Dear all
I have install asterisk 1.2.x and it is working fine my setup is
like
[*]---[Mediant2k][Avaya]
Now i want to transfer call in internal extension i have read more document on
www.voip-info.com but it is now so much clear so if u have any sample
configura
Hi All,
I am using asterisk 1.8.11 on centos 5. I have realtime sip peers with dtmf
setting rfc2833 and inband. I have also enabled blind and attended transfer
features in features.conf but still call transfers dont work. I have setup
transfer feature in past but i dont think i am missing anything
check to see if you have dtmf=rfc2833 and canreinvite=no in sip.conf general
settings.
On 7/4/07, satish patel <[EMAIL PROTECTED]> wrote:
Dear all
I have install asterisk 1.2.x and it is working fine my
setup is like
[*]---[Mediant2k][Avaya]
Now i want to trans
Hi.
Maybe you forgotten specify to allow the transferring a call.
Try with "tT" options in "Dial()" in extensions.conf.
// I don't know what's difference "t" and "T".
--
Takehiro Matsushima
2012/4/7 Rizwan Hisham :
> Hi All,
> I am using asterisk 1.8.11 on centos 5. I have realtime sip peers
On 9/4/12 3:04 am, Takehiro Matsushima wrote:
// I don't know what's difference "t" and "T".
T allows the caller to transfer. t allows the called user to transfer.
You very rarely want "Tt" - since I doubt you want an incoming caller to
be able to transfer their call all over the place. You u
Thank you so much.
OK, I understood that to transfer the call "t" is usually used, is it right?
And I should write so in my last mail.
"t" and "T" are described with same sentences in official wiki...
Regards,
Takehiro Matsushima
2012/4/9 Chris Bagnall :
> On 9/4/12 3:04 am, Takehiro Matsushi
Thanks everyone. I was using the Tt flag but in the wrong place in the dial
application.
Cheers
On Mon, Apr 9, 2012 at 4:54 PM, Takehiro Matsushima <
takehiro.dream...@gmail.com> wrote:
> Thank you so much.
>
> OK, I understood that to transfer the call "t" is usually used, is it
> right?
> And
