Hi,
I'm working with opensips-cp and trying to get the cdrviewer to work.
The acc table has the following structure:
CREATE TABLE acc (
id INT(10) UNSIGNED AUTO_INCREMENT PRIMARY KEY NOT NULL,
method VARCHAR(16) DEFAULT '' NOT NULL,
from_tag VARCHAR(64) DEFAULT '' NOT NULL,
El Miércoles, 26 de Noviembre de 2008, Juan Backson escribió:
I am having a problem with using B2BUA - Opensips - xLite behind NAT.
When the B2BUA sends back a 200OK, it is received by the SIP client behind
NAT, but the SIP client does not response properly with ACK when the xLite
is not in
On Tuesday 25 November 2008, john li wrote:
Hi,
I agree you opinion. but now the situation is that when i start a
call, there are one and only one dialog record in the table, and when i
end the call normally, the record disappear automatically.
If you use engage_media_proxy() (and I assume
Thanks for the help. Now the opensips-cp opens.In CDRViewer tool I can display information only in certain areas of the columns in the table cdrs. I have noticed that some fields such as Username Source, Domain Source and Destination does not exist between the columns of cdrs. My cdrs table has
Hi, is possible to invoke 'failure_route[X]' after OpenSIPS has sent a
negative reply by itself to the caller? In this case the reply would be sent
to the caller but it would selected as final reply.
Well, I expect this is not possible since 'failure_route' is just invoked when
the final
El Miércoles, 26 de Noviembre de 2008, Iñaki Baz Castillo escribió:
Hi, is possible to invoke 'failure_route[X]' after OpenSIPS has sent a
negative reply by itself to the caller? In this case the reply would be
sent to the caller but it would selected as final reply.
Well, I expect this is
Thanks for the help. Now the opensips-cp opens.In CDRViewer tool I can display information only in certain areas of the columns in the table cdrs. I have noticed that some fields such as Username Source, Domain Source and Destination does not exist between the columns of cdrs. My cdrs table has
textops module.
Raghavendra D P wrote:
Hi
Route :sip:190.10.19.20, sip:45:128
I am using oopensips 1.4
How to remove fist route information
*Thanks and Regards*
*Raghavendra DP**|** Tech Mahindra*
9/7, Hosur Road, Bangalore – 560 029, India
( Office:
Hi Ali,
Is the client using TCP to connect to opensips? You need to identify a
way the client finds out that openser restarted.
Regards,
Bogdan
Ali Jawad wrote:
Hi
I am using db_mode 2 but the users get logged off once I do a restart of the
daemon.
With Regards
Ali Jawad
System
Hi Jeff,
removing the registration from the proxy is not a sane thing to do :) -
you still have a client that thinks it is still registered.
What you can do is to set max contacts to 2 and disable parallel forking
in registrar (append_branch param - 0).
Hi Juan,
I need to see the request part also to figure out if the flow through
the NAT is ok or not.
As a side note - could you check if the device behind the nat is
actually receiving the 200 OK?. Because a typical reason for a missing
ACK is a missing 200 OK.
Another question - the device
Hi
I just made some changes in the trunk to fix this issue.
I decided to change some columns.
Note you can play with what info to land in what column from opensips's
configuration file. Check out the acc module.
(http://www.opensips.org/html/docs/modules/1.4.x/acc.html)
Cheers,
Dragos
Before doing this, I'd seriously consider the problems associated with
master-master replication.
um, I don't know what they are.. but I know they are real problems. Such as
collisions in auto-incrementing data.
-Brett
On Wed, Nov 26, 2008 at 3:06 PM, Uwe Kastens [EMAIL PROTECTED] wrote:
Hi
Try use this settings:
On master
auto_increment_increment = 2
auto_increment_offset= 0
On slave
auto_increment_increment = 2
auto_increment_offset= 1
This makes sure that auto-increments do not colide between master and
slave so you can achieve switchover without conflicts because
Hi,
You may be able to tell from all of my recent posts, I'm playing with
accounting. :-)
We're not supporting redirects or diversions at the moment at the
Proxy Side - so I don't think this is relevant for us.
But typically, at what point (and what values) would you record/take
for the
Hi, sst doc has an example:
http://www.opensips.org/html/docs/modules/devel/sst.html#id2503829
if (sstCheckMin(1)) {
xlog(L_ERR, 422 Session Timer Too Small reply sent.\n);
exit;
}
Well, this is not what I see. When the request has a
El Jueves, 27 de Noviembre de 2008, Iñaki Baz Castillo escribió:
Hi, sst doc has an example:
http://www.opensips.org/html/docs/modules/devel/sst.html#id2503829
if (sstCheckMin(1)) {
xlog(L_ERR, 422 Session Timer Too Small reply sent.\n);
exit;
Hi, I'd like to comment some ideas I've read in:
http://www.opensips.org/index.php?n=Development.NewDesign
---
2.2 Application Layer
- separation between low level functionalities (transport, registration, SIP,
NAT, rr+loose_route, TM, dialog, presence, etc)
Hello
I have problem for radius accounting on opensips 1.4.2-tls.
Environment:
Debian Linux 2.6.18-6-amd64
Opensips 1.4.2
FreeRADIUS 1.1.3
MYSQL 5.0.32
Tue Nov 25 18:07:23 2008 : Info: rlm_sql_mysql: Starting connect to MySQL
server for #21
Tue Nov 25 18:07:23 2008 : Info: rlm_sql_mysql:
I am using opensips1.4
I am trying B2BUA scenario with IMS, how to route back to the caller the
same message by removing its own route address and changing Call-id
Thanks and Regards
Raghavendra DP| Tech Mahindra
9/7, Hosur Road, Bangalore - 560 029, India
* Office: +91 80 4024 3458 |
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