I did not ask you to send me a ladder diagram.<br>I asked you to send me a console trace from FreeSWITCH using latest trunk (1.0.4 does not help me)<br><br>1) start FreeSWITCH<br>2) run the cli command: console loglevel debug<br>
3) run the cli command: sofia profile internal siptrace on<br>4) reproduce your issue and put the trace on freeswitch pastebin <a href="http://pastebin.freeswitch.org">http://pastebin.freeswitch.org</a> (login and pass are stated in the auth dialog)<br>
<br>Also please answer brian's question. What phones and/or sip devices are involved in this call.<br><br><br><br><div class="gmail_quote">On Wed, Nov 4, 2009 at 3:39 PM, Humberto Quintana <span dir="ltr"><<a href="mailto:hjqlopez@hotmail.com">hjqlopez@hotmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><br>
Thanks for your time,<br>
<br>
-The scenario is still the same:<br>
<br>
Always bypass media.<br>
Environment 100% NAT free :-)<br>
Call established from A to B through FS. Then... <br>
Blind transfer from B to C (Refer-to: C)<br>
RTP should go directly between A and C.<br>
<br>
<br>
-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).<br>
<br>
Please check SIP diagram here:<br>
<br>
<a href="http://provision.netcelerate.net/ngrep/blindxfer2009-11-04-v1.0.5pre3.html" target="_blank">http://provision.netcelerate.net/ngrep/blindxfer2009-11-04-v1.0.5pre3.html</a><br>
<br>
<br>
-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 ;-)<br>
<br>
<a href="http://provision.netcelerate.net/ngrep/blindxfer2009-11-04rev15332.html" target="_blank">http://provision.netcelerate.net/ngrep/blindxfer2009-11-04rev15332.html</a><br>
<br>
<br>
-You could check the sofia debug for r15332 here:<br>
<a href="http://pastebin.com/m6f2b3836" target="_blank">http://pastebin.com/m6f2b3836</a><br>
<br>
<br>
Best regards,<br>
<br>
Humberto<br>
<div class="im"><br>
><br>
> I don't know what you are talking about anymore.<br>
><br>
> The scenario I had tested is when a call is bridged in bypass_media=true<br>
> bridge<br>
> and you blind transfer that call back to the dialplan<br>
><br>
> as soon as it hits the routing state it will resume media.<br>
><br>
><br>
> it has been confirmed to not work and confirmed to have been fixed several<br>
> time and if you are still having a problem you must have something blocking<br>
> some of your packets or something .<br>
><br>
> You have to understand that sip is a protocol and your description is<br>
> completely non-standard.<br>
> Perhaps you should get a console trace and attach it to a jira. The trace<br>
> probably makes more sense to me.<br>
><br>
> sofia profile internal siptrace on<br>
> console loglevel debug<br>
><br>
> reproduce and attach the whole capture.<br>
><br>
><br>
><br>
</div><div class="im">> On Tue, Nov 3, 2009 at 6:05 PM, Humberto Quintana wrote:<br>
><br>
>><br>
>> Hi,<br>
>><br>
>> I tried r15332 and set in the sofia profile:<br>
>><br>
>> a) bypass_media_after_bridge=true only<br>
>> b) bypass_media_after_bridge=true, param name="media-option"<br>
>> value="resume-media-on-hold"/><br>
>><br>
>><br>
</div><div class="im">>> In both cases FS is hanging up the initial call (A to FS) after accepting<br>
>> the REFER to C:<br>
>><br>
>> A <- reINVITE with FS' SDP <- FS<br>
>> A -> 200 -> FS<br>
>> A <- ACK <- FS<br>
>> A <- BYE <- FS<br>
>><br>
>> The call to C is not even tried.<br>
>><br>
>> I found this line is the logs that could give some idea:<br>
>><br>
>> 2009-11-03 18:29:41.280707 [NOTICE] mod_sofia.c:733 Hangup<br>
</div>>> sofia/external/514xxxxxx at a.b.c.d [CS_ROUTING] [RECOVERY_ON_TIMER_EXPIRE]<br>
<div class="im">>> after sending the ACK for the reINVITE<br>
>><br>
>><br>
>> Regards,<br>
>><br>
>><br>
>> Humberto<br>
>><br>
>>>please try r15326<br>
>>>I think i have it working.<br>
>>><br>
>>>I recommend for optimal results you set bypass_media_after_bridge=true<br>
>>>either as a global or in your DP in place of bypass_media=true<br>
>>><br>
>>><br>
>>>On Mon, Nov 2, 2009 at 4:30 PM, Humberto Quintana<br>
</div><div class="im">>> <a href="http://hotmail.com" target="_blank">hotmail.com</a>>wrote:<br>
>>><br>
>>>> Hi Mike,<br>
>>>><br>
>>>> I re-tried with trunk rev 15319 but I got almost the same behavior:<br>
>> There<br>
>>>> is now a reINVITE (with FS' SDP) going to A when the REFER is accepted.<br>
>> But<br>
>>>> still there is no reINVITE for A (with C's SDP) after the call from FS<br>
>> to C<br>
>>>> is established.<br>
>>>><br>
>>>> Anyway, we decided for now to do a different implementation but if you<br>
>> want<br>
>>>> to explore more in this issue count me in ;-)<br>
>>>><br>
>>>><br>
>>>> Thank you very much!<br>
>>>><br>
>>>> Humberto<br>
>><br>
>><br>
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><br>
><br>
><br>
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> Anthony Minessale II<br>
><br>
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