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[hjqlopez at hotmail.com]

Hi everybody,

I'd like some help with this situation that is 'haunting' me

My scenario is as follows:

inbound-bypass-media is set in the profile because we dont want FS handling the media.

1. A calls B
2. FS sends to B the A's SDP
3. B answers
4. FS sends to A the B's SDP
5. Media going directly between A and B
6. B REFERs the call to C (blind transfer with no reINVITE for Hold)
7. FS accepts(202) the REFER and sends the NOTIFY
7a. B and FS send the BYE
8. FS sends an INVITE to C with A's SDP
9. C answers
10. FS doesn't send a reINVITE to A to let it know about C's SDP


Is that the expected FS behavior or is this a bug?


I'd appreciate any hints. Thanks,


Humberto


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[brian at freeswitch.org]

First you forgot to mention what SVN rev you're on...

/b

On Oct 30, 2009, at 5:07 PM, Humberto Quintana wrote:

[anthony.minessale at g...]

profile options

<param name="media-option" value="resume-media-on-hold"/>
<param name="media-option" value="bypass-media-after-att-xfer"/>


On Fri, Oct 30, 2009 at 5:07 PM, Humberto Quintana <hjqlopez@hotmail.com (hjqlopez@hotmail.com)> wrote:

[hjqlopez at hotmail.com]

Thanks for you answers guys,

I test the parameters you suggested
but still no audio due to the lack of reINVITE.  By the way I'm using
1.0.4 but I also tried 1.0.5pre3.

One particular condition is that there is no on-hold before the Blind Transfer.

Regards,

Humberto

[mike at jerris.com]

Please re-try with latest svn trunk.

Mike

On Nov 2, 2009, at 9:36 AM, Humberto Quintana wrote:

[hjqlopez at hotmail.com]

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

[anthony.minessale at g...]

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 <hjqlopez@hotmail.com (hjqlopez@hotmail.com)> wrote:

[hjqlopez at hotmail.com]

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"/>
<param name="media-option" value="bypass-media-after-att-xfer"/>

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@a.b.c.d [CS_ROUTING] [RECOVERY_ON_TIMER_EXPIRE]
after sending the ACK for the reINVITE


Regards,


Humberto

[brian at freeswitch.org]

Do you have ANY nat involved?

/b

On Nov 3, 2009, at 6:05 PM, Humberto Quintana wrote:

[anthony.minessale at g...]

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 <hjqlopez@hotmail.com (hjqlopez@hotmail.com)> wrote:

[hjqlopez at hotmail.com]

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

[brian at freeswitch.org]

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:

[anthony.minessale at g...]

I did not ask you to send me a ladder diagram.
I asked you to send me a console trace from FreeSWITCH using latest trunk (1.0.4 does not help me)

1) start FreeSWITCH
2) run the cli command: console loglevel debug
3) run the cli command: sofia profile internal siptrace on
4) reproduce your issue and put the trace on freeswitch pastebin http://pastebin.freeswitch.org (login and pass are stated in the auth dialog)

Also please answer brian's question.  What phones and/or sip devices are involved in this call.



On Wed, Nov 4, 2009 at 3:39 PM, Humberto Quintana <hjqlopez@hotmail.com (hjqlopez@hotmail.com)> wrote:

[hjqlopez at hotmail.com]