Hi,
Firstly let me start off by saying that asterisk is one of the most amazing pieces of
open source I have seen, it rates right up there with Apache, OpenOffice, MySQL and
even Linux itself.. Nice work!!
I have just installed my first server, thanks to the astinstall script.. and I have
read
If you search the archives you would find that for IP phone you need to
add a 't' option to the end of your dial command. The 't' option will
let the user dial '#' to get the systems attention, then dial an
extention for the transfer.
On Fri, 2003-03-14 at 03:32, =?iso-8859-1?B?V2lwZU91dCCZ ?= wro
Negative side effect with 't' option: all the local SIP-to-SIP media
streams travel trough Asterisk instead of going direct.
Right now I'm using SNOM's transfer option instead.
But now I can't use * call parking because of that. Using # is
probably better
if there are no scaling problems.
Re
Fri, 14 Mar 2003 13:50:21 +0200
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] How to transfer a call??
Negative side effect with 't' option: all the local SIP-to-SIP media
streams travel trough Asterisk instead of going direct.
Right now I'm using SNOM's transfer o
nge it to
exten => 9998,1,Dial,SIP/9998,t
Thanks.
- Original Message -
From: Pertti Pikkarainen <[EMAIL PROTECTED]>
Date: Fri, 14 Mar 2003 13:50:21 +0200
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] How to transfer a call??
Negative side effect with 't' op
t;So would I change it to
> >exten => 9998,1,Dial,SIP/9998,t
> >
> >Thanks.
> >
> >- Original Message -
> >From: Pertti Pikkarainen <[EMAIL PROTECTED]>
> >Date: Fri, 14 Mar 2003 13:50:21 +0200
> >To: [EMAIL PROTECTED]
> >Subject:
f
>exten => 9998,1,Dial,SIP/9998
>So would I change it to
>exten => 9998,1,Dial,SIP/9998,t
>
>Thanks.
>
>- Original Message -
>From: Pertti Pikkarainen <[EMAIL PROTECTED]>
>Date: Fri, 14 Mar 2003 13:50:21 +0200
>To: [EMAIL PROTECTED]
>Subject: Re
ten => 9998,1,Dial,SIP/9998,t
> > >
> > >Thanks.
> > >
> > >- Original Message -
> > >From: Pertti Pikkarainen <[EMAIL PROTECTED]>
> > >Date: Fri, 14 Mar 2003 13:50:21 +0200
> > >To: [EMAIL PROTECTED]
> > >Sub
rainen <[EMAIL PROTECTED]>
Date: Fri, 14 Mar 2003 15:15:14 +0200
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] How to transfer a call??
>
> I have it like this
>
> exten => 9998,1,Dial,SIP/9998|30|t
>
> 30 is a timeout value
> Check 'show application dial
>I have T working here.
Is your definition of T, that the caller can transfer ??
ie I P/U Zap 1 call Zap 2, & Now I press #701 & I can Park Zap2
If so is this a patch ??? pls post
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http://lists.digium.
I was petty sure that t and T worked from calls from one extension to
another. I did notice that the caller can not transfer a call that goes to
an outside line.
I can double check tomorrow.
--On Friday, March 14, 2003 11:45 AM -0800 TC <[EMAIL PROTECTED]> wrote:
I have T working here.
Is y
box.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lenz Emilitri
Sent: Friday, January 16, 2009 10:09 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to transfer a call from
digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Lenz Emilitri
> *Sent:* Friday, January 16, 2009 10:09 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] How to transfer a call from one
> AsteriskServer
erisk on the first box which kills the call.
>>>
>>>
>>>
>>> --------------
>>> *From:* asterisk-users-boun...@lists.digium.com [mailto:
>>> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Lenz Emilitri
>>> *Sent:
anything hit the second
>>> box.
>>>
>>>
>>>
>>> --
>>> *From:* asterisk-users-boun...@lists.digium.com [mailto:
>>> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Lenz Emilitri
>>> *Sent:* Friday, Jan
Hi,
I have xlite client registered with a user. Now i dial an extension say 1500
which
has the dial plan as follows.
exten==>1500,1,AGI("localhost//hello.agi")
So when i dial extenstion 1500 the script hello.agi is invoked which in turn
plays a welcome message. I now want to transfer the call now
Hi,
I would like to make a simple application with address book which
to dial the numbers and to transfer the call to the caller before the
called party is answered. How can I do that?
Regards,
Miro.
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[EMAIL
Can anyone tell me how I can completely move an established call off of one
Asterisk server to another?
In our case we have a server with our IVR. Depending upon digits entered,
the call can be transferred to any of our other servers depending where the
extension or queue reside.
We would like t
: Friday, January 16, 2009 12:17 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to transfer a call from one AsteriskServer
to another
Why don't you simply Dial() the call to a separate box keeping Asterisk out
of the audio path?
l.
2009/1/16
cial Discussion
> *Subject:* Re: [asterisk-users] How to transfer a call from one
> AsteriskServer to another
>
> Why don't you simply Dial() the call to a separate box keeping Asterisk out
> of the audio path?
>
> l.
>
> 2009/1/16 Paul
>
>> Can anyone tell
On Tue, Jul 27, 2010 at 05:42:01PM +0530, Janu Mukherjee wrote:
> Hi,
>
> I have xlite client registered with a user. Now i dial an extension say 1500
> which
> has the dial plan as follows.
> exten==>1500,1,AGI("localhost//hello.agi")
Obviously this is not the dialplan you have, as this would fa
El 27/07/10 07:12, Janu Mukherjee escribió:
>
> Hi,
>
> I have xlite client registered with a user. Now i dial an extension
> say 1500 which
> has the dial plan as follows.
> exten==>1500,1,AGI("localhost//hello.agi")
>
> So when i dial extenstion 1500 the script hello.agi is invoked which
> in t
Are you planning on connecting your two Asterisk servers with SIP or IAX?
Check out this tutorial if using SIP:
http://hostseries.com/connecting-to-asterisk-servers-via-sip/
You should be able to adapt it to your needs. Good luck!
Paul wrote:
> Can anyone tell me how I can completely move an es
Why don't you simply Dial() the call to a separate box keeping Asterisk out
of the audio path?
l.
2009/1/16 Paul
> Can anyone tell me how I can completely move an established call off of
> one Asterisk server to another?
>
> In our case we have a server with our IVR. Depending upon digits ent
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