Well, escalating the problem will be the right thing to do.
As a workaround on your side, you could try to enable the topo-hiding on
the dialog module, for your calls - this will take care of the contact
issue.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 02/26/2013 04:00 PM, brad smith wrote:
Bogdan,
Thanks for responding.
I am using vitelity for my upstream; I will send them a ticket. If
they fail to act, do you have any suggestions...switch carriers? any
config change?
Thanks again,
Brad
On Tue, Feb 26, 2013 at 7:58 AM, Bogdan-Andrei Iancu
<bog...@opensips.org <mailto:bog...@opensips.org>> wrote:
Hi Brad,
Thinks are a bit more complicated, it seems....
In the INVITE your opensips sends to 64.....93 IP, you have the
Contact with 192.168.1.21 (priv IP of asterisk).
When you receive the BYE from 64.....93 IP, the Route hdrs are ok
(the 2 hdrs added by opensips to reflect the interface exchange),
but the RURI is wrong - it must be the contact from the INVITE you
sent, but it seems to be the IP of your opensips - this makes
opensips to do act as strict router and not like a loose
router....and routing gets broken.
So, the 64.....93 party or some other behind it, screw up the
Contact in the your INVITE and this alters the in-dialog requests
- you should check with the upstream guys.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 02/25/2013 04:36 PM, brad smith wrote:
I just tested an outbound call (Asterisk originate) without
bridging and get the same '404 not here' if that helps.
Thanks again,
Brad
On Mon, Feb 25, 2013 at 8:01 AM, Vlad Paiu <vladp...@opensips.org
<mailto:vladp...@opensips.org>> wrote:
Hello,
Seems the incoming BYE does not have any Route headers, and
the loose_route() function returns false.
Since you have dialog support in your script, try
if (has_totag()) {
# sequential request withing a dialog should
# take the path determined by record-routing
if (loose_route() || match_dialog()) {
This way you will force matching of dialog sequential
requests that have no Route headers.
Best Regards,
Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com
On 02/24/2013 02:57 AM, brad smith wrote:
Hello,
I am currently running opensips 1.8.1 no tls. It
is
multi-homed with a public and private address.
I have a asterisk
1.8.19 in the lan that is connected to
opensips via lan
address.
*issue*
A caller calls in
and then I place an outbound call and finally
bridge the two
calls.
This works as
expected, except when the outbound caller
hangs up first the
BYE never gets back to Asterisk.
I can see the BYE
reach OpenSips but a '404 not here' is
returned to the ISP.
sip trace https://gist.github.com/5009662
opensips.cfg https://gist.github.com/5009704
thanks for your time.
_______________________________________________
Users mailing list
Users@lists.opensips.org <mailto:Users@lists.opensips.org>
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
_______________________________________________
Users mailing list
Users@lists.opensips.org <mailto:Users@lists.opensips.org>
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
_______________________________________________
Users mailing list
Users@lists.opensips.org <mailto:Users@lists.opensips.org>
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
_______________________________________________
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
