
> >>
> >> It works if you put the 'T' switch in the dial line.
> >>
> >> You can then transfer the call from the caller.
> >> I have tested it in the folllowing configuration and it works:
> >> Call from a Cisco 7960 to an ATA 186.
> >> Select 'Transfer" on 7960
> >> Call another extension (X-Lite)
> >> Select again transfer on 7960.
> >> The call remain between ATA and X-Lite.
> >>
> >> This is what you need?
> >>
> >> BR,
> >> Dan
> >>
> >> - Original Message -
> >> From: "Chee Foong" <[EMAIL PROTECTED]>
> >> To: <[EMAIL PROTECTED]>
> >> Sent: Wednesday, July 30, 2003 7:08 AM
> >> Subject: [Asterisk-Users] Call Transfer
> >>
> >>
> >> Hello all,
> >>
> >> I am in a situation where I need to use asterisk to call someone say
> >Party
> >> A. After the call to Party A got through, asterisk will put Party A on
> >hold,
> >> then asterisk will call Party B. If call to Party B got through,
asterisk
> >> will transfer Party A to Party B.
> >>
> >> I wonder if this features is implemented into asterisk. I have found a
> >post
> >> in asterisk mailing list:
> >> http://lists.digium.com/pipermail/asterisk-users/2003-June/013253.html
> >>
> >> but that doesn't help much.
> >>
> >> If this features is not implemented, can anyone give me some point on
how
> >to
> >> implement this in asterisk? Do I need to write an app like the Dial
apps
> >for
> >> asterisk to load at start up?
> >>
> >>
> >> thanks
> >>
> >> Foong
> >>
> >>
> >> ___
> >> Asterisk-Users mailing list
> >> [EMAIL PROTECTED]
> >> http://lists.digium.com/mailman/listinfo/asterisk-users
> >>
> >
> >___
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> >[EMAIL PROTECTED]
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>
>
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>

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Re: [Asterisk-Users] Call Transfer

[Dan]

There is no need to create a Meeting Room... just to initiate a conference
in three...

- Original Message - 
From: "Chee Foong" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, July 30, 2003 1:02 PM
Subject: Re: [Asterisk-Users] Call Transfer


> Hello
>
> But If i do that I have to create lots of conference room if I have lots
of
> caller.
>
> Foong
>
> - Original Message -
> From: "Sip Rtp" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Wednesday, July 30, 2003 5:44 PM
> Subject: Re: [Asterisk-Users] Call Transfer
>
>
> > Yes, I second to that idea.
> > I think thats only available option to put them in a
> > local conference.
> > Rgds
> > Manoj K Gupta
> >
> > - Original Message -----
> > From: "Dan" <[EMAIL PROTECTED]>
> > To: <[EMAIL PROTECTED]>
> > Sent: Wednesday, July 30, 2003 2:04 PM
> > Subject: Re: [Asterisk-Users] Call Transfer
> >
> >
> > > Hi Foong,
> > >
> > > But then... who and when will trigger the transfer
> > between the two remote
> > > extensions?
> > >
> > > I think to something like that.
> > > One of the extension calls a special number,
> > entering a password (or check
> > > after the Caller ID).
> > > Asterisk close the call, wait for answer
> > > Call the second extension, wait for answer
> > > Then, in some way (eventually through a conference
> > mode using local
> > CONSOLE
> > > as master) bridge the two calls.
> > > What do you think about that?
> > >
> > > Dan
> > >
> > >
> > > - Original Message -
> > > From: "Chee Foong" <[EMAIL PROTECTED]>
> > > To: <[EMAIL PROTECTED]>
> > > Sent: Wednesday, July 30, 2003 11:30 AM
> > > Subject: Re: [Asterisk-Users] Call Transfer
> > >
> > >
> > > > Hello Dan,
> > > >
> > > > Thanks for you reply.
> > > >
> > > > Base on you recomendation using the 'T' argument.
> > I manage to do call
> > > > transfer an it works really well.
> > > >
> > > > My problem comes when my boss comes out with a
> > superb idea where the
> > > > transfering process is automated without involving
> > a human :(
> > > >
> > > > Say asterisk get 2 numbers (from database, text
> > file, etc), one belongs
> > > > party A and the other belongs to party B. Asterisk
> > will calls both
> > parties
> > > > and do the tranfer automatically. In another
> > words, asterisk is
> > resposible
> > > > to 'press' the '#' to do the transfer. I don't
> > this can be achieve in
> > the
> > > > extension.conf not matter how you structure you
> > dial plan.
> > > >
> > > > Perhaps, the only way is to write a apps and plug
> > it into asterisk like
> > > all
> > > > the asterisk modules such as Meetme.
> > > >
> > > > Any ideas?
> > > >
> > > >
> > > > Foong
> > > >
> > > > - Original Message -
> > > > From: "Dan" <[EMAIL PROTECTED]>
> > > > To: <[EMAIL PROTECTED]>
> > > > Sent: Wednesday, July 30, 2003 3:42 PM
> > > > Subject: Re: [Asterisk-Users] Call Transfer
> > > >
> > > >
> > > > > Hi,
> > > > >
> > > > > It works if you put the 'T' switch in the dial
> > line.
> > > > >
> > > > > You can then transfer the call from the caller.
> > > > > I have tested it in the folllowing configuration
> > and it works:
> > > > > Call from a Cisco 7960 to an ATA 186.
> > > > > Select 'Transfer" on 7960
> > > > > Call another extension (X-Lite)
> > > > > Select again transfer on 7960.
> > > > > The call remain between ATA and X-Lite.
> > > > >
> > > > > This is what you need?
> > > > >
> > > > > BR,
> > > > > Dan
> > > > >
> > > > > - Original Message -
> > > > > From: "Chee Foong" <[EMAIL PROTECTED]>
> > > > > To: <[EMAIL PROTECTED]>
> > > > > Sent: Wednesday, July 30, 2003 7:08 AM
> > > > > Subject: [Asteris

