[Asterisk-Users] How to transfer a call??

[Jim Archer]

I have T working here.

--On Friday, March 14, 2003 9:40 AM -0600 Martin Pycko <martinp at digium.com> 
wrote:

> Of courese:
> exten => 9998,1,Dial,SIP/9998|30|tTm
> Notice when you don't use the timeout you do have to use the options
> separator "|" like this:
> exten => 9998,1,Dial,SIP/9998||tTm
>
> but I think that T is not yet implemented
>
> regards
> Martin
>
> On Fri, 14 Mar 2003, WipeOut . wrote:
>
>> Thanks the 'show application dial' was useful..
>>
>> Can multiple options be specified?
>> eg. exten => 9998,1,Dial,SIP/9998|30|t|T
>>
>>
>>
>> ----- Original Message -----
>> From: Pertti Pikkarainen <ppik at lanwan.fi>
>> Date: Fri, 14 Mar 2003 15:15:14 +0200
>> To: asterisk-users at lists.digium.com
>> 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'
>> >
>> >
>> > WipeOut ? wrote:
>> >
>> > > What is the correct syntax to use the 't' option??
>> > >
>> > > This is the current line in my extensions.conf
>> > > 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 <ppik at lanwan.fi>
>> > > Date: Fri, 14 Mar 2003 13:50:21 +0200
>> > > To: asterisk-users at lists.digium.com
>> > > 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 option instead.
>> > >> But now I can't use *  call parking  because of that. Using  #  is
>> > >> probably better
>> > >> if there are no scaling problems.
>> > >>
>> > >> Regards Pertti
>> > >>
>> > >>
>> > >>
>> > >> Steven Critchfield wrote:
>> > >>
>> > >>
>> > >>
>> > >>> 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 ?= wrote:
>> > >>>
>> > >>>
>> > >>>
>> > >>>
>> > >>>> 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 the Handbook (ver 1) and the white paper
>> > >>>> PDF's.. and I have managed to setup 2 extentions and make calls
>> > >>>> between them using MSN Messenger, nothing fantastic but its a
>> > >>>> start..
>> > >>>>
>> > >>>> One answer is still missing.. How do I transfer a call to another
>> > >>>> ext?? I am looking at only using IP phones and so for the test
>> > >>>> system I am using MSN Messenger.. The final solution will
>> > >>>> probably use a linux softphone line gnophone or linphone..
>> > >>>>
>> > >>>> All I have been able to find in the docs about call transfer is
>> > >>>> using a normal phone handset and hook-flash (not quite sure what
>> > >>>> that it, I am new to telephony)..
>> > >>>>
>> > >>>> So I guess what I am asking is what do I need to configure or do
>> > >>>> to be able to transfer a call from one IP ext to another??
>> > >>>>
>> > >>>> Thanks..
>> > >>>>
>> > >>>>
>> > >>>>
>> > >>>>
>> > >> _______________________________________________
>> > >> Asterisk-Users mailing list
>> > >> Asterisk-Users at lists.digium.com
>> > >> http://lists.digium.com/mailman/listinfo/asterisk-users
>> > >>
>> > >>
>> > >
>> > >
>> > >
>> >
>> > --
>> >
>> > **********************************************************************
>> > Nordic LAN&WAN Communication Oy
>> > Pertti Pikkarainen
>> > vp of engineering
>> > E-Mail: ppik at lanwan.fi
>> > tel: +358-9-5024100
>> > fax: +358-9-5023840
>> > gsm: +358-500-511467
>> >
>> > Sinikalliontie 16
>> > 02630 Espoo
>> > FINLAND
>> >
>> > **********************************************************************
>> >
>> >
>> >
>> > _______________________________________________
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>> > http://lists.digium.com/mailman/listinfo/asterisk-users
>>
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