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<div class="moz-cite-prefix">Yes, i checked it again.<br>
<br>
After FS gets Options it send this:<br>
<small><br>
FS=IP<br>
<br>
U SBC:5060 -> FS:5080<br>
OPTIONS <a class="moz-txt-link-freetext" href="sip:mod_sofia@FS:5080">sip:mod_sofia@FS:5080</a> SIP/2.0.<br>
Max-Forwards: 69.<br>
To: "018108754" <a class="moz-txt-link-rfc2396E" href="sip:018108754@FS"><sip:018108754@FS></a>;tag=4ejpv1UvcpQyr.<br>
From: 980659390605
<a class="moz-txt-link-rfc2396E" href="sip:38659390605@SBC"><sip:38659390605@SBC></a>;tag=3587719697-579422.<br>
Call-ID: 08d1e2d2-93ef-1231-8295-0030486435ea.<br>
CSeq: 2 OPTIONS.<br>
Allow: CANCEL, ACK, INVITE, BYE, OPTIONS, REGISTER, NOTIFY,
INFO, REFER, SUBSCRIBE, PRACK, MESSAGE, PUBLISH.<br>
Via: SIP/2.0/UDP
SBC:5060;branch=z9hG4bK87a03a84cada95d88feb886189e315c1.<br>
Contact: <a class="moz-txt-link-rfc2396E" href="sip:38659390605@SBC:5060"><sip:38659390605@SBC:5060></a>.<br>
Content-Length: 0.<br>
.<br>
<br>
<br>
U FS:5080 -> SBC:5060<br>
SIP/2.0 200 OK.<br>
Via: SIP/2.0/UDP
SBC:5060;branch=z9hG4bK87a03a84cada95d88feb886189e315c1.<br>
From: 980659390605
<a class="moz-txt-link-rfc2396E" href="sip:38659390605@SBC"><sip:38659390605@SBC></a>;tag=3587719697-579422.<br>
To: "018108754" <a class="moz-txt-link-rfc2396E" href="sip:018108754@FS"><sip:018108754@FS></a>;tag=4ejpv1UvcpQyr.<br>
Call-ID: 08d1e2d2-93ef-1231-8295-0030486435ea.<br>
CSeq: 2 OPTIONS.<br>
Contact: <a class="moz-txt-link-rfc2396E" href="sip:mod_sofia@FS:5080"><sip:mod_sofia@FS:5080></a>.<br>
User-Agent:
FreeSWITCH-mod_sofia/1.2.12+git~20130907T222757Z~b2c621bafc.<br>
Accept: application/sdp.<br>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
REGISTER, REFER, NOTIFY.<br>
Supported: timer, precondition, path, replaces.<br>
Allow-Events: talk, hold, conference, refer.<br>
Content-Type: application/sdp.<br>
Content-Length: 201.<br>
.<br>
v=0.<br>
o=FreeSWITCH 1378698444 1378698445 IN IP4 FS.<br>
s=FreeSWITCH.<br>
c=IN IP4 FS.<br>
t=0 0.<br>
m=audio 0 RTP/AVP 8 101 13.<br>
a=rtpmap:101 telephone-event/8000.<br>
a=fmtp:101 0-16.<br>
a=ptime:20.<br>
</small><br>
On other PBX which is working ok, after it recived options it send
200 ok with C= 0.0.0.0 in SDP.<br>
<br>
Can I block options?<br>
<br>
Miha<br>
<br>
<br>
<br>
Dne 9/9/2013 2:30 PM, piše Miha:<br>
</div>
<blockquote cite="mid:522DBF48.7030303@softnet.si" type="cite">
<pre wrap="">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:
</pre>
<blockquote type="cite">
<pre wrap="">hi,
here is a sip trace from FS. <a class="moz-txt-link-freetext" href="http://pastebin.freeswitch.org/21392">http://pastebin.freeswitch.org/21392</a>
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:
</pre>
<blockquote type="cite">
<pre wrap="">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 <a class="moz-txt-link-rfc2396E" href="mailto:petrpwp@gmail.com"><petrpwp@gmail.com></a> wrote:
</pre>
<blockquote type="cite">
<pre wrap="">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 ??? ????????
<a class="moz-txt-link-freetext" href="http://www.aqua-mail.com">http://www.aqua-mail.com</a>
"Miha" <a class="moz-txt-link-rfc2396E" href="mailto:miha@softnet.si"><miha@softnet.si></a> 8 ???????? 2013 ?. 21:36:15
???????:
</pre>
<blockquote type="cite">
<pre wrap="">Hi Peter,
I do not think that remote site did not send any RTP
</pre>
</blockquote>
<pre wrap="">media.
</pre>
<blockquote type="cite">
<pre wrap="">A and B sites are talking for 8 min and that FS sends
</pre>
</blockquote>
<pre wrap="">by.
</pre>
<blockquote type="cite">
<pre wrap="">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.
</pre>
</blockquote>
<pre wrap="">A
</pre>
<blockquote type="cite">
<pre wrap="">and B hear each other but connection is lost after 8
</pre>
</blockquote>
<pre wrap="">min by
</pre>
<blockquote type="cite">
<pre wrap="">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 <a class="moz-txt-link-rfc2396E" href="mailto:peter@olssononline.se"><peter@olssononline.se></a> wrote:
</pre>
<blockquote type="cite">
<pre wrap="">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
</pre>
</blockquote>
</blockquote>
<pre wrap="">media
</pre>
<blockquote type="cite">
<blockquote type="cite">
<pre wrap="">path that stops
the traffic.
/Peter
2013/9/6 Miha <a class="moz-txt-link-rfc2396E" href="mailto:miha@softnet.si"><miha@softnet.si></a>
</pre>
<blockquote type="cite">
<pre wrap="">Hi,
I need a little help. I am getting: hanging up,
</pre>
</blockquote>
</blockquote>
</blockquote>
<pre wrap="">cause:
</pre>
<blockquote type="cite">
<blockquote type="cite">
<pre wrap="">MEDIA_TIMEOUT
</pre>
<blockquote type="cite">
<pre wrap="">What could be a reason? after that FS sends bye.
tnx
miha
</pre>
</blockquote>
</blockquote>
</blockquote>
</blockquote>
<pre wrap="">_________________________________________________________________________
</pre>
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</pre>
</blockquote>
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<pre wrap="">Communication
</pre>
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<pre wrap="">Server
</pre>
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<blockquote type="cite">
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</pre>
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FreeSWITCH-powered IP PBX: The CudaTel Communication Server
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FreeSWITCH-powered IP PBX: The CudaTel Communication Server
<a class="moz-txt-link-freetext" href="http://www.cudatel.com">http://www.cudatel.com</a>
Official FreeSWITCH Sites
<a class="moz-txt-link-freetext" href="http://www.freeswitch.org">http://www.freeswitch.org</a>
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FreeSWITCH-powered IP PBX: The CudaTel Communication Server
<a class="moz-txt-link-freetext" href="http://www.cudatel.com">http://www.cudatel.com</a>
Official FreeSWITCH Sites
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