Re: [Asterisk-Users] Call Transfer

[Dan]

Foong,

> Actually, we have a client that is too lazy to do all the dialing, he want
a
> system that will call him and also the person he wanted to call, just like
> some receptionists do theese days. The different is that asterisk is
taking
> over the receptionist's job
... then... who decide when the call must be initiated and how?

Dan


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Re: [Asterisk-Users] Call Transfer

[Andy Powell]

Foong

Take a look at the sample.call file, modifying the settings in there and copying the 
file to /var/spool/asterisk/outgoing will cause asterisk to dial the call.. an example 
config is below

Channel: SIP/[EMAIL PROTECTED]
MaxRetries: 2
RetryTime: 60
WaitTime: 30
Context: mysipcontext2
Extension: 2000
Priority: 1

This will make asterisk dial exten 1000 in the context mysipcontext when it's answered 
it will then call exten 2000 in mysipcontext2..

All you need is a script to lookup in the database and generate the script file for 
you and it's done.

HTH

Andy


*** REPLY SEPARATOR  ***

On 30/07/2003 at 16:30 Chee Foong wrote:

>Hello Dan,
>
>Thanks for you reply.
>
>Base on you recomendation using the 'T' argument. I manage to do call
>transfer an it works really well.
>
>My problem comes when my boss comes out with a superb idea where the
>transfering process is automated without involving a human :(
>
>Say asterisk get 2 numbers (from database, text file, etc), one belongs
>party A and the other belongs to party B. Asterisk will calls both parties
>and do the tranfer automatically. In another words, asterisk is resposible
>to 'press' the '#' to do the transfer. I don't this can be achieve in the
>extension.conf not matter how you structure you dial plan.
>
>Perhaps, the only way is to write a apps and plug it into asterisk like all
>the asterisk modules such as Meetme.
>
>Any ideas?
>
>
>Foong
>
>- Original Message -
>From: "Dan" <[EMAIL PROTECTED]>
>To: <[EMAIL PROTECTED]>
>Sent: Wednesday, July 30, 2003 3:42 PM
>Subject: Re: [Asterisk-Users] Call Transfer
>
>
>> Hi,
>>
>> It works if you put the 'T' switch in the dial line.
>>
>> You can then transfer the call from the caller.
>> I have tested it in the folllowing configuration and it works:
>> Call from a Cisco 7960 to an ATA 186.
>> Select 'Transfer" on 7960
>> Call another extension (X-Lite)
>> Select again transfer on 7960.
>> The call remain between ATA and X-Lite.
>>
>> This is what you need?
>>
>> BR,
>> Dan
>>
>> - Original Message -
>> From: "Chee Foong" <[EMAIL PROTECTED]>
>> To: <[EMAIL PROTECTED]>
>> Sent: Wednesday, July 30, 2003 7:08 AM
>> Subject: [Asterisk-Users] Call Transfer
>>
>>
>> Hello all,
>>
>> I am in a situation where I need to use asterisk to call someone say
>Party
>> A. After the call to Party A got through, asterisk will put Party A on
>hold,
>> then asterisk will call Party B. If call to Party B got through, asterisk
>> will transfer Party A to Party B.
>>
>> I wonder if this features is implemented into asterisk. I have found a
>post
>> in asterisk mailing list:
>> http://lists.digium.com/pipermail/asterisk-users/2003-June/013253.html
>>
>> but that doesn't help much.
>>
>> If this features is not implemented, can anyone give me some point on how
>to
>> implement this in asterisk? Do I need to write an app like the Dial apps
>for
>> asterisk to load at start up?
>>
>>
>> thanks
>>
>> Foong
>>
>>
>> ___
>> Asterisk-Users mailing list
>> [EMAIL PROTECTED]
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>___
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Re: [Asterisk-Users] Call Transfer

