On 23/05/07, Alex Balashov <[EMAIL PROTECTED]> wrote:
Gavin,
Hi.
Does the Asterisk server's route to 192.168.45.183 traverse a firewall or router that may be blocking non-SIP ports that are dynamically allocated?
Nope, all internal.
SDP -- part of the SIP INVITE transaction payload -- negotiates arbitrary ports between the two endpoints for actually passing media. If these are being dropped somewhere along the way, you'll have no audio in one or more directions of the call path.
Yeah, I understand that. It looks like * it not sending an ACK back to the other SIP server, well it is, but not on the same port.
Best thing to do is to is a packet capture on the Asterisk server and filter on 192.168.45.183 to verify that you're seeing bidirectional media, from and to that host. Chances are something will be missing.
Yeah, we've done this, but it seems to be not replying to the correct port.
Of course, it could be a non-IP problem of some sort as well, perhaps even something fairly obvious.
Hmmm, hope so. This is the danger of too much knowledge.
-- Alex -- Alex Balashov <[EMAIL PROTECTED]> _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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