<div dir="ltr"><br><div>Thanks - Ive tried to play with these functions, but not much luck. However, thanks to the linphone mailing list, I realised that its configuration was wrong (for some stupid reason, the network settings had a null public address defined), hence the issue !</div><div><br></div><div>Moving on to the dispatcher.... some fun ahead I guess !</div><div><br></div><div>Thanks again for the help</div><div><br></div><div>J.</div></div><div class="gmail_extra"><br><div class="gmail_quote">On Thu, May 18, 2017 at 4:12 AM, Daniel-Constantin Mierla <span dir="ltr"><<a href="mailto:miconda@gmail.com" target="_blank">miconda@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div bgcolor="#FFFFFF" text="#000000">
<p>Using set_contact_alias()/handle_<wbr>ruri_alias() should get this
working in terms of routing through kamailio.</p>
<p>But probably is more sane to fix it in client side, other UAs may
complain on such domain part.</p>
<p>Cheers,<br>
Daniel<br>
</p><div><div class="h5">
<br>
<div class="m_-101634289409263049moz-cite-prefix">On 17.05.17 21:23, David Villasmil
wrote:<br>
</div>
<blockquote type="cite">You can set the contact manually, look up
set_contact_alias() ... I'm not sure whether this would be
advisable though... you need someone more
kamailio-knowledgeable...<br>
<br>
I think the problem is the linphone config, i sometimes use it and
have never seen that... <br>
<div class="gmail_quote">
<div dir="ltr">On Wed, May 17, 2017 at 8:45 PM Jean Cérien <<a href="mailto:cerien.jean@gmail.com" target="_blank">cerien.jean@gmail.com</a>>
wrote:<br>
</div>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div dir="ltr"><br>
<div>Thanks for the help. </div>
<div>I have reverted to the default config file (<a href="https://github.com/sipwise/kamailio/blob/master/etc/kamailio.cfg" target="_blank">https://github.com/sipwise/<wbr>kamailio/blob/master/etc/<wbr>kamailio.cfg</a>),
and trying to place a call between two ua (linphone &
zoiper). I am testing totally on a LAN, clients &
kamailo on the same subnet, no nat.</div>
<div><br>
</div>
<div>Register is fine, Invite is fine, I receive the 200OK
from called, then I get the ACK from the calling, and
while processing it, I get the following errors in the
log: </div>
<div>
<div><font face="monospace, monospace"><br>
</font></div>
<div><font face="monospace, monospace">May 17 14:12:32
kamailio : ERROR: <core> [resolve.c:1694]:
sip_hostport2su(): could not resolve hostname:
"(null)"</font></div>
<div><font face="monospace, monospace">May 17 14:12:32
kamailio : ERROR: <core> [forward.c:495]:
forward_request(): bad host name (null), dropping
packet</font></div>
<div><font face="monospace, monospace">May 17 14:12:32
kamailio : ERROR: sl [sl_funcs.c:363]:
sl_reply_error(): ERROR: sl_reply_error used:
Unresolvable destination (478/SL)</font></div>
</div>
<div><br>
</div>
<div>Digging a bit more, I've noticed that the calling
party, using Linphone, has the Contact field a bit weird: </div>
<div>Contact: <a class="m_-101634289409263049moz-txt-link-rfc2396E"><sip:user@(null)></a></div>
<div><br>
</div>
<div>Changing to another softphone that populates correctly
this field works ok. Is there a way to mitigate this
external issue ? The softphone works ok directly connected
to asterisk for instance</div>
<div><br>
</div>
<div>Rgds</div>
</div>
<div dir="ltr">
<div>J.</div>
<div><br>
</div>
</div>
<div class="gmail_extra"><br>
<div class="gmail_quote">On Tue, May 16, 2017 at 6:15 PM,
David Villasmil <span dir="ltr"><<a href="mailto:david.villasmil.work@gmail.com" target="_blank">david.villasmil.work@gmail.<wbr>com</a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">There
isn't an ACK received, check in kamailio side to make
sure it is received. This is most probably a nat issue.<br>
<div class="gmail_quote">
<div>
<div class="m_-101634289409263049m_-7672058114087510294h5">
<div dir="ltr">On Tue, May 16, 2017 at 11:20 PM
Jean Cérien <<a href="mailto:cerien.jean@gmail.com" target="_blank">cerien.jean@gmail.com</a>>
wrote:<br>
</div>
</div>
</div>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div>
<div class="m_-101634289409263049m_-7672058114087510294h5">
<div dir="ltr"><br>
<div>Hello</div>
<div><br>
</div>
<div>I am getting started with Kamailio (or
restarted, used it briefly years ago), with
the final objective to do load balancing.</div>
<div><br>
</div>
<div>For the time being, I am just trying to
have one asterisk and one kamailio, on the
same box. I have setup a box with an
asterisk 11.3, and kamailio 4.4. I've taken
the config file from <a href="http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb" target="_blank">http://kb.asipto.com/<wbr>asterisk:realtime:kamailio-4.<wbr>0.x-asterisk-11.3.0-astdb</a>. </div>
<div><br>
</div>
<div>My idea is that asterisk runs on port
5080, while kamailio on port 5060. Client
interacts with Kamailio on port 5060. </div>
<div><br>
</div>
<div>It almost works... Registration is fine,
but when I send an invite, it is properly
acknowledged (by asterisk 100 trying then
200 OK) - but the OK message gets repeated
multiple times and asterisk issues its
infamous 'Retransmission timeout reached
...' - as if Kamailio wasnt processing it.
