On Mon, May 24, 2010 at 2:33 PM, Klaus Darilion <klaus.mailingli...@pernau.at> wrote: > > > On 21.05.2010 23:46, Daniel-Constantin Mierla wrote: >> >> 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"?
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 > > 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 > >>> "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 >> > -- JR Richardson Engineering for the Masses _______________________________________________ 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
