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..
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