See below ngrep traces between asterisk and
kamailio</div>
<div><br>
</div>
<div>Any ideas where to look ?</div>
<div><br>
</div>
<div>Thanks</div>
<div>J.</div>
<div><br>
</div>
<div>
<div><font face="monospace, monospace">#</font></div>
<div><font face="monospace, monospace">U
+18.289105 <a href="http://192.168.2.228:5060" target="_blank">192.168.2.228:5060</a>
-> <a href="http://192.168.2.228:5080" target="_blank">192.168.2.228:5080</a></font></div>
<div><font face="monospace, monospace">
INVITE <a href="mailto:sip%3A102@192.168.2.228" target="_blank">sip:102@192.168.2.228</a>
SIP/2.0..Record-Route:
<a class="m_-101634289409263049moz-txt-link-rfc2396E"><sip:192.168.2.228;lr=on;ftag=<wbr>1034946464></a>..Via:
SIP/2.0/UDP
192.168.2.228;branch=<wbr>z9hG4bK9e22.<wbr>bff83e1026a667df63b9922</font></div>
<div><font face="monospace, monospace">
4198ef593.0..Via: SIP/2.0/UDP
192.168.2.200:5085;received=<wbr>192.168.2.200;rport=5085;<wbr>branch=z9hG4bK1247964647..<wbr>From:
<<a href="mailto:sip%3A199@192.168.2.228" target="_blank">sip:199@192.168.2.228</a>>;tag=<wbr>1034946464..To:
<<a class="m_-101634289409263049moz-txt-link-freetext">sip:102@</a></font></div>
<div><font face="monospace, monospace">
192.168.2.228>..Call-ID:
1571382735..CSeq: 21 INVITE..Contact:
<a class="m_-101634289409263049moz-txt-link-rfc2396E"><sip:iper@(null)></a>..Content-<wbr>Type:
application/sdp..Allow: INVITE, ACK,
CANCEL, OPTIONS, BYE, REFER, NOTIFY</font></div>
<div><font face="monospace, monospace"> ,
MESSAGE, SUBSCRIBE, INFO..Max-Forwards:
69..User-Agent: Linphone/3.6.1
(eXosip2/4.1.0)..Subject: Phone
call..Content-Length:
437....v=0..o=199 2799 2990 IN IP4 192.</font></div>
<div><font face="monospace, monospace">
168.2.200..s=Talk..c=IN IP4
192.168.2.200..t=0 0..m=audio 7078
RTP/AVP 124 111 110 0 8
101..a=rtpmap:124 opus/48000..a=fmtp:124
useinbandfec=1; usedtx=1..a=rtpmap:111 s</font></div>
<div><font face="monospace, monospace">
peex/16000..a=fmtp:111
vbr=on..a=rtpmap:110
speex/8000..a=fmtp:110
vbr=on..a=rtpmap:101
telephone-event/8000..a=fmtp:<wbr>101
0-11..m=video 9078 RTP/AVP 103
99..a=rtpmap:103</font></div>
<div><font face="monospace, monospace">
VP8/90000..a=rtpmap:99
MP4V-ES/90000..a=fmtp:99
profile-level-id=3..
