The set-up is like below:

1. Freeswitch is configured as UA with Asterisk TDM card (Openzap) 1 FXO + 1 FXS and registered with sipx.
2. SNOM300 registered with sipx.
3. Grandstream IP phone also registered with sipx.

Procedure followed for Blind Transfer:
1. Make a call from SNOM300 to Freeswitch FXS. Accept the call at Freeswitch FXS.
2. Using "transfer UUID -both Grandstream-number"  command at the Freeswitch, I tried Blind transfer.  On Ethereal, I saw INVITE with modified From-header.
     ---- Are we not using REFER for Call Transfer here???
3. Call gets transferred to Grandstream now . This looks fine.

Procedure followed for Attended Transfer:
1. Make a call from SNOM300 to Freeswitch FXS. Accept the call at Freeswitch FXS.
2. Now use FLASH Key on the analog phone connected to freeswitch FXS to put the established call on HOLD.
3. After getting dial tone, dial GRANDSTREAM number.
4. Accept the call at GRANDSTREAM. Now again using FLASH key, swap the call with SNOM300.
5. Use the command " transfer SNOM-UUID -bleg GRANDSTREAM-number"
6. Call is getting swapped back to GRANDSTREAM and the initial call with SNOM is getting disconnected.
Note: Default dial plan with entries for both SNOM300 and GRANDSTREAM is used.
Please let me know if I am going wrong somewhere.

Thanks and regards,
Irshad.





On Dec 27, 2007 7:51 PM, Brian West <brian.west@mac.com > wrote:
I have tested Attended and Blind transfers with these phones:

Polycom 650,
Snom 300, 320, 360 and 370
Linksys SPA962

You'll need to refer to your ip phones documentation on how to perform
an attended transfer.  If you can tell me which phone you're using I
might be able to find out.

/b


On Dec 27, 2007, at 1:07 AM, Irshad Ahamed wrote:

> Hi Mike,
>
> Can you explain me the procedure of successful Attended Call
> Transfer? It would be really of great help to me.
>
> Thanks n regards,
> Irshad.
>
> On Dec 20, 2007 6:40 PM, Michael Jerris < mike@jerris.com > wrote:
> This should work fine with current svn trunk, we were testing that
> last night and had no issues, if you are having a problem, could you
> better detail what you are doing on your phone and what the result is
> and post that and a sip trace on http://jira.freeswitch.org.
>
> Thanks
> Mike
>
> On Dec 20, 2007, at 7:21 AM, Irshad Ahamed wrote:
>
> > Hi,
> > I have installed Freeswitch successfully on my machine.
> > I was trying to perform an Attended Call transfer to other endpoint
> > in the same domain as a Transferor. But I couldn't find how exactly
> > I should do that. It would be really great if someone helps me out
> > in telling the procedures to follow.
> >
> > I have tried out Call Transfer as Transferee and it works fine.
> >
> > Thanks in Advance...
> >
> >
> > Best Regards,
> > Irshad.
> > _______________________________________________
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> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
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>
>
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