On 04/11/14 20:04, Glenn Larsen wrote: > On Tue, Nov 4, 2014 at 6:17 PM, Julien <julien.linph...@gmail.com> wrote: >> A user connected at eth0 shall be able to get the presence status of users >> on eth1 and users shall be able to call each other. >> >> What I did so far: >> I added listen statements for both interfaces to the Kamailio config file: >> -listen=udp:eth0_IP:5060 >> -listen=udp:eth1_IP:5060 > Have you done similar configuration for the RTP proxy? Since Kamailio > will setup the call, but will not "carry" the audio. If your RTP proxy > has been configured, have you updated the kamailio config to use the > RTP proxy? > > See example 1.14 on this page: > http://kamailio.org/docs/modules/4.2.x/modules/rtpproxy.html#idp84376 > A tutorial I wrote for bridging IPv4 and IPv6 can be useful to go through:
- http://kb.asipto.com/kamailio:kamailio-mixed-ipv4-ipv6 It show bridging of sip and rtp between not directly routable networks. Instead of doing kamailio.cfg rtpproxy_manage() decisions on af (address family), it has to be done on dst_ip (local ip on which the packet was received) and rtp proxy started with "-l ip1/ip2" instead of "-l ip1 -6 /ip2". Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Nov 24-27, Berlin - http://www.asipto.com _______________________________________________ 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
