Google search for SIP ALG problem to see if this is relevant for your case.
Regards, Adrian > On 13 Jan 2021, at 13:08, Mark Allen <m...@allenclan.co.uk> wrote: > > Hi all - I've been banging my head against this but not succeeding. > > Our setup... > > UAC 192.168.x.x > | > Router 5.x.x.x > | > (internet) > | > Firewall 46.x.x.x maps > | directly to > OpenSIPS 192.168.x.x Mid-registrar > | > Asterisk 192.168.x.x > > > Current situation: > - UAC can register on Asterisk via OpenSIPS > - UAC can call destination registered on Asterisk on local n/w to Asterisk box > - Destination extension rings and can pick up call > - There is no audio either way & call drops after about 30 secs (Asterisk > kills call with "Requested channel not available" because not RTP traffic is > reaching destination) > > I have tried passing audio through Mediaproxy on OpenSIPS box but with no > success. Using Wireshark I can see RTP traffic initiated at both ends, but it > doesn't reach the other end either way. > > Is there some definitive guide to setting this up correctly or are there > specific steps that I need to follow? > > _______________________________________________ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
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