</font></div>
<div><font face="monospace, monospace">#</font></div>
<div><font face="monospace, monospace">U
+0.000683 <a href="http://192.168.2.228:5080" target="_blank">192.168.2.228:5080</a>
-> <a href="http://192.168.2.228:5060" target="_blank">192.168.2.228:5060</a></font></div>
<div><font face="monospace, monospace">
OPTIONS <a href="http://sip:199@192.168.2.228:5060" target="_blank">sip:199@192.168.2.228:5060</a>
SIP/2.0..Via: SIP/2.0/UDP
192.168.2.228:5080;branch=<wbr>z9hG4bK7c2d71fb..Max-Forwards:
70..From: "asterisk" <<a href="http://sip:199@192.168.2.228:5080" target="_blank">sip:199@192.168.2.228:5080</a>>;</font></div>
<div><font face="monospace, monospace">
tag=as57c98c4b..To: <<a href="http://sip:199@192.168.2.228:5060" target="_blank">sip:199@192.168.2.228:5060</a>>..<wbr>Contact:
<<a href="http://sip:199@192.168.2.228:5080" target="_blank">sip:199@192.168.2.228:5080</a>>..<wbr>Call-ID:
<a class="m_-101634289409263049moz-txt-link-abbreviated" href="mailto:52fa034372ce18ca2b93fc1817ad38a5@192.168.2.228:5080..CSeq" target="_blank">52fa034372ce18ca2b93fc1817ad38<wbr>a5@192.168.2.228:5080..CSeq</a>: 102 OPTIONS</font></div>
<div><font face="monospace, monospace">
..User-Agent: Asterisk PBX 11.3.0..Date:
Tue, 16 May 2017 21:08:29 GMT..Allow:
INVITE, ACK, CANCEL, OPTIONS, BYE,
REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH..Supported: re</font></div>
<div><font face="monospace, monospace">
places, timer..Content-Length: 0....
</font></div>
<div><font face="monospace, monospace">#</font></div>
<div><font face="monospace, monospace">U
+0.001035 <a href="http://192.168.2.228:5080" target="_blank">192.168.2.228:5080</a>
-> <a href="http://192.168.2.228:5060" target="_blank">192.168.2.228:5060</a></font></div>
<div><font face="monospace, monospace">
OPTIONS <a href="http://sip:199@192.168.2.228:5060" target="_blank">sip:199@192.168.2.228:5060</a>
SIP/2.0..Via: SIP/2.0/UDP
192.168.2.228:5080;branch=<wbr>z9hG4bK4248158e..Max-Forwards:
70..From: "asterisk" <<a href="http://sip:199@192.168.2.228:5080" target="_blank">sip:199@192.168.2.228:5080</a>>;</font></div>
<div><font face="monospace, monospace">
tag=as30d9cfb4..To: <<a href="http://sip:199@192.168.2.228:5060" target="_blank">sip:199@192.168.2.228:5060</a>>..<wbr>Contact:
<<a href="http://sip:199@192.168.2.228:5080" target="_blank">sip:199@192.168.2.228:5080</a>>..<wbr>Call-ID:
<a class="m_-101634289409263049moz-txt-link-abbreviated" href="mailto:4331da391bca02965b2af65254717a18@192.168.2.228:5080..CSeq" target="_blank">4331da391bca02965b2af65254717a<wbr>18@192.168.2.228:5080..CSeq</a>: 102 OPTIONS</font></div>
<div><font face="monospace, monospace">
..User-Agent: Asterisk PBX 11.3.0..Date:
Tue, 16 May 2017 21:08:29 GMT..Allow:
INVITE, ACK, CANCEL, OPTIONS, BYE,
REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH..Supported: re</font></div>
<div><font face="monospace, monospace">
places, timer..Content-Length: 0....
