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
_______________________________________________
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
