I'll have to defer to those more experienced than I in such matters. However, I can offer two tips:

#1 - turn off the crazy sofia debugging - it's just noise. All you need to do to enable SIP trace is "sofia global siptrace on"
#2 - when you pastebin the console output use the FreeSWITCH log syntax highlighting - it makes it *much* easier to see what's going on.

-MC

On Mon, Apr 4, 2011 at 10:51 PM, Joegen E. Baclor <joegen@opensipstack.org> wrote:
Hi Michael,

I have pasted both working and none working logs on pastebin.

FreeSWITCH Version 1.0.7 (hacked-20110326T123355Z)
working:  http://pastebin.freeswitch.org/16008
not working:  http://pastebin.freeswitch.org/16009

The call flow for the working call is
UA1 ->  (FSBridgeDialPlan) -> (SIP-Loopback) -> (FSIVRApp)
FSIVRApp knows the uuid of the bridge call.  Pressing # on the IVR results to a uuid_deflect on the bridged channel.  This works and call successfully transfers to the new destination.

The call flow for the none working call is

1.  UA1 -> UA2  is in conversation
2.  UA1 puts UA2 on hold

-- start of FS interaction here --

3.  UA1 ->  (FSBridgeDialPlan) -> (SIP-Loopback) -> (FSIVRApp)  (on line 2)
4.  UA1 sends REFER (replacing its call with UA2) to FSBridgeDialPlan.
5.  Flow is now UA2 ->  ([REPLACED]FSBridgeDialPlan) -> (SIP-Loopback) -> (FSIVRApp)
6.  UA2 presses #.
7.  IVRApp performs uuid_deflect on FSBridgeDialPlan.
8. FSBridgeDialPlan drops call (no REFER is done)

Thanks for your help.

Joegen


On 04/05/2011 12:35 PM, Michael Collins wrote:
What do you see on the console when you try this? A console debug log with siptrace would go a long way toward figuring out what is happening.

-MC

On Mon, Apr 4, 2011 at 9:27 PM, Joegen E. Baclor <joegen@opensipstack.org> wrote:
Hi List,

I have a scenario where a bridged call has been replaced due to a
consultative transfer.  This works pretty well and audio is
bidirectional.  I have the original uuid of the call in a var
somewhere.  The trouble begins when I uuid_deflect the bridged call once
again to attempt another transfer.  Sofia disconnects the channel.  I am
using the original uuid of the call (uuid prior to replaces).  Is this
the right way of doing it?

Joegen

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