Yes, i checked it again.

After FS gets Options it send this:

FS=IP

U SBC:5060 -> FS:5080
OPTIONS sip:mod_sofia@FS:5080 SIP/2.0.
Max-Forwards: 69.
To: "018108754" <sip:018108754@FS>;tag=4ejpv1UvcpQyr.
From: 980659390605 <sip:38659390605@SBC>;tag=3587719697-579422.
Call-ID: 08d1e2d2-93ef-1231-8295-0030486435ea.
CSeq: 2 OPTIONS.
Allow: CANCEL, ACK, INVITE, BYE, OPTIONS, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, MESSAGE, PUBLISH.
Via: SIP/2.0/UDP SBC:5060;branch=z9hG4bK87a03a84cada95d88feb886189e315c1.
Contact: <sip:38659390605@SBC:5060>.
Content-Length: 0.
.


U FS:5080 -> SBC:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP SBC:5060;branch=z9hG4bK87a03a84cada95d88feb886189e315c1.
From: 980659390605 <sip:38659390605@SBC>;tag=3587719697-579422.
To: "018108754" <sip:018108754@FS>;tag=4ejpv1UvcpQyr.
Call-ID: 08d1e2d2-93ef-1231-8295-0030486435ea.
CSeq: 2 OPTIONS.
Contact: <sip:mod_sofia@FS:5080>.
User-Agent: FreeSWITCH-mod_sofia/1.2.12+git~20130907T222757Z~b2c621bafc.
Accept: application/sdp.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY.
Supported: timer, precondition, path, replaces.
Allow-Events: talk, hold, conference, refer.
Content-Type: application/sdp.
Content-Length: 201.
.
v=0.
o=FreeSWITCH 1378698444 1378698445 IN IP4 FS.
s=FreeSWITCH.
c=IN IP4 FS.
t=0 0.
m=audio 0 RTP/AVP 8 101 13.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.

On other PBX which is working ok, after it recived options it send 200 ok with C= 0.0.0.0 in SDP.

Can I block options?

Miha



Dne 9/9/2013 2:30 PM, piše Miha:
Hi,

I also noticed that media stops after FS recives OPTION and send 200ok 
sdp ()

Is it possible to disable that?

Miha

Dne 9/9/2013 12:55 PM, piše Miha:
hi,

here is a sip trace from FS. http://pastebin.freeswitch.org/21392

IF i do bypasse media call is not dropped but I can not use bypass media:)

tnx for all help!

miha

Dne 9/8/2013 8:48 PM, piše Miha:
Hi Brian and Pete,

I do not think that this is FW issue. This is happening
only from calls from one provider, but I will check FW so
that I can remove this option. Could this be FW if media A
and B are hearing each other but FS send bye after 8 min?

No, phone is not place on MUTE.

I have tried from other softswtch (not FS) and I am not
experiencing this issue.



On Sun, 08 Sep 2013 22:05:37 +0400
   Pete Procenko <petrpwp@gmail.com> wrote:
Hi!
I have had similar situation, except with UDP rtp stream.
In my case the
reason was the firewall timeout, which drops udp session
after several
minutes regardless of voice activity. Maybe look at this
level?

Best regards,
     Pete


?????????? AquaMail ??? ????????
http://www.aqua-mail.com


"Miha" <miha@softnet.si> 8 ???????? 2013 ?. 21:36:15
???????:
Hi Peter,

I do not think that remote site did not send any RTP
media.
A and B sites are talking for 8 min and that FS sends
by.
Scenario is like this:

A calls B. Leg from A to FS is set on g711a but B is
sending all the time invites with sdp to FS with g711a.
A
and B hear each other but connection is lost after 8
min by
fs with bye.

When B answers to A, A site is sending options and FS
replys with 200 ok SDP () (empty sep).
What could be a reason?

tnx!
Miha

On Sat, 7 Sep 2013 12:55:35 +0200
   Peter Olsson <peter@olssononline.se> wrote:
It means that the remote endpoint didn't send any RTP
media to you for some
time. Or maybe there is a firewall issue along the
media
path that stops
the traffic.
/Peter

2013/9/6 Miha <miha@softnet.si>
Hi,

I need a little help. I am getting: hanging up,
cause:
MEDIA_TIMEOUT
What could be a reason? after that FS sends bye.

tnx
miha


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