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

[Olle E. Johansson]

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