Hi,
I’ve recently tried out CDR-stats, and after a bumpy ride to get it going ;) ,
I’m happily surprised with it.
Especially the fraud detection mechanisms (which could use some extra’s), and
the eye-candy makes it attractive.
It is based on a mongodb which has to be filled with CDRs.
As ther
I would imagine db_update_period is only for db_mode 2 .
In db_mode : 1
I the status does not seem to update at all on SIP BYE
with db_mode : 2
the status seems to update file, but the records dont delete..
and when I use " kamctl mi dlg_list "
I can see all dialogs have "ref_count:: 1"
even thou
I think its dependent on db_update_period parameter. So any changes
happening to dialogs are likely not committed to db at max this time value.
kamailio.org/docs/modules/4.1.x/modules/dialog.html#idp125200
Thank you.
On Thu, Feb 6, 2014 at 5:35 AM, jay binks wrote:
> So I have tracked this
So I have tracked this a little further with the "dlg_list" MI command..
and what Im finding is that the ref_count does not decrement lower than 1
...
I guess ill have to start tearing apart my config to find what section
causes this, but any suggestions would be great !
On 5 February 2014 21:5
Sorry, I have never used siremis...
On 6 February 2014 14:18, arun Jayaprakash wrote:
> I think I found the link, let me give it a shot.
>
> http://siremis.asipto.com/install-mi-commands-panel/
>
> Thanks,
> Arun
>
> --
> *From:* jay binks
> *To:* arun Jayaprakash
I think I found the link, let me give it a shot.
http://siremis.asipto.com/install-mi-commands-panel/
Thanks,
Arun
From: jay binks
To: arun Jayaprakash ; Kamailio (SER) - Users
Mailing List
Sent: Wednesday, February 5, 2014 10:00 PM
Subject: Re: [SR-Users]
Thank you, that works! How do I call it from Siremis?
Regards,
Arun
From: jay binks
To: arun Jayaprakash ; Kamailio (SER) - Users
Mailing List
Sent: Wednesday, February 5, 2014 10:00 PM
Subject: Re: [SR-Users] MI commands ?
I call my "MI Commands" like
I call my "MI Commands" like this ...
kamctl mi dlg_list
On 6 February 2014 10:29, arun Jayaprakash wrote:
> Hello, what do I need to do to be able to run MI commands on Kamailio 4.0?
> When I run MI commands from Siremis no results come back. Thank you.
>
> Arun
>
>
Hello, what do I need to do to be able to run MI commands on Kamailio 4.0? When
I run MI commands from Siremis no results come back. Thank you.
Arun___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
Read the default configuration to understand how routing within Kamailio
works.
Regarding REGISTER: just remove all the REGISTER handling (and PUBLISH,
SUBSCRIBE and NOTIFY) in the config, then the REGISTERs will be routed
like all other messages.
Then take a look at the PSTN gateway feature
On 05.02.2014 11:20, Daniel Tryba wrote:
The problem I have with add_contact_alias/handle_ruri_alias is that any
kamailio in the path will parse these hints whether or not it is actually ment
for that kamailio.
handle_ruri_alias should only be used by the last proxy in the chain,
thus by the
I am a bit confused. I suspect with SBC you mean the Kamailio proxy?
On 05.02.2014 14:54, Diego Alejandro Ozuna Escalada wrote:
When a UA sends an INVITE to the SBC, the responses (back) to UA are
being sent with the correct external source IP (X.X.248.194) but if
there are retransmissions of t
Hello,
I'm trying the simplest case first. I don't understand why you are saying
that most of the people don't use mediaproxy-ng for WebRTC to WebRTC calls.
If my firewall is a restrictive one I need to use turn server and
mediaproxy-ng can do turn too? Probably I'm not seeing the big picture.
Reg
Hey,
If you're trying to connect two WebRTC endpoints with each, you don't
need any of mediaproxy-ng's magic to get it working. All the previous
replies were assuming that you were trying to connect a WebRTC endpoint
with a non-WebRTC one, which is usually what people are trying to do.
In your ca
Dear list,
I'm having a problem.
I have an SBC server with two network interfaces, one external IP
(X.X.248.194) and other internal IP (X.X.248.21). Messages are sent to the
proxy sip server through internal ip with this configuration:
route [sip-serverr] {
rewritehostport ("ip_sip_prox
On Wednesday 05 February 2014 12:04:02 Andrew Pogrebennyk wrote:
> On 02/05/2014 11:20 AM, Daniel Tryba wrote:
> > Off to figure out why uac_test incorrectly flags the onhold process as
> > needing the nat helper...
>
> Well, nat_uac_test is pretty straightforward, it does what the flags
> tell it
Im fairly use to the dialog module,
and I can not seem to get the dialogs to be removed from the dialogs table
on hangup ( BYE ).
What am I missing here..
this seems very basic and is essential to using dialog module for for
channel limiting.
any help would be great.
maybe there is an example con
On 02/05/2014 11:20 AM, Daniel Tryba wrote:
> Off to figure out why uac_test incorrectly flags the onhold process as
> needing
> the nat helper...
Well, nat_uac_test is pretty straightforward, it does what the flags
tell it to do:
http://kamailio.org/docs/modules/stable/modules/nathelper.html#id
Hi,
See presence_reginfo - Extension to Presence server for registration
info replication (RFC3680)
http://kamailio.org/docs/modules/stable/modules/presence_reginfo.html
On 02/05/2014 07:56 AM, Premchandiran wrote:
> Hi All,
>
> May I know whether kamailio supports event header with reg (EVENT:r
Hi,
I am using http_query to make some HTTP API calls. I have found that the
response of the HTTP call always has the initial 1062 characters missing.
The HTTP call is sending strings of different lengths but the initial 1062
characters are always missing.
Can anybody help me with this issue or ha
I'm experiencing some problems getting 2 Kamailios to work together.
A clients - A kamailio - B kamailio - B clients
|
the outside world
A and B client may or may not be NATed to their own kamailio instance.
A and B kamailio (with forced rtpprox
Hello,
Thank you for your detailed explication but I miss some information or I'm
unable to understand it properly. What I'm trying to do is to use
mediaproxy-ng as a turn server between 2 WebRTC endpoints (when at least
one is behind restrictive firewall). Trying to replicate what you explained
on
Dear All,
We would like to store Instant Messages to offline users which contain image instead of text.
Is it possible? Is this can be done by Msilo or via other module?
Best regards.
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users
Uff, that could have led to some ugly failures!!
Ur right! That's sth my sipp-tests don't cover...
Thx aaa lot!
greetz
Am 05.02.2014 09:17, schrieb Daniel-Constantin Mierla:
> Hello,
>
> I haven't read full thread, but I noticed in the config snippet below
> that you use $shv(...) - this is s
> listen=tcp:10.2.3.4:5080
You only have a single socket: TCP.
If you want to use websocket over TLS you need at least also:
listen=tls:10.2.3.4:5090 (or whatever port you like, eg. 443)
Further, if you talk to Asterisk with UDP, you also need a "udp" listen
statement.
regards
Klaus
__
Hello,
I haven't read full thread, but I noticed in the config snippet below
that you use $shv(...) - this is single value variable that stores the
value in shared memory. The $ci is not replaced in definition of
$shv($ci), but is used as literal token '$ci'.
You should use htable module, de
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