25 May
2010
25 May
'10
1:57 a.m.
On Mon, May 24, 2010 at 2:33 PM, Klaus Darilion
<klaus.mailinglists(a)pernau.at> wrote:
On 21.05.2010 23:46, Daniel-Constantin Mierla wrote:
sipp send invite to kamailio which forwards to asterisk in dispatcher
list, asterisk responds
back to kamailio which forwards that response back to sipp and I get the error:
SIP/2.0 481 Call leg/transaction does not exist on sipp.
So I think this is not supposed to work like I want it to. The
dispatcher module is for
stateless processing only, so even though I have RR and TM functions
in my routing script
it does not act properly. I don't think I can use dispatcher for what
I want, which is a
stateful load balancer.
I am looking at 3.0 and carrierroute or lcr module.
Thanks.
JR
Hello,
On 5/21/10 10:47 PM, JR Richardson wrote:
>
> Hi All,
>
> I'm doing some testing with kamailio 1.5:
>
> kamailio1:/etc/kamailio# kamailio -V
> version: kamailio 1.5.4-notls (i386/linux)
> flags: STATISTICS, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST,
> SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
> MAX_URI_SIZE 1024, BUF_SIZE 65535, PKG_SIZE 4194304
> poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
> svnrevision: 2:6005M
> @(#) $Id: main.c 5608 2009-02-13 16:48:17Z henningw $
> main.c compiled on 10:14:11 May 18 2010 with gcc 4.3.2
>
> Using dispatcher module trying to load balance SIP calls across some
> Asterisk servers. I have it working fine when I test in this
> scenario:
>
> sip phone dial out><asterisk><kamailio><round robin to several
> asterisk servers
>
> This works stateful and stateless, handles everything gracefully.
>
> This scenario is giving me fits:
>
> sipp dial out><kamailio><round robin to several asterisk servers
>
> I get retransmits on every call back to sipp with errors like
what means "call back"?
You are operating sipp in uac mode - thus it is not capable of receiving
requests.
Maybe Asterisk is send reINVITEs which are not handled correctly by sipp.
set canreinvite=no in sip.conf (Asterisk)
regards
klaus
--
JR Richardson
Engineering for the Masses
"Discarding message which can't be mapped to a known SIPp call" and
"SIP/2.0 481 Call leg/transaction does not exist"
This happens with kamailio setup stateful or stateless. I'm wondering
if sipp is the problem or just doesn't play well with kamailio?
I've kept the config as simple as possible for testing, it is listed
here http://pastebin.com/BZ8hJvJv
Here is my sipp usage:
sipp -sn uac 10.10.12.53 -i 10.10.14.97 -s 55 -d 7000 -l 10 -r 1
-trace_err
Any insight would be appriciated.
the problem is in your sipp scenario. The uac calls do not map to uas.
kamailio does not reply 481, check the uas scenario, that is the one
that sends back the 481.
Cheers,
Daniel