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

[Brian West]

What phones are you using?  I do this exact same scenario without a  
problem over and over testing with anthm.  So I would like to know  
what phones you're using.

/b

On Nov 4, 2009, at 3:39 PM, Humberto Quintana wrote:

>
> 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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