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<div class="moz-cite-prefix">HI Michale,<br>
<br>
after the call is answered (second trasfer) they send first invite
withoud SDP, we send back 200 ok SDP and than they sends ACK SDP
in which they send right ip.<br>
<br>
According to RFC this is ok. Is this maybe bug on FS side?<br>
<br>
Miha<br>
<br>
Dne 6/20/2013 8:11 AM, piše Miha:<br>
</div>
<blockquote cite="mid:51C29D12.1070900@softnet.si" type="cite">
<meta content="text/html; charset=ISO-8859-1"
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<div class="moz-cite-prefix">HI Michel,<br>
<br>
so after other side puts you on hold our side gets reinvite with
ip in sdp 0.0.0.0 which is ok. After other side pickup a call
from trasfer there should be another reinvite with ip in sdp for
media if I am right? I guess this is how it shoud work.<br>
Do you maybe know if this is also defined in RFC as I unable to
find this thing, because this voip proved is saying that also
right is after call is pickuped from on hold thay can send
invite whithout SDP.<br>
<br>
tnx!<br>
<br>
miha<br>
<br>
Dne 6/19/2013 4:32 PM, piše Michel Brabants:<br>
</div>
<blockquote
cite="mid:CAHt+dNFTj_hdy2qjU4_Af4eYkHYkVfvgU0y0DkHGCwb+pVAdag@mail.gmail.com"
type="cite">This is normal behaviour for Freeswitch as it is a
b2b. A phone will also interpret the sdp as being put on-hold.
Like I said, normally, they should be sending a reinvite with a
non-0.0.0.0-sdp-ip so that the phone/freeswitch knows where to
send the rtp to.<br>
<br>
You can also use another option for hold_music in freeswitch. I
think it is hold_music=hold (I'm using this) where it will not
send music on hold, but will pass the holdsignalling to the
phone. However, this won't change your case normally, as there
should still be a reinvite with ip afterwards as far as I can
tell (didn't look at trace yet).<br>
<br>
Michel<br>
<br>
<div class="gmail_quote">On Wed, Jun 19, 2013 at 2:53 PM, Miha <span
dir="ltr"><<a moz-do-not-send="true"
href="mailto:miha@softnet.si" target="_blank">miha@softnet.si</a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0
.8ex;border-left:1px #ccc solid;padding-left:1ex">
<div text="#000000" bgcolor="#FFFFFF">
<div>HI,<br>
<br>
I can see where is a problem. After 2-trasfer is made
other side sends in sdp 0.0.0.0. which is ok, but when
call is picked up we get only invite, without SDP, so I
guess FS sends media to 0.0.0.0.<br>
<br>
Is this normal behaviour for musing on hold? this proved
is saying that thay have problems only with as.<br>
<br>
p.s.: they are using netcentrex-pbx.<br>
<br>
thanks!<br>
MIha<br>
<br>
Dne 6/19/2013 11:28 AM, piše Miha:<br>
</div>
<div>
<div class="h5">
<blockquote type="cite">
<div>Hi Michel,<br>
<br>
I was just replaying to your email as I saw you
email:) So, yes unhold-reinvite is a problem.<br>
<br>
Here is a trace of a call:<br>
<br>
<a moz-do-not-send="true"
href="http://pastebin.freeswitch.org/21075"
target="_blank">http://pastebin.freeswitch.org/21075</a><br>
<br>
Thanks for help!<br>
miha<br>
<br>
Dne 6/19/2013 10:58 AM, piše Michel Brabants:<br>
</div>
<blockquote type="cite">After rereading the issue,
it seems similar to the freeswitch-"issue", but I
would say it is a bug in the netcentrex-pbx. I
suppose netcentrex stays in the signalling path
(like a b2b) in this case as they're sending a
reinvite. <br>
Without a siptrace, I would say that they should
also sent the unhold-reinvite.<br>
<br>
Michel<br>
<br>
<div class="gmail_quote">On Wed, Jun 19, 2013 at
10:39 AM, Miha <span dir="ltr"><<a
moz-do-not-send="true"
href="mailto:miha@softnet.si"
target="_blank">miha@softnet.si</a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="margin:0
0 0 .8ex;border-left:1px #ccc
solid;padding-left:1ex">HI,<br>
<br>
I am having one issue with music on hold.<br>
<br>
A calls B (B is at different voip provider
that is using netcentrex<br>
softswitch), after B do xfer it puts A on hold
and send reinvite with<br>
0.0.0.0 in sdp, when C try to pickup this
call, just different music on<br>
hold appears but there is no media from C.<br>
<br>
how to deal with this issue?<br>
<br>
tnx!<br>
<br>
miha<br>
<br>
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