anyone has the 't' option working with current cvs? I can't make it work. but remeber it was working (don't know what cvs)
Matteo Il ven, 2003-03-14 alle 14:15, Pertti Pikkarainen ha scritto: > 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 <[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' 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 > >>[EMAIL PROTECTED] > >>http://lists.digium.com/mailman/listinfo/asterisk-users > >> > >> > > > > > > -- Matteo Brancaleoni <[EMAIL PROTECTED]> Espia - Emmegi Srl _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