[Chee Foong]

Hi Sip,

I achieve that by adding the following extension into extension.conf:

exten => _9,1,Dial(H323/{EXTEN:1})

foong

- Original Message - 
From: "Sip Rtp" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, July 30, 2003 5:45 PM
Subject: Re: [Asterisk-Users] Call Transfer


> Hi
> I would like to further ask if it is possible to
> transfer a call from
> openphone to pstn. i.e. i use openphone and asterisk
> -oh323 channel driver
> to make a call to a PSTN number through zap channel
> connected on that
> end.Then i wanna transfer that PSTN number to some
> other openphone
> extension/alias
> May i have a look at your extension to conf, as i am
> not clear with how to
> implement this.
> 
> Rgds
> Manoj k Gupta
> 
> 
> 
> 
> - Original Message -
> From: "Chee Foong" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Wednesday, July 30, 2003 2:00 PM
> Subject: Re: [Asterisk-Users] Call Transfer
> 
> 
> > Hello Dan,
> >
> > Thanks for you reply.
> >
> > Base on you recomendation using the 'T' argument. I
> manage to do call
> > transfer an it works really well.
> >
> > My problem comes when my boss comes out with a
> superb idea where the
> > transfering process is automated without involving a
> human :(
> >
> > Say asterisk get 2 numbers (from database, text
> file, etc), one belongs
> > party A and the other belongs to party B. Asterisk
> will calls both parties
> > and do the tranfer automatically. In another words,
> asterisk is resposible
> > to 'press' the '#' to do the transfer. I don't this
> can be achieve in the
> > extension.conf not matter how you structure you dial
> plan.
> >
> > Perhaps, the only way is to write a apps and plug it
> into asterisk like
> all
> > the asterisk modules such as Meetme.
> >
> > Any ideas?
> >
> >
> > Foong
> >
> > - Original Message -
> > From: "Dan" <[EMAIL PROTECTED]>
> > To: <[EMAIL PROTECTED]>
> > Sent: Wednesday, July 30, 2003 3:42 PM
> > Subject: Re: [Asterisk-Users] Call Transfer
> >
> >
> > > Hi,
> > >
> > > It works if you put the 'T' switch in the dial
> line.
> > >
> > > You can then transfer the call from the caller.
> > > I have tested it in the folllowing configuration
> and it works:
> > > Call from a Cisco 7960 to an ATA 186.
> > > Select 'Transfer" on 7960
> > > Call another extension (X-Lite)
> > > Select again transfer on 7960.
> > > The call remain between ATA and X-Lite.
> > >
> > > This is what you need?
> > >
> > > BR,
> > > Dan
> > >
> > > - Original Message -
> > > From: "Chee Foong" <[EMAIL PROTECTED]>
> > > To: <[EMAIL PROTECTED]>
> > > Sent: Wednesday, July 30, 2003 7:08 AM
> > > Subject: [Asterisk-Users] Call Transfer
> > >
> > >
> > > Hello all,
> > >
> > > I am in a situation where I need to use asterisk
> to call someone say
> Party
> > > A. After the call to Party A got through, asterisk
> will put Party A on
> > hold,
> > > then asterisk will call Party B. If call to Party
> B got through,
> asterisk
> > > will transfer Party A to Party B.
> > >
> > > I wonder if this features is implemented into
> asterisk. I have found a
> > post
> > > in asterisk mailing list:
> > >
> http://lists.digium.com/pipermail/asterisk-users/2003-June/013253.html
> > >
> > > but that doesn't help much.
> > >
> > > If this features is not implemented, can anyone
> give me some point on
> how
> > to
> > > implement this in asterisk? Do I need to write an
> app like the Dial apps
> > for
> > > asterisk to load at start up?
> > >
> > >
> > > thanks
> > >
> > > Foong
> > >
> > >
> > > ___
> > > Asterisk-Users mailing list
> > > [EMAIL PROTECTED]
> > >
> http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> >
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> >
> http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> 
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Re: [Asterisk-Users] Call Transfer

