[Asterisk-Users] Re: Grandstream transfer into outer space

Matthew B Marlowe
Thu, 26 Feb 2004 15:04:07 -0000

Yes asterisk works with the transfer button



Sincerely,
Matthew Marlowe
Gear 3 Technologies, LLC
609.252.1155 x614
www.gear3.com

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-----Original Message-----
From: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Olle E.
Johansson
Sent: Thursday, February 26, 2004 4:36 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Re: Grandstream transfer into outer space

Stephen R. Besch wrote:

> Olle E. Johansson wrote:
>
>> Going back to the subject, what does the grandstream really do,
>> SIP-wise, when you press the transfer button?
>>
>
> 4.3.7 Call Transfer The user can transfer an active call to a third
> phone by using the "Transfer" button. The sequence is like this:
The
> user presses the "Transfer" button and if the other voice channel
is
> available (i.e., there is no other active conversation besides the
> current one), he/she will hear a dial tone. He/She can then dial the
3rd
> phone and then hangs up his own phone. 2 kinds of blind call transfers

> are supported: using REFER and using BYE/Also. The SIP message flow
> based on SIP REFER method looks something like this:
>
> Call Flow Diagram For Blind Call Transfer:
>
>  From Transferee to Transferor
>
>      INVITE ->
>     <-100/180/200
>      ACK ->
>     <- RTP Media ->
>     <- REFER
>      202 ->
>      NOTIFY ->
>     <- 200
>     <- BYE
>      200 ->
>
>  From Transferee to Recipient
>
>     INVITE ->
>     <-  100/180/200
>      ACK ->
>     <- RTP Media ->
>
>
> The SIP message flow based on BYE/Also  method looks something like
this:
>
>  From Transferee to Transferor
>
>      INVITE ->
>     <- 100/180/200
>     ACK ->
>     <- RTP Media ->
>     <- REFER
>     501 Not Implemented ->
>     <- BYE with "Also:"
>     200 ->
>
>  From Transferee to Recipient
>     INVITE ->
>     <- 100/180/200
>     ACK ->
>     <- RTP Media ->
>
> I have no idea if this is accurate, I just copied it and replaced the
> arrows indicating direction with "->" and "<-".
You can download the
> manual itself from the GS web site.
I'll do that.

Does Asterisk work with this transfer button or not? We have
implementation of both REFER
and BYE/also in the sip channel.

/O