</font></div>
<div><font face="monospace, monospace">#</font></div>
<div><font face="monospace, monospace">U
+0.000407 <a href="http://192.168.2.228:5080" target="_blank">192.168.2.228:5080</a>
-> <a href="http://192.168.2.228:5060" target="_blank">192.168.2.228:5060</a></font></div>
<div><font face="monospace, monospace">
SIP/2.0 100 Trying..Via: SIP/2.0/UDP
192.168.2.228;branch=<wbr>z9hG4bK9e22.<wbr>bff83e1026a667df63b99224198ef5<wbr>93.0;received=192.168.2.228..<wbr>Via:
SIP/2.0/UDP 192.168.2.200:5085;rec</font></div>
<div><font face="monospace, monospace">
eived=192.168.2.200;rport=<wbr>5085;branch=z9hG4bK1247964647.<wbr>.Record-Route:
<a class="m_-101634289409263049moz-txt-link-rfc2396E"><sip:192.168.2.228;lr=on;ftag=<wbr>1034946464></a>..From: <<a href="mailto:sip%3A199@192.168.2.228" target="_blank">sip:199@192.168.2.228</a>>;tag=<wbr>1034946464..To:
<sip</font></div>
<div><font face="monospace, monospace"> :<a href="mailto:102@192.168.2.228" target="_blank">102@192.168.2.228</a>>..Call-ID:
1571382735..CSeq: 21 INVITE..Server:
Asterisk PBX 11.3.0..Allow: INVITE, ACK,
CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
NOTIFY, INFO, PUBLIS</font></div>
<div><font face="monospace, monospace">
H..Supported: replaces, timer..Contact:
<<a href="http://sip:102@192.168.2.228:5080" target="_blank">sip:102@192.168.2.228:5080</a>>..<wbr>Content-Length:
0....
</font></div>
<div><font face="monospace, monospace">#</font></div>
<div><font face="monospace, monospace">U
+0.003961 <a href="http://192.168.2.228:5080" target="_blank">192.168.2.228:5080</a>
-> <a href="http://192.168.2.228:5060" target="_blank">192.168.2.228:5060</a></font></div>
<div><font face="monospace, monospace">
SIP/2.0 200 OK..Via: SIP/2.0/UDP
192.168.2.228;branch=<wbr>z9hG4bK9e22.<wbr>bff83e1026a667df63b99224198ef5<wbr>93.0;received=192.168.2.228..<wbr>Via:
SIP/2.0/UDP 192.168.2.200:5085;receive</font></div>
<div><font face="monospace, monospace">
d=192.168.2.200;rport=5085;<wbr>branch=z9hG4bK1247964647..<wbr>Record-Route:
<a class="m_-101634289409263049moz-txt-link-rfc2396E"><sip:192.168.2.228;lr=on;ftag=<wbr>1034946464></a>..From: <<a href="mailto:sip%3A199@192.168.2.228" target="_blank">sip:199@192.168.2.228</a>>;tag=<wbr>1034946464..To:
<<a class="m_-101634289409263049moz-txt-link-freetext">sip:102</a></font></div>
<div><font face="monospace, monospace"> @<a href="http://192.168.2.228" target="_blank">192.168.2.228</a>>;tag=<wbr>as497f35c0..Call-ID:
1571382735..CSeq: 21 INVITE..Server:
Asterisk PBX 11.3.0..Allow: INVITE, ACK,
CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
NOTIFY, I</font></div>
<div><font face="monospace, monospace">
NFO, PUBLISH..Supported: replaces,
timer..Contact: <<a href="http://sip:102@192.168.2.228:5080" target="_blank">sip:102@192.168.2.228:5080</a>>..<wbr>Content-Type:
application/sdp..Content-<wbr>Length:
312....v=0..o=root 350189084 350189084 I</font></div>
<div><font face="monospace, monospace"> N
IP4 192.168.2.228..s=Asterisk PBX
11.3.0..c=IN IP4 192.168.2.228..t=0
0..m=audio 17148 RTP/AVP 0 8
101..a=rtpmap:0 PCMU/8000..a=rtpmap:8
PCMA/8000..a=rtpmap:101 telep</font></div>
<div><font face="monospace, monospace">
hone-event/8000..a=fmtp:101
0-16..a=silenceSupp:off - - -
-..a=ptime:20..a=sendrecv..m=<wbr>video 0
RTP/AVP 103 99..