[Chee Foong]

Hello

But If i do that I have to create lots of conference room if I have lots of
caller.

Foong

- Original Message -
From: "Sip Rtp" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, July 30, 2003 5:44 PM
Subject: Re: [Asterisk-Users] Call Transfer


> Yes, I second to that idea.
> I think thats only available option to put them in a
> local conference.
> Rgds
> Manoj K Gupta
>
> - Original Message -
> From: "Dan" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Wednesday, July 30, 2003 2:04 PM
> Subject: Re: [Asterisk-Users] Call Transfer
>
>
> > Hi Foong,
> >
> > But then... who and when will trigger the transfer
> between the two remote
> > extensions?
> >
> > I think to something like that.
> > One of the extension calls a special number,
> entering a password (or check
> > after the Caller ID).
> > Asterisk close the call, wait for answer
> > Call the second extension, wait for answer
> > Then, in some way (eventually through a conference
> mode using local
> CONSOLE
> > as master) bridge the two calls.
> > What do you think about that?
> >
> > Dan
> >
> >
> > - Original Message -
> > From: "Chee Foong" <[EMAIL PROTECTED]>
> > To: <[EMAIL PROTECTED]>
> > Sent: Wednesday, July 30, 2003 11:30 AM
> > Subject: Re: [Asterisk-Users] Call Transfer
> >
> >
> > > Hello Dan,
> > >
> > > Thanks for you reply.
> > >
> > > Base on you recomendation using the 'T' argument.
> I manage to do call
> > > transfer an it works really well.
> > >
> > > My problem comes when my boss comes out with a
> superb idea where the
> > > transfering process is automated without involving
> a human :(
> > >
> > > Say asterisk get 2 numbers (from database, text
> file, etc), one belongs
> > > party A and the other belongs to party B. Asterisk
> will calls both
> parties
> > > and do the tranfer automatically. In another
> words, asterisk is
> resposible
> > > to 'press' the '#' to do the transfer. I don't
> this can be achieve in
> the
> > > extension.conf not matter how you structure you
> dial plan.
> > >
> > > Perhaps, the only way is to write a apps and plug
> it into asterisk like
> > all
> > > the asterisk modules such as Meetme.
> > >
> > > Any ideas?
> > >
> > >
> > > Foong
> > >
> > > - Original Message -
> > > From: "Dan" <[EMAIL PROTECTED]>
> > > To: <[EMAIL PROTECTED]>
> > > Sent: Wednesday, July 30, 2003 3:42 PM
> > > Subject: Re: [Asterisk-Users] Call Transfer
> > >
> > >
> > > > Hi,
> > > >
> > > > It works if you put the 'T' switch in the dial
> line.
> > > >
> > > > You can then transfer the call from the caller.
> > > > I have tested it in the folllowing configuration
> and it works:
> > > > Call from a Cisco 7960 to an ATA 186.
> > > > Select 'Transfer" on 7960
> > > > Call another extension (X-Lite)
> > > > Select again transfer on 7960.
> > > > The call remain between ATA and X-Lite.
> > > >
> > > > This is what you need?
> > > >
> > > > BR,
> > > > Dan
> > > >
> > > > - Original Message -
> > > > From: "Chee Foong" <[EMAIL PROTECTED]>
> > > > To: <[EMAIL PROTECTED]>
> > > > Sent: Wednesday, July 30, 2003 7:08 AM
> > > > Subject: [Asterisk-Users] Call Transfer
> > > >
> > > >
> > > > Hello all,
> > > >
> > > > I am in a situation where I need to use asterisk
> to call someone say
> > Party
> > > > A. After the call to Party A got through,
> asterisk will put Party A on
> > > hold,
> > > > then asterisk will call Party B. If call to
> Party B got through,
> > asterisk
> > > > will transfer Party A to Party B.
> > > >
> > > > I wonder if this features is implemented into
> asterisk. I have found a
> > > post
> > > > in asterisk mailing list:
> > > >
> http://lists.digium.com/pipermail/asterisk-users/2003-June/013253.html
> > > >
> > > > but that doesn't help much.
> > &g

