>A-Z route over sip is available.
>
>1) several backup routes per one destination
>2) availability of test
>3) technical support
>
>
>Plz contact me offlist for rate and further informations
>
>
>MSN:dhadda@hotmail.com
>Mohammad
>
>
>
>
>
Hello,
I've started to use the 0.9.0 release and to me it seems to work as a charm :-)
However does anybody know which SIP clients which is compliant for the PA?
I'm using kphone and it works fine, but I would like to try out some for use on Microsoft?
Messenger 5.0 does not work.
Mvh,
Helge Waastad
Hello.
Regarding to this subject. I followed some instructions for the
AVPOPS module and i have created the next lines in mysql database. This is
the description of my usr_preferences table:
+-----------+---------------+------+-----+---------+-------+
| Field | Type | Null | Key | Default | Extra |
+-----------+---------------+------+-----+---------+-------+
| uuid | varchar(64) | | | | |
| username | varchar(100) | | PRI | | |
| domain | varchar(128) | | PRI | | |
| attribute | varchar(32) | | PRI | | |
| value | varchar(128) | | PRI | | |
| type | int(11) | | | 0 | |
| modified | timestamp(14) | YES | | NULL | |
+-----------+---------------+------+-----+---------+-------+
I inserted the next data in the table:
+------+----------+--------+-----------+---------------+------+-------------
---+
| uuid | username | domain | attribute | value | type | modified
|
+------+----------+--------+-----------+---------------+------+-------------
---+
| | | | 00562 | 192.168.1.254 | 0 |
20050111152728 |
+------+----------+--------+-----------+---------------+------+-------------
---+
So what i want to do is check the first part of the RURI in the INVITE
message, for example if the INVITE is something like this
INVITE sip:005621111111@sip.mydomain.com SIP/2.0
My question is how can i match the 00562111111 with the 00562 in the field
"attribute" and therefore change the domain in the uri to:
INVITE sip:005621111111@192.168.1.254 SIP/2.0
I guess that a simple t_relay() then would send the message to the new
gateway.
If this possible?
Thanks in advance.
Best Regards,
Ricardo.-
-----Mensaje original-----
De: Ricardo Martinez [mailto:rmartinez@redvoiss.net]
Enviado el: Lunes, 10 de Enero de 2005 18:14
Para: serusers(a)lists.iptel.org
Asunto: RE: [Serusers] Dynamic Routes with SER.
Thanks Samuel,
I think the dispatcher module does not acomplish my requeriments for
now. I more interested now in the avpops module. I read some lines about
the module a few months ago for a similar problem. Is someone using this
module for something like this?. In particular i want to mantain a route
table for all my destinations and the respective terminating gateway. For
example :
Code Gateway
56 | 192.168.0.1
1 | 192.168.0.2
44 | 192.168.0.3
and so on....
This table is consulted for all the incoming request and then SER answer
with a Redirect for the particular GW.
Can someone provide me and example on how the AVPOPS module can handle this?
I will really appreciate....
Thanks in advance
Best Regards.
Ricardo.-
-----Mensaje original-----
De: Samuel Osorio Calvo [mailto:samuel.osorio@nl.thalesgroup.com]
Enviado el: Lunes, 10 de Enero de 2005 12:30
Para: dnay(a)ionosphere.net; serusers(a)lists.iptel.org; rmartinez(a)redvoiss.net
Asunto: RE: [Serusers] Dynamic Routes with SER.
Hi,
There are a couple of modules in version 0.9 which can be used for
dynamic routing depending on the desired functinality.
One is called dispatcher, and is meant to be a load balancer without
fair distribution. I think that you define a list of possible servers in
a database and the dispatcher module forwards message to one of them
(together with all within-dialog future responses and requests).
The other option is to use avpops module which provides an
attribute-value functionallity. You can use them to select the next hop
depending on the incoming SIP message parameters and the values present
in a database.
I would advise you to take a look at the documentation of both modules
to see if they suit your requirements before using the exec commands,
which are not 100% reliable and increases the memory requirements as you
have mentioned. From the exec module's readme:
1.5. Known Issues
There is currently no guarantee that scripts ever return and
stop blocking SIP server. (There is kill.c but it is not used
along with the current mechanisms based on popen. Besides that
kill.c is ugly).