</font></div>
<div><font face="monospace, monospace">#</font></div>
<div><font face="monospace, monospace">U
+0.087859 <a href="http://192.168.2.228:5060" target="_blank">192.168.2.228:5060</a>
-> <a href="http://192.168.2.228:5080" target="_blank">192.168.2.228:5080</a></font></div>
<div><font face="monospace, monospace">
SIP/2.0 200 OK..Via: SIP/2.0/UDP
192.168.2.228:5080;branch=<wbr>z9hG4bK7c2d71fb..From:
"asterisk" <<a href="http://sip:199@192.168.2.228:5080" target="_blank">sip:199@192.168.2.228:5080</a>>;<wbr>tag=as57c98c4b..To:
<<a href="http://sip:199@192.168.2.228:506" target="_blank">sip:199@192.168.2.228:506</a></font></div>
<div><font face="monospace, monospace">
0>;tag=524348182..Call-ID:
<a class="m_-101634289409263049moz-txt-link-abbreviated" href="mailto:52fa034372ce18ca2b93fc1817ad38a5@192.168.2.228:5080..CSeq" target="_blank">52fa034372ce18ca2b93fc1817ad38<wbr>a5@192.168.2.228:5080..CSeq</a>:
102 OPTIONS..Allow: INVITE, ACK, BYE,
CANCEL, OPTIONS, MESSAGE, SUBSCRIBE,
NOTIFY,</font></div>
<div><font face="monospace, monospace">
INFO..Accept:
application/sdp..User-Agent:
Linphone/3.6.1
(eXosip2/4.1.0)..Content-<wbr>Length: 0....
</font></div>
<div><font face="monospace, monospace">#</font></div>
<div><font face="monospace, monospace">U
+0.000213 <a href="http://192.168.2.228:5060" target="_blank">192.168.2.228:5060</a>
-> <a href="http://192.168.2.228:5080" target="_blank">192.168.2.228:5080</a></font></div>
<div><font face="monospace, monospace">
SIP/2.0 200 OK..Via: SIP/2.0/UDP
192.168.2.228:5080;branch=<wbr>z9hG4bK4248158e..From:
"asterisk" <<a href="http://sip:199@192.168.2.228:5080" target="_blank">sip:199@192.168.2.228:5080</a>>;<wbr>tag=as30d9cfb4..To:
<<a href="http://sip:199@192.168.2.228:506" target="_blank">sip:199@192.168.2.228:506</a></font></div>
<div><font face="monospace, monospace">
0>;tag=939659485..Call-ID:
<a class="m_-101634289409263049moz-txt-link-abbreviated" href="mailto:4331da391bca02965b2af65254717a18@192.168.2.228:5080..CSeq" target="_blank">4331da391bca02965b2af65254717a<wbr>18@192.168.2.228:5080..CSeq</a>:
102 OPTIONS..Allow: INVITE, ACK, BYE,
CANCEL, OPTIONS, MESSAGE, SUBSCRIBE,
NOTIFY,</font></div>
<div><font face="monospace, monospace">
INFO..Accept:
application/sdp..User-Agent:
Linphone/3.6.1
(eXosip2/4.1.0)..Content-<wbr>Length: 0....
</font></div>
<div><font face="monospace, monospace">#</font></div>
<div><font face="monospace, monospace">U
+0.011138 <a href="http://192.168.2.228:5080" target="_blank">192.168.2.228:5080</a>
-> <a href="http://192.168.2.228:5060" target="_blank">192.168.2.228:5060</a></font></div>
<div><font face="monospace, monospace">
SIP/2.0 200 OK..Via: SIP/2.0/UDP
192.168.2.228;branch=<wbr>z9hG4bK9e22.<wbr>bff83e1026a667df63b99224198ef5<wbr>93.0;received=192.168.2.228..<wbr>Via:
SIP/2.0/UDP 192.168.2.200:5085;receive</font></div>
<div><font face="monospace, monospace">
d=192.168.2.200;rport=5085;<wbr>branch=z9hG4bK1247964647..<wbr>Record-Route:
<a class="m_-101634289409263049moz-txt-link-rfc2396E"><sip:192.168.2.228;lr=on;ftag=<wbr>1034946464></a>..From: <<a href="mailto:sip%3A199@192.168.2.228" target="_blank">sip:199@192.168.2.228</a>>;tag=<wbr>1034946464..To:
<<a class="m_-101634289409263049moz-txt-link-freetext">sip:102</a></font></div>
<div><font face="monospace, monospace"> @<a href="http://192.168.2.228" target="_blank">192.168.2.228</a>>;tag=<wbr>as497f35c0..Call-ID:
1571382735..CSeq: 21 INVITE..Server:
Asterisk PBX 11.3.0..Allow: INVITE, ACK,
CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
NOTIFY, I</font></div>
<div><font face="monospace, monospace">
NFO, PUBLISH..Supported: replaces,
timer..Contact: <<a href="http://sip:102@192.168.2.228:5080" target="_blank">sip:102@192.168.2.228:5080</a>>..<wbr>Content-Type:
application/sdp..Content-<wbr>Length:
312....v=0..o=root 350189084 350189084 I</font></div>
<div><font face="monospace, monospace"> N
IP4 192.168.2.228..s=Asterisk PBX
11.3.0..c=IN IP4 192.168.2.228..t=0
0..m=audio 17148 RTP/AVP 0 8
101..a=rtpmap:0 PCMU/8000..a=rtpmap:8
PCMA/8000..a=rtpmap:101 telep</font></div>
<div><font face="monospace, monospace">
hone-event/8000..a=fmtp:101
0-16..a=silenceSupp:off - - -
-..a=ptime:20..a=sendrecv..m=<wbr>video 0
RTP/AVP 103 99..
