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

[Pertti Pikkarainen]

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.. 
>>    
>>