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
-- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr
Powered by Outblaze _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
