Rich Adamson wrote:
Now, I would very much like to remove the "canreinvite=no" from the
provider's definition on sip.conf, but doing so causes Asterisk to send
a re-invite to the provider pointing to a private IP. I thought that
correct localnet entries would solve this...
By changing to canreinvite=yes, are you expecting the asterisk box to
act as a router, passing rtp traffic from your sip provider through
the box to a sip phone with a private address (without passing
asterisk code in the middle of the rtp session)?
No, sorry. I'm looking for Asterisk to not issue the re-invites if the
two devices can't see each other. Think of mobile users who are often
behind the corporate firewall but also travel. I'm trying to avoid
having the media path be "user->corporate lan->pstn provider". I want it
to be "user->pstn provider".
If not to accomplish this, what are the localnet configuration entries for?
When a user is in the office, his phone registers with asterisk, and he
places calls through asterisk to the sip provider. But, when he's
out of the office, he takes his phone with him, and you are wanting
him to make use of the canreinvite=yes to allow his phone to connect
directly to the sip provider avoiding asterisk (from an rtp perspective).
Is that right?
Yes. A softphone on a laptop makes for a more believable example though.
So, is this possible? And, if not, what do the "localnet" entries provide?
I don't believe what you want is possible. The issue is in how sip
handles reinvites and nating (or lack thereof). What will further
complicate this is the laptop, when out of the office, is likely to
be behind yet a different nat box and the addressing used behind
that box may be completely different from your office nat.
In that case, what are the localnet configuration parameters for?
A.
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