Re: [Asterisk-Users] Call Transfer

[Sip Rtp]

Hi
I would like to further ask if it is possible to
transfer a call from
openphone to pstn. i.e. i use openphone and asterisk
-oh323 channel driver
to make a call to a PSTN number through zap channel
connected on that
end.Then i wanna transfer that PSTN number to some
other openphone
extension/alias
May i have a look at your extension to conf, as i am
not clear with how to
implement this.

Rgds
Manoj k Gupta




- Original Message -
From: "Chee Foong" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, July 30, 2003 2:00 PM
Subject: Re: [Asterisk-Users] Call Transfer


> Hello Dan,
>
> Thanks for you reply.
>
> Base on you recomendation using the 'T' argument. I
manage to do call
> transfer an it works really well.
>
> My problem comes when my boss comes out with a
superb idea where the
> transfering process is automated without involving a
human :(
>
> Say asterisk get 2 numbers (from database, text
file, etc), one belongs
> party A and the other belongs to party B. Asterisk
will calls both parties
> and do the tranfer automatically. In another words,
asterisk is resposible
> to 'press' the '#' to do the transfer. I don't this
can be achieve in the
> extension.conf not matter how you structure you dial
plan.
>
> Perhaps, the only way is to write a apps and plug it
into asterisk like
all
> the asterisk modules such as Meetme.
>
> Any ideas?
>
>
> Foong
>
> - Original Message -
> From: "Dan" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Wednesday, July 30, 2003 3:42 PM
> Subject: Re: [Asterisk-Users] Call Transfer
>
>
> > Hi,
> >
> > It works if you put the 'T' switch in the dial
line.
> >
> > You can then transfer the call from the caller.
> > I have tested it in the folllowing configuration
and it works:
> > Call from a Cisco 7960 to an ATA 186.
> > Select 'Transfer" on 7960
> > Call another extension (X-Lite)
> > Select again transfer on 7960.
> > The call remain between ATA and X-Lite.
> >
> > This is what you need?
> >
> > BR,
> > Dan
> >
> > - Original Message -
> > From: "Chee Foong" <[EMAIL PROTECTED]>
> > To: <[EMAIL PROTECTED]>
> > Sent: Wednesday, July 30, 2003 7:08 AM
> > Subject: [Asterisk-Users] Call Transfer
> >
> >
> > Hello all,
> >
> > I am in a situation where I need to use asterisk
to call someone say
Party
> > A. After the call to Party A got through, asterisk
will put Party A on
> hold,
> > then asterisk will call Party B. If call to Party
B got through,
asterisk
> > will transfer Party A to Party B.
> >
> > I wonder if this features is implemented into
asterisk. I have found a
> post
> > in asterisk mailing list:
> >
http://lists.digium.com/pipermail/asterisk-users/2003-June/013253.html
> >
> > but that doesn't help much.
> >
> > If this features is not implemented, can anyone
give me some point on
how
> to
> > implement this in asterisk? Do I need to write an
app like the Dial apps
> for
> > asterisk to load at start up?
> >
> >
> > thanks
> >
> > Foong
> >
> >
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> >
http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
>
http://lists.digium.com/mailman/listinfo/asterisk-users


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Re: [Asterisk-Users] Call Transfer

[Sip Rtp]

Yes, I second to that idea.
I think thats only available option to put them in a
local conference.
Rgds
Manoj K Gupta

- Original Message -
From: "Dan" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, July 30, 2003 2:04 PM
Subject: Re: [Asterisk-Users] Call Transfer


