Hello,
On 12/2/11 5:24 AM, Sammy Govind wrote:
Hello again,
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.
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.
Cheers,
Daniel
So the SIP flow now is:
- invite from phone to kamailio
- kamailio asks for authentication - 407
- ack
- invite with credentials, kamailio forwards to asterisk
- asterisk starts processing the invite and call can be cancelled now.
Thanks alot
--
Best Regards,
Sammy.
On Thu, Dec 1, 2011 at 12:01 PM, Sammy Govind <govoi...@gmail.com
<mailto:govoi...@gmail.com>> wrote:
Hey Daniel,
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.
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.
Anyway I'll use some brain of mine on this and let you know what
resolved it, or what I'm missing.
Thanks,
Sammy
On Wed, Nov 30, 2011 at 5:47 PM, Daniel-Constantin Mierla
<mico...@gmail.com <mailto:mico...@gmail.com>> wrote:
Hello,
is the SIP trace complete?
What I could find inside is:
- invite from phone to kamailio
- kamailio asks for authentication - 407
- ack
- invite with credentials, kamailio forwards to asterisk
- asterisk asks for authentication - 401
- ack
- there is no new INVITE with credentials for kamailio and
asterisk
- but the phone starts sending CANCELs -- since there is no
active INVITE transaction, kamailio just drops it due to
config rules
- after a while asterisk starts sending like 180 ringing, then
200ok ... really strange
Maybe you haven't captured all the sip traffic. If you want to
use ngrep, do on kamailio server:
ngrep -d any -qt -W byline port 5060
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).
From kamailio point of view, if there is no INVITE following
the 401 reply to xlite, there is no active invite transaction
to cancel.
Cheers,
Daniel
On 11/30/11 12:02 AM, Daniel-Constantin Mierla wrote:
Hello,
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.
Cheers,
Daniel
On 11/29/11 5:07 AM, Sammy Govind wrote:
Hello again,
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.
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.
Please see the Continuous CANCEL requests which aren't
terminating the call.
Thanks,
Sammy.
On Mon, Nov 28, 2011 at 4:41 PM, Sammy Govind
<govoi...@gmail.com <mailto:govoi...@gmail.com>> wrote:
Thanks for your reply I will attach the wireshark traces
as soon as I get to my workstation.
BR,
Sammy.
On Mon, Nov 28, 2011 at 3:33 PM, Daniel-Constantin
Mierla <mico...@gmail.com <mailto:mico...@gmail.com>> wrote:
Hello,
send the ngrep trace of such call, from the initial
INVITE, you can use:
ngrep -d any -qt -W byline port 5060
The sip trace will help to see what is wrong with
that CANCEL.
Cheers,
Daniel
On 11/28/11 7:19 AM, Sammy Govind wrote:
Anyone please help.
On Sat, Nov 26, 2011 at 10:39 PM, Sammy Govind
<govoi...@gmail.com <mailto:govoi...@gmail.com>> wrote:
Hello list,
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.
root@SBCserver:~# kamailio -V
version: kamailio 3.1.5 (x86_64/linux) 76fff5
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
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE
262144, MAX_LISTEN 16, MAX_URI_SIZE 1024,
BUF_SIZE 65535, PKG_SIZE 4MB
poll method support: poll, epoll_lt, epoll_et,
sigio_rt, select.
id: 76fff5
compiled on 08:21:33 Oct 27 2011 with gcc 4.6.1
root@SBCserver:~#
Problem:
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.
# CANCEL processing
if (is_method("CANCEL"))
{
xlog("L_NOTICE","$rm from $fu (IP:$si:$sp)
---CAPTURED IN MAIN---\n");
if (t_check_trans()){
t_relay();
xlog("L_NOTICE","$rm from $fu
(IP:$si:$sp) ---CHECK TRANS TRUE---\n");
}
xlog("L_NOTICE","$rm from $fu (IP:$si:$sp)
---CHECK TRANS FALSE---\n");
exit;
}
Also the CANCEL fails the has_totag() condition !
The same Call CANCEL scenario works fine for
any client on Public IP !
Hope to get some pointers for the solution.
Regards,
Sammy.
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users
mailing list
sr-users@lists.sip-router.org
<mailto:sr-users@lists.sip-router.org>
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierla --http://www.asipto.com
Kamailio Advanced Training, Dec 5-8,
Berlin:http://asipto.com/u/kat
http://linkedin.com/in/miconda -- http://twitter.com/miconda
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org <mailto:sr-users@lists.sip-router.org>
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierla --http://www.asipto.com
Kamailio Advanced Training, Dec 5-8, Berlin:http://asipto.com/u/kat
http://linkedin.com/in/miconda -- http://twitter.com/miconda
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org <mailto:sr-users@lists.sip-router.org>
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierla --http://www.asipto.com
Kamailio Advanced Training, Dec 5-8, Berlin:http://asipto.com/u/kat
http://linkedin.com/in/miconda -- http://twitter.com/miconda
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierla -- http://www.asipto.com
Kamailio Advanced Training, Dec 5-8, Berlin: http://asipto.com/u/kat
http://linkedin.com/in/miconda -- http://twitter.com/miconda
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
