Going back to the subject, what does the grandstream really do, SIP-wise, when you press
the transfer button?
Olle,
The following is an exact transcription of the description given in the BT101 manual for Blind Transfers:
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.
Stephen R. Besch
_______________________________________________ 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