> Hi Foong,
>
> But then... who and when will trigger the transfer
between the two remote
> extensions?
>
> I think to something like that.
> One of the extension calls a special number,
entering a password (or check
> after the Caller ID).
> Asterisk close the call, wait for answer
> Call the second extension, wait for answer
> Then, in some way (eventually through a conference
mode using local
CONSOLE
> as master) bridge the two calls.
> What do you think about that?
>
> Dan
>
>
> - Original Message -
> From: "Chee Foong" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Wednesday, July 30, 2003 11:30 AM
> Subject: Re: [Asterisk-Users] Call Transfer
>
>
> > Hello Dan,
> >
> > Thanks for you reply.
> >
> > Base on you recomendation using the 'T' argument.
I manage to do call
> > transfer an it works really well.
> >
> > My problem comes when my boss comes out with a
superb idea where the
> > transfering process is automated without involving
a human :(
> >
> > Say asterisk get 2 numbers (from database, text
file, etc), one belongs
> > party A and the other belongs to party B. Asterisk
will calls both
parties
> > and do the tranfer automatically. In another
words, asterisk is
resposible
> > to 'press' the '#' to do the transfer. I don't
this can be achieve in
the
> > extension.conf not matter how you structure you
dial plan.
> >
> > Perhaps, the only way is to write a apps and plug
it into asterisk like
> all
> > the asterisk modules such as Meetme.
> >
> > Any ideas?
> >
> >
> > Foong
> >
> > - Original Message -
> > From: "Dan" <[EMAIL PROTECTED]>
> > To: <[EMAIL PROTECTED]>
> > Sent: Wednesday, July 30, 2003 3:42 PM
> > Subject: Re: [Asterisk-Users] Call Transfer
> >
> >
> > > Hi,
> > >
> > > It works if you put the 'T' switch in the dial
line.
> > >
> > > You can then transfer the call from the caller.
> > > I have tested it in the folllowing configuration
and it works:
> > > Call from a Cisco 7960 to an ATA 186.
> > > Select 'Transfer" on 7960
> > > Call another extension (X-Lite)
> > > Select again transfer on 7960.
> > > The call remain between ATA and X-Lite.
> > >
> > > This is what you need?
> > >
> > > BR,
> > > Dan
> > >
> > > - Original Message -
> > > From: "Chee Foong" <[EMAIL PROTECTED]>
> > > To: <[EMAIL PROTECTED]>
> > > Sent: Wednesday, July 30, 2003 7:08 AM
> > > Subject: [Asterisk-Users] Call Transfer
> > >
> > >
> > > Hello all,
> > >
> > > I am in a situation where I need to use asterisk
to call someone say
> Party
> > > A. After the call to Party A got through,
asterisk will put Party A on
> > hold,
> > > then asterisk will call Party B. If call to
Party B got through,
> asterisk
> > > will transfer Party A to Party B.
> > >
> > > I wonder if this features is implemented into
asterisk. I have found a
> > post
> > > in asterisk mailing list:
> > >
http://lists.digium.com/pipermail/asterisk-users/2003-June/013253.html
> > >
> > > but that doesn't help much.
> > >
> > > If this features is not implemented, can anyone
give me some point on
> how
> > to
> > > implement this in asterisk? Do I need to write
an app like the Dial
apps
> > for
> > > asterisk to load at start up?
> > >
> > >
> > > thanks
> > >
> > > Foong
> > >
> > >
> > > ___
> > > Asterisk-Users mailing list
> > > [EMAIL PROTECTED]
> > >
http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> >
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> >
http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
>
>
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
>
http://lists.digium.com/mailman/listinfo/asterisk-users


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Re: [Asterisk-Users] Call Transfer, Budgettone 100

[denon]

Last I checked, SIP transfer to park doesn't work .. only way to do it is 
using T and a # transfer .. which is ugly.  Has this been fixed?

-d

At 10:51 AM 7/30/2003 +0200, you wrote:
park the call

On Wednesday 30 July 2003 11:16 am, [EMAIL PROTECTED] wrote:
> hi,
> can someone who has used Budgettone phones tell me how to do the
> following:
>
> an incoming call comes in and is answered by the receptionist.
> she need to put the call on hold, speak to whoever the call is for,
> and either (after that) pass on the call, otherwise speak again to
> whoever was on the call and hang up ..
>
> so far i've got as far as a blind transfer by pressing transfer button
> and then the new extension ..
>
> cheers
> Dave
> ---
> Email sent using AnyEmail (http://netbula.com/anyemail/)
> Netbula LLC is not responsible for the content of this email
>
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Re: [Asterisk-Users] Call Transfer, Budgettone 100

[Michael Bielicki]

park the call

On Wednesday 30 July 2003 11:16 am, [EMAIL PROTECTED] wrote:
> hi,
> can someone who has used Budgettone phones tell me how to do the
> following:
>
> an incoming call comes in and is answered by the receptionist.
> she need to put the call on hold, speak to whoever the call is for,
> and either (after that) pass on the call, otherwise speak again to
> whoever was on the call and hang up ..
>
> so far i've got as far as a blind transfer by pressing transfer button
> and then the new extension ..
>
> cheers
> Dave
> ---
> Email sent using AnyEmail (http://netbula.com/anyemail/)
> Netbula LLC is not responsible for the content of this email
>
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Re: [Asterisk-Users] Call Transfer

[Chee Foong]

Dan,

Asterisk is suppose to trigger the transfer when it successfully call both
extensions
Do you mean I have to create conference room for every call? that would not
be practicle.

