[Freeswitch-users] Having a problem with attended transfer when FS is the transfer target

Wed Apr 21 08:23:45 PDT 2010

I'm trying to understand this:

If FS is acting as a phone in your scenario why are you sending a refer to
it and not the server?
In most situations there is a b2bua server who routes the calls and takes
all the REFER.
Is this one of those PROXY only sip servers?

I think you would need to produce a full debug log of this, and if you are
using some kind of proxy based setup we would need some way to easily
reproduce it or visit your lab because we do not typically use anything of
the sort.

Execute these commands and reproduce it and capture the whole log and put it
on
http://pastebin.freeswitch.org

sofia profile internal siptrace on
console loglevel debug

On Wed, Apr 21, 2010 at 9:13 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> instead of emailing again when impatient for an answer (something we frown
> upon here in this busy list)
> produce a reproducible step by step process to duplicate your issue.  We
> are trying to help people but we don't have the time to do the leg work for
> everyone who asks a question when we get hundreds of emails a day.
>
>
>
>
> On Wed, Apr 21, 2010 at 9:01 AM, Mardy Marshall <mardy at voysys.com> wrote:
>
>> Just following up...  Does anyone have any suggestions on how to proceed
>> with this?  I've run out of ideas.
>>
>> Thanks,
>>
>> -Mardy
>>
>> On Apr 19, 2010, at 8:21 PM, Mardy Marshall wrote:
>>
>> The phones that I am using are not registered with FS.  They are
>> registered with another proxy based PBX.  I am simply using FS as B2BUA
>> which is also registered with the PBX.  And yes, I can successfully transfer
>> a call to another phone with this setup.
>>
>> To simplify things I tried the same scenario using FSComm in place of my
>> own FS application and tried to transfer a call to FSComm with the same
>> results.  And just in case there might be a problem specific to FSComm, I
>> set up a clean install of FS 1.0.6 and tried transferring a call to the FS
>> echo application with the same results.  By the way, I have no problems with
>> blind transfers, only attended transfers.
>>
>> -Mardy
>>
>> On Apr 19, 2010, at 7:53 PM, Anthony Minessale wrote:
>>
>> did you try just setting up 2 phones on plain fresh FS install, and
>> calling them normally and transferring them around?
>> That description is still pretty vague? What is an Event Socket
>> application, which has nothing to do with sip and sip transfers, that's a FS
>> protocol.
>>
>>
>> On Mon, Apr 19, 2010 at 6:33 PM, Mardy Marshall <mardy at voysys.com> wrote:
>>
>>> I have two phones (Polycom) and an event_socket application, all of which
>>> are using a SIP proxy for call routing.  The first phone calls the second
>>> phone.  The second phone then attempts to transfer the call to the
>>> FS/event_socket application by first placing the call on hold and then
>>> calling the FS application, followed by a consultative transfer.  The REFER
>>> dialog occurs between the two phones and an INVITE w/Replaces is sent to FS.
>>>  The transferred call leg appears to be answered by FS and the application
>>> receives a uuid_bridge event with the UUID of the new call leg.  The problem
>>> that I see is that the original call leg, created when the user called the
>>> FS application to announce the transfer, does not get canceled by FS and
>>> subsequently does not send the BYE back to the Polycom.  Is there something
>>> that I need to do at the event_socket application to complete the transfer?
>>>  I've tried killing the UUID associated with the first call leg as well as
>>> issuing an "answer" command to the transferred call leg UUID, but no luck.
>>>
>>> -Mardy
>>>
>>>
>>> On Apr 19, 2010, at 6:19 PM, Anthony Minessale wrote:
>>>
>>> but what is the client sending the REFER?
>>>
>>> FS gets refer+replaces all the time, if it's the one where the dest is on
>>> another box (aka the nightmare xfer that you should see references to in the
>>> debug log if so) then it will not complete until that far end call is
>>> answered.
>>>
>>> FS handles this scenerio for us hundreds of times a day using a wide
>>> range of sip devices so perhaps
>>> your UA has an interop problem.
>>>
>>>
>>> On Mon, Apr 19, 2010 at 4:47 PM, Mardy Marshall <mardy at voysys.com>wrote:
>>>
>>>>
>>>> On Apr 19, 2010, at 5:12 PM, João Mesquita wrote:
>>>>
>>>> uuid_simplify will issue the refer...
>>>>
>>>>
>>>> I looked at uuid_simplify and if I understand it correctly it is for use
>>>> when one wants to act as the transfer controller.  In my case, FS is the
>>>> transfer destination.  Another phone has already generated the refer and FS
>>>> has been sent an invite with replaces.
>>>>
>>>>
>>>> May I ask what application you are developing?
>>>>
>>>>
>>>> An ACD.
>>>>
>>>>
>>>> Regards,
>>>> João Mesquita
>>>> FSComm developer
>>>>
>>>>
>>>> On Mon, Apr 19, 2010 at 11:27 AM, Mardy Marshall <mardy at voysys.com>wrote:
>>>>
>>>>> I'm having a problem with attended transfers where the destination of
>>>>> the transfer is a FreeSWITCH based application such as FSComm.  (It should
>>>>> be noted that in my setup the phone performing the transfer and the caller
>>>>> which is being transferred are parties of another SIP server.)  What I see,
>>>>> from a SIP signaling standpoint, is that after FreeSWITCH receives and
>>>>> acknowledges the INVITE w/Replaces it does not terminate the initial call
>>>>> leg by sending a BYE to the transfer controller.  From the FreeSWITCH
>>>>> application side, FS still thinks that both the initial call leg and
>>>>> transferred call leg are active.  I experimented with trying to explicitly
>>>>> terminate the initial call leg by using uuid_kill, but this caused FS to
>>>>> kill all legs of the call.  Is there a specific action that the application
>>>>> must take in order for the transfer to complete?
>>>>>
>>>>> -Mardy
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
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>>>>
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>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
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>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
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>>
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>> IRC: irc.freenode.net #freeswitch
>>
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>>
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>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
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>
> AIM: anthm
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>
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> pstn:+19193869900
>

-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
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Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
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