that seems to work, (i did not add anything new for (has uri==myself)) i'm not sure yet why this is working, so i will study it a bit more. if this doesn't bring up new bugs, then my whole problem seems to be solved.
2014-06-12 9:17 GMT+02:00 pa...@eremina.net <eremina....@gmail.com>: > Hi again. > You can try to use topology hiding in you kamailio( don't forget add some > code for message which has uri==myself it present in docs). > > I use it and ack processing well. > > I think it's because some sip servers can't work with SIP proxy. it > created only for PBX. > > > 2014-06-10 18:16 GMT+06:00 Gijs Kwakkel <kwakkel1...@gmail.com>: > >> client1 <> provider <> kamailio <> asterisk <> client2 >> >> when client1 calls client2, everything seems to work. sounds works. >> after a few seconds the call gets terminated. >> >> after tcpdump syslog and config checks i figured out that asterisk sends >> 200 OK through kamailio to the provider, after this the provider sends a >> ACK to kamailio, kamailio however sends this ACK to itself and not to >> asterisk. >> >> I added the config, syslog and wireshark output (converted) >> >> >> _______________________________________________ >> 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 >> >> > > _______________________________________________ > 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 > >
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