[Freeswitch-users] no REINVITE on Blind Transfer with bypass_media

[Humberto Quintana]

Thanks for your time,

-The scenario is still the same: 

Always bypass media. 
Environment 100% NAT free :-)
Call established from A to B through FS. Then...  
Blind transfer from B to C (Refer-to: C)
RTP should go directly between A and C.


-With 1.0.4 and 1.0.5pre3, FS actually INVITEs C after receiving the REFER-to:C, BUT there is no 2-way audio.  Only RTP from C to A (due to the lack of reINVITE to A, after C answers). 

Please check SIP diagram here:

http://provision.netcelerate.net/ngrep/blindxfer2009-11-04-v1.0.5pre3.html


-What it's wrong with r15332 is there is not such call to C. For sure I know SIP is a protocol, may be my description was not clear but this SIP diagram speaks by itself ;-)

http://provision.netcelerate.net/ngrep/blindxfer2009-11-04rev15332.html


-You could check the sofia debug for r15332 here:
http://pastebin.com/m6f2b3836


Best regards,

Humberto

>
> I don't know what you are talking about anymore.
>
> The scenario I had tested is when a call is bridged in bypass_media=true
> bridge
> and you blind transfer that call back to the dialplan
>
> as soon as it hits the routing state it will resume media.
>
>
> it has been confirmed to not work and confirmed to have been fixed several
> time and if you are still having a problem you must have something blocking
> some of your packets or something .
>
> You have to understand that sip is a protocol and your description is
> completely non-standard.
> Perhaps you should get a console trace and attach it to a jira. The trace
> probably makes more sense to me.
>
> sofia profile internal siptrace on
> console loglevel debug
>
> reproduce and attach the whole capture.
>
>
>
> On Tue, Nov 3, 2009 at 6:05 PM, Humberto Quintana wrote:
>
>>
>> Hi,
>>
>> I tried r15332 and set in the sofia profile:
>>
>> a) bypass_media_after_bridge=true only
>> b) bypass_media_after_bridge=true, param name="media-option"
>> value="resume-media-on-hold"/>
>> 
>>
>> In both cases FS is hanging up the initial call (A to FS) after accepting
>> the REFER to C:
>>
>> A <- reINVITE with FS' SDP <- FS
>> A -> 200 -> FS
>> A <- ACK <- FS
>> A <- BYE <- FS
>>
>> The call to C is not even tried.
>>
>> I found this line is the logs that could give some idea:
>>
>> 2009-11-03 18:29:41.280707 [NOTICE] mod_sofia.c:733 Hangup
>> sofia/external/514xxxxxx at a.b.c.d [CS_ROUTING] [RECOVERY_ON_TIMER_EXPIRE]
>> after sending the ACK for the reINVITE
>>
>>
>> Regards,
>>
>>
>> Humberto
>>
>>>please try r15326
>>>I think i have it working.
>>>
>>>I recommend for optimal results you set bypass_media_after_bridge=true
>>>either as a global or in your DP in place of bypass_media=true
>>>
>>>
>>>On Mon, Nov 2, 2009 at 4:30 PM, Humberto Quintana 
>> hotmail.com>wrote:
>>>
>>>> Hi Mike,
>>>>
>>>> I re-tried with trunk rev 15319 but I got almost the same behavior:
>> There
>>>> is now a reINVITE (with FS' SDP) going to A when the REFER is accepted.
>> But
>>>> still there is no reINVITE for A (with C's SDP) after the call from FS
>> to C
>>>> is established.
>>>>
>>>> Anyway, we decided for now to do a different implementation but if you
>> want
>>>> to explore more in this issue count me in ;-)
>>>>
>>>>
>>>> Thank you very much!
>>>>
>>>> Humberto
>>
>>
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>
>
>
> --
> Anthony Minessale II
>
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