<div>Hi,</div><div><br></div>I wouldn't say it a bug then, since its ok from specs poiint of view. The real issue was that we had Asterisk realtime previously which had the same extension set enabled in DB.<div><br></div>
<div>So the users with same extensions registering on Kamailio making calls via asterisk had trouble because asterisk started matching the incoming callerid-name with the ones in its realtime DB. So obviously Asterisk had to ask for AUTH since the phone is not actually registered but still sending INVITES.</div>
<div><br></div><div>So I had to disable the asterisk realtime and asterisk stopped asking for the extension specific password from kamailio and kamailio just relays the Auth request to x-lite/eyebeam. </div><div><br></div>
<div>The phones since already have gone through the authentication process maybe failing to understand why is this Authentication request required and drop it or something. This was observed in multiple vendor IP-Phones and softphones.</div>
<div><br></div><div>But at the end, I really thank you for pointing out the issue cause and it really resolved it, else I was totally lost in TM module pages to find any function which I missed to match these CANCELs.</div>
<div><br></div><div><br></div><div>Best Regards,</div><div>Sammy</div><div><br><br><div class="gmail_quote">On Fri, Dec 2, 2011 at 10:55 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,<div class="im"><br>
<br>
On 12/2/11 5:24 AM, Sammy Govind wrote:
</div><blockquote type="cite">Hello again,
<div><br>
</div><div class="im">
<div>You were right, as soon as I made changes in asterisk SIP
profile for the Kamailio proxy server and stopped the 401 Auth
from Asterisk to Kamailio the CANCELS started to work fine. <br>
</div>
</div></blockquote>
well, the 401 from asterisk is ok from specs point of view (although
many phones don't work with many challenges), but this case revealed
some bugs in asterisk as well as in xlite, both of them had
misbehavior.<br>
<br>
Cheers,<br>
Daniel<div><div class="h5"><br>
<br>
<blockquote type="cite">
<div>
<br>
</div>
<div>So the SIP flow now is: </div>
<div><br>
<div>- invite from phone to kamailio<br>
- kamailio asks for authentication - 407<br>
- ack<br>
- invite with credentials, kamailio forwards to asterisk</div>
<div>- asterisk starts processing the invite and call can be
cancelled now.</div>
<div><br>
</div>
<div><br>
</div>
<div>Thanks alot</div>
<div><br>
</div>
<div>--</div>
<div><br>
</div>
<div>Best Regards,</div>
<div>Sammy.<br>
<br>
<div class="gmail_quote">On Thu, Dec 1, 2011 at 12:01 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">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
<div>
<div><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" 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>
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><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><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>
<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>
<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>
</div>
</div>
</blockquote>
</div>
<br>
</div>
</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>
<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>