The set-up is like below:<br><br>1. Freeswitch is configured as UA with Asterisk TDM card (Openzap) 1 FXO + 1 FXS and registered with sipx.<br>2. SNOM300 registered with sipx.<br>3. Grandstream IP phone also registered with sipx.
<br><br>Procedure followed for Blind Transfer:<br>1. Make a call from SNOM300 to Freeswitch FXS. Accept the call at Freeswitch FXS.<br>2. Using "transfer UUID -both Grandstream-number" command at the Freeswitch, I tried Blind transfer. On Ethereal, I saw INVITE with modified From-header.
<br> ---- Are we not using REFER for Call Transfer here???<br>3. Call gets transferred to Grandstream now . This looks fine.<br><br>Procedure followed for Attended Transfer:<br>1. Make a call from SNOM300 to Freeswitch FXS. Accept the call at Freeswitch FXS.
<br>2. Now use FLASH Key on the analog phone connected to freeswitch FXS to put the established call on HOLD. <br>3. After getting dial tone, dial GRANDSTREAM number.<br>4. Accept the call at GRANDSTREAM. Now again using FLASH key, swap the call with SNOM300.
<br>5. Use the command " transfer SNOM-UUID -bleg GRANDSTREAM-number"<br>6. Call is getting swapped back to GRANDSTREAM and the initial call with SNOM is getting disconnected.<br>Note: Default dial plan with entries for both SNOM300 and GRANDSTREAM is used.
<br>Please let me know if I am going wrong somewhere.<br><br>Thanks and regards,<br>Irshad.<br><br><br><br><br><br><div class="gmail_quote">On Dec 27, 2007 7:51 PM, Brian West <<a href="mailto:brian.west@mac.com">brian.west@mac.com
</a>> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">I have tested Attended and Blind transfers with these phones:<br><br>Polycom 650,
<br>Snom 300, 320, 360 and 370<br>Linksys SPA962<br><br>You'll need to refer to your ip phones documentation on how to perform<br>an attended transfer. If you can tell me which phone you're using I<br>might be able to find out.
<br><font color="#888888"><br>/b<br></font><div><div></div><div class="Wj3C7c"><br><br>On Dec 27, 2007, at 1:07 AM, Irshad Ahamed wrote:<br><br>> Hi Mike,<br>><br>> Can you explain me the procedure of successful Attended Call
<br>> Transfer? It would be really of great help to me.<br>><br>> Thanks n regards,<br>> Irshad.<br>><br>> On Dec 20, 2007 6:40 PM, Michael Jerris < <a href="mailto:mike@jerris.com">mike@jerris.com</a>
> wrote:<br>> This should work fine with current svn trunk, we were testing that<br>> last night and had no issues, if you are having a problem, could you<br>> better detail what you are doing on your phone and what the result is
<br>> and post that and a sip trace on <a href="http://jira.freeswitch.org" target="_blank">http://jira.freeswitch.org</a>.<br>><br>> Thanks<br>> Mike<br>><br>> On Dec 20, 2007, at 7:21 AM, Irshad Ahamed wrote:
<br>><br>> > Hi,<br>> > I have installed Freeswitch successfully on my machine.<br>> > I was trying to perform an Attended Call transfer to other endpoint<br>> > in the same domain as a Transferor. But I couldn't find how exactly
<br>> > I should do that. It would be really great if someone helps me out<br>> > in telling the procedures to follow.<br>> ><br>> > I have tried out Call Transfer as Transferee and it works fine.<br>
> ><br>> > Thanks in Advance...<br>> ><br>> ><br>> > Best Regards,<br>> > Irshad.<br>> > _______________________________________________<br>> > Freeswitch-dev mailing list<br>
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