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

Anthony Minessale anthony.minessale at gmail.com
Tue Apr 27 08:47:11 PDT 2010


There is no sign of a REFER packet or anything indicating an attended
transfer in this trace.
Did you send the wrong one possibly?


On Mon, Apr 26, 2010 at 2:50 PM, Mardy Marshall <mardy at voysys.com> wrote:

> Here is the setup that was used to reproduce the consultative transfer
> problem.
>
> There are two boxes, the first is running a proxy based PBX (sipXecs) and
> the
> second is running FreeSWITCH 1.0.6.  The PBX has two Polycom phones,
> extension
> 200 and 202, registered with it.  The PBX has configured a mapping rule
> which
> will transform requests to extension 9996 to sip:9996 at 192.168.0.16:5060which
> is the address of the second box running FreeSWITCH.  FreeSWITCH has been
> configured to allow connections from the PBX box via an ACL configuration
> and
> the public dialplan includes an "echo" extension:
>
>     <extension name="echo">
>       <condition field="destination_number" expression="^9996$">
>         <action application="answer"/>
>         <action application="echo"/>
>       </condition>
>     </extension>
>
> Any of the phones registered with the PBX can dial extension 9996 and be
> connected to the FreeSWITCH echo application.  But when one phone attempts
> to transfer another phone to extension 9996 via a consultative transfer,
> FreeSWITCH does not properly complete the transfer.  You can see in the log
> at 18:55:58.4295322851 the INVITE w/Replaces is being sent to FreeSWITCH.
> FreeSWITCH accepts the INVITE but never sends a BYE to the phone which
> initiated the transfer.  Without that terminating BYE, the transfer
> controller thinks that the transfer failed.
>
> The corresponding FreeSWITCH log file -
> http://pastebin.freeswitch.org/12806
>
> If it will help, I can also forward a corresponding PCAP file.
>
> Thanks
>
> -Mardy
>
> On Apr 21, 2010, at 11:23 AM, Anthony Minessale wrote:
>
> 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
>>>>>>
>>>>>>
>>>>>> _______________________________________________
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>>>>>>
>>>>>
>>>>> _______________________________________________
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>>>>>
>>>>>
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>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> Anthony Minessale II
>>>>
>>>> FreeSWITCH http://www.freeswitch.org/
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>>>>
>>>> AIM: anthm
>>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
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>>>>
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>>>>
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>>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
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>>>
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>>>
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>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
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>>
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>> 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
>>
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>> pstn:+19193869900
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> 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
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
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>


-- 
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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