so that Asterisk responds
correctly to these requests?
At the moment, I'm seeing Looking for s in default and then a 404 Not
Found being returned - which can't be right.
Thanks!
Alex Lake
DIGITAL MAIL LIMITED
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Hope this is the right place to report/ask for help...
Have have a 1.2.7.1 installation running reasonably happily for a while.
Thought we might give 1.4.0b3 a go. Ran it on a local test machine (that
has the single port card) and all was well. However, when I run it on a
machine with the
I'm trying to implement a dual Asterisk Box setup where our carrier will
be delivering SIP traffic to one of 2 units.
The idea is that to have a high uptime by handling traffic on box A
while box B is being upgraded, then box B takes over the traffic,
allowing box A to be upgraded.
The
It's beginning to look as though Asterisk can't send 302 responses. Is
this really the case?
I'm trying to implement a dual Asterisk Box setup where our carrier
will be delivering SIP traffic to one of 2 units.
The idea is that to have a high uptime by handling traffic on box A
while box B
I'm finding that I'm a bit disappointed that Asterisk doesn't naturally
forward the Remote-Party-ID from inbound SIP calls (where
trustedrpid=yes) to outbound SIP calls. I guess this is going to be
something we have to use SER for, unless we make our own custom build
(which I'm reluctant to
You've not said much about your firewall setup. I presume you've opened
up 5060 and RTP ports?
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I guess you could post your config files here and hope that someone
feels inclined to look them over! ;-)
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I'm going to dip my toe in the water to help out here, although I'm just
a newbie...
It looks to me as though your x-lite is coming in and being assigned the
sip context, which includes just the right to call internal destinations.
(Bizarrely?) Your default context seems to allow everything
Thanks for that. We did track it down to a problem with native bridging.
In this case, Asterisk assumed that the VPN was publicly accessible -
but it isn't!
The fix we've found is to setup all VPN-based sip devices with
canreinvite=no, but I'm not sure if this is the best way to do that.
I've got a one-way audio problem, but I've looked through a few
documents on the subject and I'm not sure that it's the same issue.
User A calls a local Asterisk user B via a public SIP gateway
(voiptalk.org) using (sip:[EMAIL PROTECTED])
B is connected to the Asterisk server via VPN
B is
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