[Freeswitch-users] Transfer attempt for a previously a replaced call fails

[Anthony Minessale]

The case where you are saying it doesnt work, the refer is to a call
on another box so its doing what we call a nightmare transfer where FS
INVITES the remote leg then cross connects them.  According to the log
as soon as they are bridge one leg of the call seems to be hungup or
disconnected. Is this when you press #?


On Thu, Apr 7, 2011 at 11:05 AM, Joegen E. Baclor
<joegen at opensipstack.org> wrote:
> Just bumping this thread.   If I need to provide more info, just let me
> know.  Or if this is a known bug and a fix is due for a future version that
> is also acceptable.
>
> On 04/06/2011 08:02 AM, Joegen E. Baclor wrote:
>
> I'll keep that in mind.  If more information is needed to get into the
> bottom of this, I will happily oblige.  Thanks for helping.
>
> On 04/06/2011 03:09 AM, Michael Collins wrote:
>
> 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 at 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 at 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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>>
>>
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>
>
>
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-- 
Anthony Minessale II

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