Hi,
Thanks for your quickly support, but after recompile with the lastest git
branch 3.0 version, I still meet the same problem.
>From my xmpp client, i chat to sip client (e.g 101*sip.htk@xmpp.htk.com),
>the xmpp server looking for its database and not found user 101*sip.htk.com.
>So,
Hi Klaus,
I am able to watch the video on demand content, but rtsp stream I couldn't.
The error I am getting in asterisk is
no media found.
Thanks
Jack
On Thu, Oct 14, 2010 at 4:24 PM, jack nicolson
wrote:
> Hi Klaus,
>
> Yes the video is playing fine when I tried asterisk directly, I am usin
Hi Nicolas,
On 10/14/10 11:58, "Nicolas Rüger" wrote:
Hey all,
thanks for all your advices.
I changed the route as you told me, but still nothing in DB table "dialog"... :(
You should see something in the DB only as long as the call is in progress.
The XLOG message you suggested (see belo
Hi,
That solved the problem with OCS. :)
/Morten
On Thu, Oct 14, 2010 at 6:34 PM, Ovidiu Sas wrote:
> The spec requires just lr. There were some buggy clients that
> couldn't do just lr and therefor lr=on was introduced.
> If it works with lr, then don't enable lr=on (which is disabled by defa
Hey all,
thanks for all your advices.
I changed the route as you told me, but still nothing in DB table "dialog"... :(
The XLOG message you suggested (see below) says the following:
INFO:
Hello,
take the latest git version from branch kamailio_3.0. There were some
fixes to xmpp module after 3.0.1.
Cheers,
Daniel
On 10/14/10 7:41 PM, Huy Nguyen wrote:
Hi all,
I'm trying to integrate xmpp module to kamailio 3.0.1.
But i wondered about the server mode, if not use the local xm
Hi all,
I'm trying to integrate xmpp module to kamailio 3.0.1.
But i wondered about the server mode, if not use the local xmpp server, how to
establish the s2s connection with the other xmpp server ? I think it must have
the running port 5269 on two xmpp servers to setup the connection.
Thank
Timo & Ovidiu,
Yes, you are correct. Thanks!
So, hopefully this is better:
route {
# Initial invite only
if (is_method("INVITE") && (! has_totag() ) ) {
dlg_manage();
}
if (is_method("BYE")) {
xlog ("L_INFO", "Completed $dlg(from_uri) to $dlg(to_
dlg_manage() is to be called _only_ for initial INVITEs. Once the
dialog is created, all the subsequent in-dialog requests will be
automatically handled:
http://www.kamailio.org/docs/modules/3.1.x/modules_k/dialog.html#id2715451
Calling dlg_manage() on requests that are not generating a SIP dialo
On 10/14/10 09:41, Ovidiu Sas wrote:
For CDR purposes, the duration of a call is for how long that call was
in conversation, which is reflected by $DLG_lifetime:
Thanks Ovidiu!
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailin
For CDR purposes, the duration of a call is for how long that call was
in conversation, which is reflected by $DLG_lifetime:
http://www.kamailio.org/docs/modules/3.1.x/modules_k/dialog.html#id2745926
There's no need for 'fancy' duration computation like $var(elapsed) =
( $Ts - $dlg(start_ts) );
R
Hey,
On 14.10.2010 18:22, Nathan Angelacos wrote:
> Actually, you want dlg_manage() to be called for every message, not just
> "INVITES"
>
> route {
> dlg_manage();
> }
>
>>
>>[...]
Are you sure dlg_manage() is really required to be called on every
message of the dialog (or suppo
The spec requires just lr. There were some buggy clients that
couldn't do just lr and therefor lr=on was introduced.
If it works with lr, then don't enable lr=on (which is disabled by default):
modparam("rr", "enable_full_lr", 0)
http://www.kamailio.org/docs/modules/3.1.x/modules_k/rr.html#id2805
Hi Nicolas,
On 10/14/10 08:50, "Nicolas Rüger" wrote:
Hello,
I'm trying to get dialog module working..
...my problem is that nothing is stored in the database in table "dialog" yet.
I'm looking for CDRs there...
Can anyone please help me, so that I can see the CDRs in the database!???
The d
Hi,
I was mistaken. This is not the problem. OCS kan handle r-r with multiple entry.
This one works OK - The OCS sends a PRACK
SIP/2.0 180 Ringing
Via: SIP/2.0/TCP x.x.42.177:65371;rport=65371;branch=z9hG4bK991996e6
From:
;epid=E1A3C38520;tag=74504cfc7a
To: ;tag=1c564710455
Call-ID: 707d32ae-ac
Hi,
On 14.10.2010 17:50, "Nicolas Rüger" wrote:
> I'm trying to get dialog module working..
