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

[WipeOut .]

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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