Hey Daniel, <div><br></div><div>I've exactly followed your point, I'll try some stuff on asterisk server to stop asking for 401 Auth to Kamailio., maybe this will eliminate the need for another INVITE with authentication params.</div>
<div><br><div>But one thing which just makes me curious is that a soft phone directly coming from a Public IP is always able to successfully CANCEL the call.</div><div><br></div><div>Anyway I'll use some brain of mine on this and let you know what resolved it, or what I'm missing.</div>
<div><br></div><div>Thanks,</div><div>Sammy<br><br><div class="gmail_quote">On Wed, Nov 30, 2011 at 5:47 PM, Daniel-Constantin Mierla <span dir="ltr"><<a href="mailto:miconda@gmail.com">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">
Hello,<br>
<br>
is the SIP trace complete?<br>
<br>
What I could find inside is:<br>
- invite from phone to kamailio<br>
- kamailio asks for authentication - 407<br>
- ack<br>
- invite with credentials, kamailio forwards to asterisk<br>
- asterisk asks for authentication - 401<br>
- ack<br>
- there is no new INVITE with credentials for kamailio and asterisk<br>
- but the phone starts sending CANCELs -- since there is no active
INVITE transaction, kamailio just drops it due to config rules<br>
- after a while asterisk starts sending like 180 ringing, then 200ok
... really strange<br>
<br>
Maybe you haven't captured all the sip traffic. If you want to use
ngrep, do on kamailio server:<div class="im"><br>
<br>
ngrep -d any -qt -W byline port 5060<br>
<br></div>
If that's all the traffic, then xlite and asterisk seems to have
some bugs - both were aware of 401 reply (asterisk generated it,
xlite sent the ACK for it) -- so no ongoing call to CANCEL by xlite,
or to answer by Asterisk (the 180, 200 replies).<br>
<br>
From kamailio point of view, if there is no INVITE following the 401
reply to xlite, there is no active invite transaction to cancel.<br>
<br>
Cheers,<br>
Daniel<div><div class="h5"><br>
<br>
On 11/30/11 12:02 AM, Daniel-Constantin Mierla wrote:
<blockquote type="cite">
Hello,<br>
<br>
I will look over it soon - since you sent pcap I couldn't look at
it directly from the email. ngrep outputs plain text which is easy
to read from email, the reason I am asking mainly for ngrep traces
since many times I am not around a computer where is convenient to
open pcap file. On the other hand, if it is a transmission problem
(at transport layer), pcap file is better.<br>
<br>
Cheers,<br>
Daniel<br>
<br>
On 11/29/11 5:07 AM, Sammy Govind wrote:
<blockquote type="cite">Hello again,
<div><br>
</div>
<div>Please see the attached wireshark trace, I tried for a
sipgrep trace but couldn't somehow. I hope this will get me
some clue on what I'm doing wrong.</div>
<div><br>
</div>
<div>This is a setup with Kamailio in front of Asterisk Servers.
Kamailio is multihomed and MS are on private IPs, all the
calls are routed to MSs and then comeback for further
dial-outs.</div>
<div><br>
</div>
<div>Please see the Continuous CANCEL requests which aren't
terminating the call.</div>
<div><br>
</div>
<div>Thanks,</div>
<div>Sammy.</div>
<div><br>
</div>
<div>
<div class="gmail_quote">On Mon, Nov 28, 2011 at 4:41 PM,
Sammy Govind <span dir="ltr"><<a href="mailto:govoiper@gmail.com" target="_blank">govoiper@gmail.com</a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Thanks
for your reply I will attach the wireshark traces as soon
as I get to my workstation.
<div><br>
</div>
<div>BR,</div>
<div>Sammy.
<div>
<div><br>
<br>
<div class="gmail_quote">On Mon, Nov 28, 2011 at
3:33 PM, 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"> Hello,<br>
<br>
send the ngrep trace of such call, from the
initial INVITE, you can use:<br>
<br>
ngrep -d any -qt -W byline port 5060<br>
<br>
The sip trace will help to see what is wrong
with that CANCEL.<br>
<br>
Cheers,<br>
Daniel
<div>
<div><br>
<br>
On 11/28/11 7:19 AM, Sammy Govind wrote: </div>
</div>
<blockquote type="cite">
<div>
<div>Anyone please help.<br>
<br>
<div class="gmail_quote">On Sat, Nov 26,
2011 at 10:39 PM, Sammy Govind <span dir="ltr"><<a href="mailto:govoiper@gmail.com" target="_blank">govoiper@gmail.com</a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"> Hello list,
<div><br>
</div>
<div>I'm using Kamailio 3.1.5 in
front of asterisk servers.