>
> ...my problem is that nothing is stored in the database in table "dialog"
> yet. I'm looking for CDRs there...
>
> I inserted in kamailio.cfg:
>
> [...]
>
> loadmodule "dialog.so"
>
> [
Hello,
I'm trying to get dialog module working..
...my problem is that nothing is stored in the database in table "dialog" yet.
I'm looking for CDRs there...
Can anyone please help me, so that I can see the CDRs in the database!???
I inserted in kamailio.cfg:
[...]
loadmodule "dialog.
Sergey Okhapkin writes:
> Is Record-Route: a,b,c
>
> analog of
>
> Record-Route: a
> Record-Route: b
> Record-Route: c
yes. new rr entries are added in front of the list.
-- juha
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
As long as the order of the Record-route headers is preserved,
multiple Record-Route headers can be compacted into a single header
and kamailio is able to deal with it.
Please post a trace of a call exposing the issue.
Regards,
Ovidiu Sas
On Thu, Oct 14, 2010 at 10:40 AM, Morten Isaksen wrote:
>
Is Record-Route: a,b,c
analog of
Record-Route: a
Record-Route: b
Record-Route: c
or
Record-Route: c
Record-Route: b
Record-Route: a
?
On Thursday 14 October 2010, Juha Heinanen wrote:
> Morten Isaksen writes:
> > When OpenSer sends the message to Kamailio the recourd-routes look like
> > this
Morten Isaksen writes:
> When OpenSer sends the message to Kamailio the recourd-routes look like this:
>
> Record-Route:
> Record-Route:
>
> But when the message comes back from Kamailio it is:
> Record-Route:
> ,,
>
> OpenSER forwards the message to the OCS with Record-route unchanged
> and
I found a place where there it had: "select * from subscriber where
username = $fU" - so the $fU was missing the ' around it: '$fU' -
problem solved.
Thanks for your input!
On Thu, Oct 14, 2010 at 10:36 AM, Anders wrote:
> the 'x' is the same as $fU or src_user, and looking through the
> con
Hi,
I think I have found a issue with recourd-route in Kamailio 3.0.3.
My old setup was:
Microsoft OCS <--> (TCP) OpenSER (UDP) <---> (UDP) Mediant 2000 (ISDN).
That worked fine.
Now I have inserted a new server in this setup.
Microsoft OCS <--> (TCP) OpenSER (UDP) <---> (UDP) Kamailio (UDP)
the 'x' is the same as $fU or src_user, and looking through the
config I don't see these with problems anywhere.
On Thu, Oct 14, 2010 at 10:28 AM, Uriel Rozenbaum
wrote:
> Probably you filled some modparam with 'x' or you are using SQL
> OPS and sent a Query with a 'x' column that doe
Probably you filled some modparam with 'x' or you are using SQL
OPS and sent a Query with a 'x' column that does not exist.
On Thu, Oct 14, 2010 at 11:00 AM, Anders wrote:
> Hi,
>
> I was wondering if a message like this below in the syslog contains
> any lead as to where to look for the
Hi all
I'm testing the htable module at Kamailio 3.1. Currently it's an extremely
simple table with a single row loaded from a postgres DB.
These are the relevant lines in the configuration file:
modparam("htable", "db_url", "postgres://user:passw...@localhost
:5432/database")
modparam("htable",
Hi,
I was wondering if a message like this below in the syslog contains
any lead as to where to look for the problem. The [23582] is a lead to
where the problem occured?
Oct 14 13:34:52 server1 /usr/local/sbin/kamailio[23582]:
ERROR:db_mysql:db_mysql_submit_query: driver error on query: Unknown
c
Hello,
Marius Zbihlei has patched the userblacklist module, so that it can handle
characters as well (NOT in main release 3.1 yet). Thanks Marius.
Therefore it might now be possible to filter general SIP-URIs!??
My idea is simple and described here. Please give some Feedback!!!
As the "prefix"
Hi Klaus,
Yes the video is playing fine when I tried asterisk directly, I am using
mp4play and rtsp method to play the content,
Thanks
Jack
On Thu, Oct 14, 2010 at 12:51 PM, Klaus Darilion <
klaus.mailingli...@pernau.at> wrote:
>
>
> Am 13.10.2010 16:46, schrieb jack nicolson:
>
> Hi,
>>
>>
Hello Marius,
oh sorry...
I did now "cherry-pick" both patches and it compiled well this time.
I just checked the functionality and it seems working alright now :)
Thank you very much for all the effort you put in this fix and your patience
with all my questions...
