Now I started over with a simpler configuration without NAT traversal stuff,
and the all dialogs seem to get cleared correctly. Now I have to cross check my
two configurations...
Is there something I've to take care for if I want to do NAT traversal and
channel limiting with dialog profiles at
Richard,
Branch routing is something I've never been able to completely wrap my mind
around. I'm not using any branch routing in my current configuration, at
least to the best of my knowledge. I'll read up on the different types of
routes to see if I can figure this one out. Do you have any qui
Hi Sergio,
I think I also want OpenSIPS to handle UA registrations as well, so I can remove
that burden from the Asterisk boxes. Then OpenSIPS would rewrite the
URI to send it to Asterisk. (Though maybe this isn't possible or makes things
infinitely more complicated?)
In addition, almost 100% of my
Hi James.
Your case sounds like you would use your OpenSIPS just as a proxy. Asterisk
would be your UAS.
If that is your situation, you could start from the example config file
which is installed with source code;
>From that file you can take validations for SIP signaling; you would rewrite
uri
Hi,
Does anyone have some good examples of an OpenSIPs
configuration that integrates with Asterisk?
Essentially I want to use OpenSIPs as the UA, but still run all
the calls through Asterisk (for dialplan, etc..)
I've tried searching for some good examples, but I haven't found any
for Asterisk yet
Add the headers in branch routes. Headers added in primary routing
can't be removed in later processing.
On Mar 6, 2009, at 10:40 PM, Jeff Pyle wrote:
> Hello,
>
> I’m using serial forking to send requests to multiple PSTN
> carriers. Some
> of these carriers want P-Asserted-Identity/Priva