Kamailio handles all the SIP
registrations. Calls from SIP
phones are forwarded to asterisks
and then dialled out to Kamailio.</div>
<div><br>
</div>
<div>
<div><font face="'courier new',
monospace">root@SBCserver:~#
kamailio -V</font></div>
<div><font face="'courier new',
monospace">version: kamailio
3.1.5 (x86_64/linux) 76fff5</font></div>
<div><font face="'courier new',
monospace">flags: STATS: Off,
USE_IPV6, USE_TCP, USE_TLS,
TLS_HOOKS, USE_RAW_SOCKS,
DISABLE_NAGLE, USE_MCAST,
DNS_IP_HACK, SHM_MEM,
SHM_MMAP, PKG_MALLOC,
DBG_QM_MALLOC, USE_FUTEX,
FAST_LOCK-ADAPTIVE_WAIT,
USE_DNS_CACHE,
USE_DNS_FAILOVER, USE_NAPTR,
USE_DST_BLACKLIST,
HAVE_RESOLV_RES</font></div>
<div><font face="'courier new',
monospace">ADAPTIVE_WAIT_LOOPS=1024,
MAX_RECV_BUFFER_SIZE 262144,
MAX_LISTEN 16, MAX_URI_SIZE
1024, BUF_SIZE 65535, PKG_SIZE
4MB</font></div>
<div><font face="'courier new',
monospace">poll method
support: poll, epoll_lt,
epoll_et, sigio_rt, select.</font></div>
<div><font face="'courier new',
monospace">id: 76fff5</font></div>
<div><font face="'courier new',
monospace">compiled on
08:21:33 Oct 27 2011 with gcc
4.6.1</font></div>
<div><font face="'courier new',
monospace">root@SBCserver:~#</font></div>
<div><br>
</div>
<div><br>
</div>
</div>
<div>Problem: </div>
<div>When call is initiated from a
softphone and is in ringing phase,
CANCEL just don't work. I've done
some initial debugging and
the following piece of code in
main route is failing.</div>
<div><br>
</div>
<div>
<div><font face="'courier new',
monospace"># CANCEL processing</font></div>
<div><font face="'courier new',
monospace">if
(is_method("CANCEL"))</font></div>
<div><font face="'courier new',
monospace">{</font></div>
<div><font face="'courier new',
monospace">
xlog("L_NOTICE","$rm from $fu
(IP:$si:$sp) ---CAPTURED IN
MAIN---\n");</font></div>
<div><font face="'courier new',
monospace"> if
(t_check_trans()){</font></div>
<div><font face="'courier new',
monospace"> t_relay();</font></div>
<div><font face="'courier new',
monospace">
xlog("L_NOTICE","$rm from $fu
(IP:$si:$sp) ---CHECK TRANS
TRUE---\n");</font></div>
<div><font face="'courier new',
monospace"> }</font></div>
<div><font face="'courier new',
monospace">
xlog("L_NOTICE","$rm from $fu
(IP:$si:$sp) ---CHECK TRANS
FALSE---\n");</font></div>
<div><span> exit;</span></div>
<div><font face="'courier new',
monospace">}</font></div>
</div>
<div><br>
</div>
<div>Also the CANCEL fails the
has_totag() condition !</div>
<div><br>
</div>
<div>The same Call CANCEL scenario
works fine for any client on
Public IP !</div>
<div><br>
</div>
<div>Hope to get some pointers for
the solution.</div>
<div><br>
</div>
<div>Regards,</div>
<div>Sammy.</div>
</blockquote>
</div>
<br>
<br>
<fieldset></fieldset>
<br>
</div>
</div>
<pre>_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
<a href="mailto:sr-users@lists.sip-router.org" target="_blank">sr-users@lists.sip-router.org</a>
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</font></span></pre>
<span><font color="#888888"> </font></span></blockquote>
<span><font color="#888888"> <br>
<pre cols="72">--
Daniel-Constantin Mierla -- <a href="http://www.asipto.com" target="_blank">http://www.asipto.com</a>
Kamailio Advanced Training, Dec 5-8, Berlin: <a href="http://asipto.com/u/kat" target="_blank">http://asipto.com/u/kat</a>
<a href="http://linkedin.com/in/miconda" target="_blank">http://linkedin.com/in/miconda</a> -- <a href="http://twitter.com/miconda" target="_blank">http://twitter.com/miconda</a></pre>
</font></span></div>
</blockquote>
</div>
<br>
</div>
</div>
</div>
</blockquote>
</div>
<br>
</div>
<br>
<fieldset></fieldset>
<br>
<pre>_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
<a href="mailto:sr-users@lists.sip-router.org" target="_blank">sr-users@lists.sip-router.org</a>
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</blockquote>
<br>
<pre cols="72">--
Daniel-Constantin Mierla -- <a href="http://www.asipto.com" target="_blank">http://www.asipto.com</a>
Kamailio Advanced Training, Dec 5-8, Berlin: <a href="http://asipto.com/u/kat" target="_blank">http://asipto.com/u/kat</a>
<a href="http://linkedin.com/in/miconda" target="_blank">http://linkedin.com/in/miconda</a> -- <a href="http://twitter.com/miconda" target="_blank">http://twitter.com/miconda</a></pre>
<br>
<fieldset></fieldset>
<br>
<pre>_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
<a href="mailto:sr-users@lists.sip-router.org" target="_blank">sr-users@lists.sip-router.org</a>
<a href="http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users" target="_blank">http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users</a>
</pre>
</blockquote>
<br>
<pre cols="72">--
Daniel-Constantin Mierla -- <a href="http://www.asipto.com" target="_blank">http://www.asipto.com</a>
Kamailio Advanced Training, Dec 5-8, Berlin: <a href="http://asipto.com/u/kat" target="_blank">http://asipto.com/u/kat</a>
<a href="http://linkedin.com/in/miconda" target="_blank">http://linkedin.com/in/miconda</a> -- <a href="http://twitter.com/miconda" target="_blank">http://twitter.com/miconda</a></pre>
</div></div></div>
</blockquote></div><br></div></div>