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[durk.debeer at isp.sol...]

Hello all,
I have observed an issue on using Freeswitch and some SIP-phones. Ok the problem is this. Some phones, when attempting an attendant transfer, put the recipient of the transfer on hold. This results in Freeswitch starting to stream MOH music to the phone put on hold, if implemented. When now the original phone is pasing the transfer, Freeswitch is not going to process this transfer because the recipient end of it is on hold. It is however terminating the connections it has with the phone initiating the transfer. This means that the recipient of the transfer is never coming of hold again until it terminates the call.
Is there a way to detect this behaviour, so I can get the recipient of hold before Freeswitch is processing the transfer?.
Kind regards
Durk

[brian at freeswitch.org]

This makes no sense.... Can you try to explain it more? I do attended transfers with sip phones every day without a single problem. Maybe i'm missing what you're talking about.

/b

On Jun 8, 2009, at 2:37 AM, Durk de Beer wrote:

[intralanman at freeswi...]

Durk de Beer wrote:

[durk.debeer at isp.sol...]

Ok I've recieved an error message so if this message is being send a second
time my deepest apologies for it.

Hello Brian,

I observed the problem by a Siemens Gigaset DE380 IPR and A Cisco 7960 and
7965. What happens is this, a call is coming in on Freeswitch and is being
bridged to lets say the Siemens on extension 100. Now the person accepting
the call on extension 100 wants to transfer the call to an other extension
lets say 200. The normal scenario would be extension 100 puts the original
call on hold, Freeswitch streams moh to the original call, extension 100
dials 200, the having a conversation, 100 transfers the original call too
200. Now here the problem begins. Normally extension 100 would send a refer
sip message to freeswitch, who would then connect the original call to
extension 200. The Siemens and Cisco phones do not send a refer sip message
first. What the do is putting extension 200 on hold by means of an send only
sip message. When Freeswitch is receiving this it streams moh to extension
200. After this the phones are sending the transfer by means of a sip refer
message. When Freeswitch is receiving this it can't perform this transfer,
that of original call to extension 200, because extension 100 has put
extension 200 on receive only and extension 200 is receiving the moh.
Resulting in an original call receiving moh and an extension 200 receiving
moh. When this situation arises there's no way in connecting these to
together. So what I need is a way to detect that there is an transfer by
means of an sip refer message to a extension that has being put on hold. If
so I need to get freeswitch to break this hold and transfer the original
call to this extension. I hope that this will make the problem a little bit
clearer.

Kind regards
Durk

[durk.debeer at isp.sol...]

Hallo Ray

Siemens type Gigaset DE380 IPR, Cisco type 7965 and 7960.
Also tested Grandstream type 2010 and Linksys type 921 no problem with these
phones.
I've downloaded a new firmware for the Siemens but wasn't able to test it
jet.

Durk