--On Friday, March 14, 2003 9:40 AM -0600 Martin Pycko <[EMAIL PROTECTED]> 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 <[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' > > > 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 > >> > >> > > > > > > > > -- > > ********************************************************************** > Nordic LAN&WAN Communication Oy > Pertti Pikkarainen > vp of engineering > E-Mail: [EMAIL PROTECTED] > tel: +358-9-5024100 > fax: +358-9-5023840 > gsm: +358-500-511467 > > Sinikalliontie 16 > 02630 Espoo > FINLAND > > ********************************************************************** > > > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users
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