We are needing to modify the configure of a currently operating
OpenSER to properly relay the SUBSCRIBE and SIP-NOTIFY messages that
are sent between Asterisk and a phone that supports BLF (like the
Snom 300 or Yealink T26). Our setup includes an OpenSER 1.2 &
Asterisk 1.4.17 in the same box. O
We're running up against a problem where engage_media_proxy seems to
handle multiple branches to the same endpoint incorrectly. We've been
getting sporadic cases of no audio and we can now reproduce it. This
is the simplest setup where it happens:
R
|
A - P ---
Whether there is a possibility to consider for acc passing through the module
b2bua calls? Use of flags(1) in opensips.cfg does not influence calls going
through the module b2bua.
Thanks
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Thanks for the info. Looks like I'm dealing with two external problems
then. A device that is not doing what it should by specifying the
transport, and this other server we are communicating with that regardless
of if it get's a UDP request, it will first try TCP on the way out. Ah the
joy of wo
I put the gwid of the gateway I into the attr col in dr_routing, then add
that column into the acc table. That meets the needs you described
-Brett
On Mar 6, 2010, at 6:04 AM, Oleg Burlacu wrote:
The routing decision in based on prefix and preference.
But the idea is to put into accouning al
The routing decision in based on prefix and preference.
But the idea is to put into accouning also the gateway id, not only the
ip/port/transport combination (because there can be identical
ip/port/transport).
I think already to assign multiple ip aliases on the gateway, and route the
calls to diff
What then would you like to make the routing decision based on?
On Sat, Mar 6, 2010 at 2:34 AM, Oleg Burlacu wrote:
> Hi all,
>
> Is there a method of getting inside the opensips config script one
> custom value for the gateway to which the call is routed? Not IP, port,
> transport - because th
Hi all,
Is there a method of getting inside the opensips config script one
custom value for the gateway to which the call is routed? Not IP, port,
transport - because they are identical in my situation.
Thanks,
Oleg
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Daniel Goepp wrote:
> We actually use record_route_preset not record_route, I would have
> presumed the logic would be the same for both though regarding this.
rr_preset() is for setting your own rr header, overriding the automatic
building of the headers. So you have to red pill - let opensips