9 Dec
2005
9 Dec
'05
12:51 p.m.
Jayesh Nambiar wrote:
hi klaus,
I have the loose route block in my cfg file. I handle both the nated
clients as well as non-nated clients in the loose route section properly.
Actually the problem is I suppose with the audio codes gateway that I
use. When I tried to transfer using an x-ten softphone the call was
transferred sucessfully.
If anybody has some idea about what setting must be done, kindly share
it with me.
Yes. add an ngrep dump to your problem description.
klaus
thanks a lot for your response.
jayesh
*/Klaus Darilion <klaus.mailinglists(a)pernau.at>/* wrote:
Probably you have problems with relaying of in-dialog requests. Many
phones send a reINVITE just before the call transfer, to put the
original participant "on hold".
Do you have a
if (loose_route()) {
t_relay();
}
block in your openser.cfg?
klaus
Jayesh Nambiar wrote:
------------------------------------------------------------------------
------------------------------------------------------------------------
hi klaus,
Thanx for your response. I have hard phones connected to Audio codes
4-port gateway. The gateway is configured to send the calls to
openser
for processing. suppose there are three users
connected to it.
user1 calls user2. The call goes through. Now user 2 presses
flash and
tries to call user3, but both the phone gets
disconnected.
I did an ngrep and what i noticed was this:
When user2 tries to transfer the call to user 3, the "INVITE"
goes
from
user2 to user1 again and the call gets
disconnected due to request
timeout. I am not able to see any REFER method in the ngrep.
Any advice will be appreciated.
Thanx again.
Jayesh
*/Klaus Darilion /* wrote:
Yes, REFER method is for call transfer. Which clients do you use?
Does
the call orginate also from a SIP phone or from a
gateway? What
exactly
is the problem? Do you see the REFER request?
Including an "ngrep
port
5060" would be useful.
klaus
Jayesh Nambiar wrote:
------------------------------------------------------------------------
hi everyone,
I was just trying to transfer the call to another user. All users
are
registered at the same openser server. Now when I
receive a call,
is it
not possible to transfer the call to another IP
user.
I thought REFER mehtod does this, but i think I am wrong. Is this
possible, can somebody pls clear my understanding regarding this
concept.
thanks in advance!!
jayesh
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