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

[Martin Pycko]

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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