</font></div>
<div><font face="monospace, monospace">#</font></div>
<div><font face="monospace, monospace">U
+0.200291 <a href="http://192.168.2.228:5080" target="_blank">192.168.2.228:5080</a>
-> <a href="http://192.168.2.228:5060" target="_blank">192.168.2.228:5060</a></font></div>
<div><font face="monospace, monospace">
SIP/2.0 200 OK..Via: SIP/2.0/UDP
192.168.2.228;branch=<wbr>z9hG4bK9e22.<wbr>bff83e1026a667df63b99224198ef5<wbr>93.0;received=192.168.2.228..<wbr>Via:
SIP/2.0/UDP 192.168.2.200:5085;receive</font></div>
<div><font face="monospace, monospace">
d=192.168.2.200;rport=5085;<wbr>branch=z9hG4bK1247964647..<wbr>Record-Route:
<a class="m_-101634289409263049moz-txt-link-rfc2396E"><sip:192.168.2.228;lr=on;ftag=<wbr>1034946464></a>..From: <<a href="mailto:sip%3A199@192.168.2.228" target="_blank">sip:199@192.168.2.228</a>>;tag=<wbr>1034946464..To:
<<a class="m_-101634289409263049moz-txt-link-freetext">sip:102</a></font></div>
<div><font face="monospace, monospace"> @<a href="http://192.168.2.228" target="_blank">192.168.2.228</a>>;tag=<wbr>as497f35c0..Call-ID:
1571382735..CSeq: 21 INVITE..Server:
Asterisk PBX 11.3.0..Allow: INVITE, ACK,
CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
NOTIFY, I</font></div>
<div><font face="monospace, monospace">
NFO, PUBLISH..Supported: replaces,
timer..Contact: <<a href="http://sip:102@192.168.2.228:5080" target="_blank">sip:102@192.168.2.228:5080</a>>..<wbr>Content-Type:
application/sdp..Content-<wbr>Length:
312....v=0..o=root 350189084 350189084 I</font></div>
<div><font face="monospace, monospace"> N
IP4 192.168.2.228..s=Asterisk PBX
11.3.0..c=IN IP4 192.168.2.228..t=0
0..m=audio 17148 RTP/AVP 0 8
101..a=rtpmap:0 PCMU/8000..a=rtpmap:8
PCMA/8000..a=rtpmap:101 telep</font></div>
<div><font face="monospace, monospace">
hone-event/8000..a=fmtp:101
0-16..a=silenceSupp:off - - -
-..a=ptime:20..a=sendrecv..m=<wbr>video 0
RTP/AVP 103 99..
