-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.
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
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
Do you have ANY nat involved?
/b
On Nov 3, 2009, at 6:05 PM, Humberto Quintana wrote:
2009-11-03 18:29:41.280707 [NOTICE] mod_sofia.c:733 Hangup sofia/
external/514xxx...@a.b.c.d [CS_ROUTING] [RECOVERY_ON_TIMER_EXPIRE]
after sending the ACK for the reINVITE
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
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
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
param name=media-option
Please re-try with latest svn trunk.
Mike
On Nov 2, 2009, at 9:36 AM, Humberto Quintana wrote:
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
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
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 hjqlo...@hotmail.comwrote:
Hi Mike,
I re-tried with trunk
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
First you forgot to mention what SVN rev you're on...
/b
On Oct 30, 2009, at 5:07 PM, Humberto Quintana wrote:
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
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 hjqlo...@hotmail.comwrote:
Hi everybody,
I'd like some help with this situation that is 'haunting' me :-)
My
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