Regards,
Nicolas
P.S.
Jon Bonilla (Manwe) writes:
> > Sure, I just meant a very small change in a module (regex). I use it
> > in production for long months with no problems, so I wonder if it
> > could be commited in 1.5.5 so I wouldn't need to modify the sources in
> > new deployments :)
> >
>
> I think that's a ba
El Thu, 14 Oct 2010 10:26:41 +0200
Iñaki Baz Castillo escribió:
> 2010/10/14 Daniel-Constantin Mierla :
> >> Hi, is it possible to commit some "featured" changes for 1.5.5? or
> >> just security fixes?
> >
> > stable branches are for bug fixes. New features can be committed in
> > development bra
On 10/14/2010 12:34 PM, "Nicolas Rüger" wrote:
Hello,
thank you, but "make all" reports an error this time:
userblacklist.c: In function ‘check_user_list’:
userblacklist.c:290: error: ‘match_mode’ undeclared (first use in this
function)
userblacklist.c:290: error: (Each undeclared ident
Hello,
thank you, but "make all" reports an error this time:
userblacklist.c: In function ‘check_user_list’:
userblacklist.c:290: error: ‘match_mode’ undeclared (first use in this
function)
userblacklist.c:290: error: (Each undeclared identifier is reported only once
userblacklist.c:290: er
On 10/14/2010 11:36 AM, "Nicolas Rüger" wrote:
Hello Marius,
thanks for the advice. I forgot about the modparam. It seems working now as I
don't get the error any more :)
BUT blacklisting does not work for mesee:
In kamailio.cfg I added the following in routing logic:
if (is_method("IN
Hello Marius,
thanks for the advice. I forgot about the modparam. It seems working now as I
don't get the error any more :)
BUT blacklisting does not work for mesee:
In kamailio.cfg I added the following in routing logic:
if (is_method("INVITE")) {
if (!check_user_blacklist("$rU", "$
2010/10/14 Daniel-Constantin Mierla :
>> Hi, is it possible to commit some "featured" changes for 1.5.5? or
>> just security fixes?
>
> stable branches are for bug fixes. New features can be committed in
> development branches, otherwise will introduce instability to stable
> branches for people re
On 10/14/10 10:19 AM, Iñaki Baz Castillo wrote:
2010/10/14 Daniel-Constantin Mierla:
Hello,
Based on our project policy to support officially the last two stable
branches, I was thinking of packaging kamailio 1.5.5 to mark its end. New
fixes will go to svn if it is the case, but no new rele
2010/10/14 Daniel-Constantin Mierla :
> Hello,
>
> Based on our project policy to support officially the last two stable
> branches, I was thinking of packaging kamailio 1.5.5 to mark its end. New
> fixes will go to svn if it is the case, but no new release in 1.5.x series
> will be done.
Hi, is
On 10/13/2010 09:57 PM, "Nicolas Rüger" wrote:
Hey Marius,
thnaks again. I now did the following:
git clone git://git.sip-router.org/sip-router kamailio
cd kamailio
git checkout -b 3.1 origin/3.1
git cherry-pick 2f8f8e58
make FLAVOUR=kamailio include_modules="db_mysql perl" cfg
m
Hello,
Based on our project policy to support officially the last two stable
branches, I was thinking of packaging kamailio 1.5.5 to mark its end.
New fixes will go to svn if it is the case, but no new release in 1.5.x
series will be done.
Also, since testing for 3.1.0 brought some fixes to
Michel,
On 10/14/2010 03:06 AM, michel freiha wrote:
*Dear All,
I'm planning to use Kamailio in a high network traffic environment...
I'm sure that it can handle several thousand of calls when using SIP
over UDP but I have a doubt regarding TLS/TCPusage with Kamailio...I
saw on the website tha
Hello,
I think I tried everything to get asterisk let registration be allowed. I
set insecure=port,invite on the main trunk from kamailio, for every friend
(every user) which register is forwarded..Asterisk keeps sending 401
Unauthorized. Is there any trick about it?
On Wed, Oct 13, 2010 at 1:59
Am 13.10.2010 16:46, schrieb jack nicolson:
Hi,
I am trying to use kamailio as SIP server, so that it receive SIP invite
and rewrite to asterisk, the set up is working fine for audio stuff,
however for video it is not working properly,
For rtsp stream I am getting no media found in asterisk
*Dear All,
I'm planning to use Kamailio in a high network traffic environment... I'm
sure that it can handle several thousand of calls when using SIP over UDP
but I have a doubt regarding TLS/TCPusage with Kamailio...I saw on the
website that the latest version fixed that BUG that I had bad experi
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