Or do you have a example dialplan to to illustrate you suggestion?

Actually, we have a client that is too lazy to do all the dialing, he want a
system that will call him and also the person he wanted to call, just like
some receptionists do theese days. The different is that asterisk is taking
over the receptionist's job

thanks


Foong


> Hi Foong,
>
> But then... who and when will trigger the transfer between the two remote
> extensions?
>
> I think to something like that.
> One of the extension calls a special number, entering a password (or check
> after the Caller ID).
> Asterisk close the call, wait for answer
> Call the second extension, wait for answer
> Then, in some way (eventually through a conference mode using local
CONSOLE
> as master) bridge the two calls.
> What do you think about that?
>
> Dan
>
>
> - Original Message -
> From: "Chee Foong" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Wednesday, July 30, 2003 11:30 AM
> Subject: Re: [Asterisk-Users] Call Transfer
>
>
> > Hello Dan,
> >
> > Thanks for you reply.
> >
> > Base on you recomendation using the 'T' argument. I manage to do call
> > transfer an it works really well.
> >
> > My problem comes when my boss comes out with a superb idea where the
> > transfering process is automated without involving a human :(
> >
> > Say asterisk get 2 numbers (from database, text file, etc), one belongs
> > party A and the other belongs to party B. Asterisk will calls both
parties
> > and do the tranfer automatically. In another words, asterisk is
resposible
> > to 'press' the '#' to do the transfer. I don't this can be achieve in
the
> > extension.conf not matter how you structure you dial plan.
> >
> > Perhaps, the only way is to write a apps and plug it into asterisk like
> all
> > the asterisk modules such as Meetme.
> >
> > Any ideas?
> >
> >
> > Foong
> >
> > - Original Message -
> > From: "Dan" <[EMAIL PROTECTED]>
> > To: <[EMAIL PROTECTED]>
> > Sent: Wednesday, July 30, 2003 3:42 PM
> > Subject: Re: [Asterisk-Users] Call Transfer
> >
> >
> > > Hi,
> > >
> > > It works if you put the 'T' switch in the dial line.
> > >
> > > You can then transfer the call from the caller.
> > > I have tested it in the folllowing configuration and it works:
> > > Call from a Cisco 7960 to an ATA 186.
> > > Select 'Transfer" on 7960
> > > Call another extension (X-Lite)
> > > Select again transfer on 7960.
> > > The call remain between ATA and X-Lite.
> > >
> > > This is what you need?
> > >
> > > BR,
> > > Dan
> > >
> > > - Original Message -
> > > From: "Chee Foong" <[EMAIL PROTECTED]>
> > > To: <[EMAIL PROTECTED]>
> > > Sent: Wednesday, July 30, 2003 7:08 AM
> > > Subject: [Asterisk-Users] Call Transfer
> > >
> > >
> > > Hello all,
> > >
> > > I am in a situation where I need to use asterisk to call someone say
> Party
> > > A. After the call to Party A got through, asterisk will put Party A on
> > hold,
> > > then asterisk will call Party B. If call to Party B got through,
> asterisk
> > > will transfer Party A to Party B.
> > >
> > > I wonder if this features is implemented into asterisk. I have found a
> > post
> > > in asterisk mailing list:
> > > http://lists.digium.com/pipermail/asterisk-users/2003-June/013253.html
> > >
> > > but that doesn't help much.
> > >
> > > If this features is not implemented, can anyone give me some point on
> how
> > to
> > > implement this in asterisk? Do I need to write an app like the Dial
apps
> > for
> > > asterisk to load at start up?
> > >
> > >
> > > thanks
> > >
> > > Foong
> > >
> > >
> > > ___
> > > Asterisk-Users mailing list
> > > [EMAIL PROTECTED]
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> >
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
>
>
> ___
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>

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