</font></div>
<div><font face="monospace, monospace">#</font></div>
<div><font face="monospace, monospace">U
+0.400478 <a href="http://192.168.2.228:5080" target="_blank">192.168.2.228:5080</a>
-> <a href="http://192.168.2.228:5060" target="_blank">192.168.2.228:5060</a></font></div>
<div><font face="monospace, monospace">
SIP/2.0 200 OK..Via: SIP/2.0/UDP
192.168.2.228;branch=<wbr>z9hG4bK9e22.<wbr>bff83e1026a667df63b99224198ef5<wbr>93.0;received=192.168.2.228..<wbr>Via:
SIP/2.0/UDP 192.168.2.200:5085;receive</font></div>
<div><font face="monospace, monospace">
d=192.168.2.200;rport=5085;<wbr>branch=z9hG4bK1247964647..<wbr>Record-Route:
<a class="m_-101634289409263049moz-txt-link-rfc2396E"><sip:192.168.2.228;lr=on;ftag=<wbr>1034946464></a>..From: <<a href="mailto:sip%3A199@192.168.2.228" target="_blank">sip:199@192.168.2.228</a>>;tag=<wbr>1034946464..To:
<<a class="m_-101634289409263049moz-txt-link-freetext">sip:102</a></font></div>
<div><font face="monospace, monospace"> @<a href="http://192.168.2.228" target="_blank">192.168.2.228</a>>;tag=<wbr>as497f35c0..Call-ID:
1571382735..CSeq: 21 INVITE..Server:
Asterisk PBX 11.3.0..Allow: INVITE, ACK,
CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
NOTIFY, I</font></div>
<div><font face="monospace, monospace">
NFO, PUBLISH..Supported: replaces,
timer..Contact: <<a href="http://sip:102@192.168.2.228:5080" target="_blank">sip:102@192.168.2.228:5080</a>>..<wbr>Content-Type:
application/sdp..Content-<wbr>Length:
312....v=0..o=root 350189084 350189084 I</font></div>
<div><font face="monospace, monospace"> N
IP4 192.168.2.228..s=Asterisk PBX
11.3.0..c=IN IP4 192.168.2.228..t=0
0..m=audio 17148 RTP/AVP 0 8
101..a=rtpmap:0 PCMU/8000..a=rtpmap:8
PCMA/8000..a=rtpmap:101 telep</font></div>
<div><font face="monospace, monospace">
hone-event/8000..a=fmtp:101
0-16..a=silenceSupp:off - - -
-..a=ptime:20..a=sendrecv..m=<wbr>video 0
RTP/AVP 103 99..
</font></div>
<div><font face="monospace, monospace">#</font></div>
</div>
<div><br>
</div>
</div>
</div>
</div>
______________________________<wbr>_________________<br>
Kamailio (SER) - Users Mailing List<br>
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</blockquote>
</div>
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______________________________<wbr>_________________<br>
Kamailio (SER) - Users Mailing List<br>
<a href="mailto:sr-users@lists.kamailio.org" target="_blank">sr-users@lists.kamailio.org</a><br>
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______________________________<wbr>_________________<br>
Kamailio (SER) - Users Mailing List<br>
<a href="mailto:sr-users@lists.kamailio.org" target="_blank">sr-users@lists.kamailio.org</a><br>
<a href="https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users" rel="noreferrer" target="_blank">https://lists.kamailio.org/<wbr>cgi-bin/mailman/listinfo/sr-<wbr>users</a><br>
</blockquote>
</div>
<br>
<fieldset class="m_-101634289409263049mimeAttachmentHeader"></fieldset>
<br>
<pre>______________________________<wbr>_________________
Kamailio (SER) - Users Mailing List
<a class="m_-101634289409263049moz-txt-link-abbreviated" href="mailto:sr-users@lists.kamailio.org" target="_blank">sr-users@lists.kamailio.org</a>
<a class="m_-101634289409263049moz-txt-link-freetext" href="https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users" target="_blank">https://lists.kamailio.org/<wbr>cgi-bin/mailman/listinfo/sr-<wbr>users</a>
</pre>
</blockquote>
<br>
</div></div><span class="HOEnZb"><font color="#888888"><pre class="m_-101634289409263049moz-signature" cols="72">--
Daniel-Constantin Mierla
<a class="m_-101634289409263049moz-txt-link-abbreviated" href="http://www.twitter.com/miconda" target="_blank">www.twitter.com/miconda</a> -- <a class="m_-101634289409263049moz-txt-link-abbreviated" href="http://www.linkedin.com/in/miconda" target="_blank">www.linkedin.com/in/miconda</a>
Kamailio Advanced Training - May 22-24 (USA) - <a class="m_-101634289409263049moz-txt-link-abbreviated" href="http://www.asipto.com" target="_blank">www.asipto.com</a>
Kamailio World Conference - May 8-10, 2017 - <a class="m_-101634289409263049moz-txt-link-abbreviated" href="http://www.kamailioworld.com" target="_blank">www.kamailioworld.com</a></pre>
</font></span></div>
<br>______________________________<wbr>_________________<br>
Kamailio (SER) - Users Mailing List<br>
<a href="mailto:sr-users@lists.kamailio.org">sr-users@lists.kamailio.org</a><br>
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<br></blockquote></div><br></div>