-You could check the sofia debug for r15332 here: http://pastebin.freeswitch.org/11008
Phone/Devices: The caller is the DID provider's Switch. The callee (which also sends the REFER) is Asterisk 1.4.26. Testing with other devices(linksys SPA962,Grandstream GXV3000) gathers the same results.
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 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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