[asterisk-users] SIP Dial Command to a non-Asterisk url

[Gavin Henry]

On 23/05/07, Alex Balashov <abalashov at evaristesys.com> 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   <sasha at presidium.org>
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