Enjoy SER,
Samuel.
Unclassified.
>>> Ricardo Martinez <rmartinez(a)redvoiss.net> 01/10/05 05:22PM >>>
Darren:
Thanks for your quick answer. I was thinking in a similar
solution,
using a external program, in fact i use the exec_dset() functionality
for
something similar. I was thinking add to this solution a Redirect
Server?.
Maybe the call to the dynamic route proccess is in the Redirect
Server,
calling the exec_dset() function, so all the extra resource needed to
perform these tasks are used in the redirect server and no in my
Proxy-Registrar Server. Is just an idea..
IS this scalable maybe?
Is this posible?
Thanks in advance
Best Regards,
Ricardo Martinez.-
-----Mensaje original-----
De: Darren Nay [mailto:dnay@ionosphere.net]
Enviado el: Lunes, 10 de Enero de 2005 11:06
Para: Ricardo Martinez; 'serusers(a)lists.iptel.org'
Asunto: Re: [Serusers] Dynamic Routes with SER.
Take a look in the SER Admin guide at "Executing an External Program".
You
can write an application with perl, C, etc.. Which will process the
call
info and return a destination set. This functionality is available
using
the exec_dset function in SERs exec module.
I use this feature in our SER deployment and it works great. The only
problem I've found so far is that it definitely does increases the
resources
required per call, and hurts your scalability. Especially if you use
perl.
Try to use C if at all possible, or persistent perl at the very least.
Darren Nay
On 1/10/05 11:01 AM, "Ricardo Martinez" <rmartinez(a)redvoiss.net>
wrote:
> Hello List.
> I have a question on how to handle dynamic routes in SER. For what
> i know, SER by himself is not the best way to handle dynamic routes,
since
> every change in a route implies a change in the ser.cfg file and then
a
> restart of the service. So i'm thinking how to solve this problem
and a
was
> hoping that someone could give some advice on this issue.
> Any idea is welcome.
>
> Thanks in advance.
> BEst Regards.
>
> Ricardo Martinez.-
>
> _______________________________________________
> Serusers mailing list
> serusers(a)lists.iptel.org
> http://lists.iptel.org/mailman/listinfo/serusers
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Hi CJ,
Sorry for disturbing you but I have the same pb than you've got regarding
the serctl start with message.
Starting SER: cat: /var/run/ser.pid: No such file or directory.
SO for fun I touched a file with that name in that directory to see what
would happen. I then get:
Starting SER: PID file exists! (/var/run/ser.pid) already running?
This is all obtained by running "serctl start". Prior to running I ran "ser
-ddd -P /var/run/ser.pid" then I run serctrl, but am left being stuck in
this situation.
I'm stuck too in this situation and perhaps you have resolved this
situation.
Hope you can help me.
best regards,
Christian Thomas
Diretor Executivo
cthomas(a)canalwest.com <mailto:cthomas@canalwest.com>
www.canalwest.com
Tel +55 (48) 2107 2728 / Fax +55 (48) 333 3745
Atendimento SP (11) 5644 9889
Atendimento RJ (21) 3523 0393
Atendimento SC (48) 3027 2450
Hello,
I just discovered, that the q-value in the location table does not reflect
the q-value of the REGISTER request. Two examples:
- q-value in Contact-header = 0,8 --> location table shows 0.01
- no q-value in Contact-header (default=0)--> location table shows -1.00
Any comments on this?
Regards
Franz
Dear ser users,
Sorry to trouble you all.
I'm developing my own sip client based on the nist jain sip library.
I used SJphone with my SER and it works fine.
When I change to JSphone(sip-communicator),something strange happened.
I read the sourcecode of JSphone.
It seems that the field nonce and cnonce in the "Authentication" field are both null in JSphone.
However,I can see the values in SJphone are not null.
Both clients can register onto my server.
I wonder if this is a bug in ser.
Thank you for your reply.
Sincerely,
Kun
Dear Ser users
I was wondering if the same method to make acc in mysql, is aplicable to
accounting in radius scenario. or i need to enable something in the
radius.conf.
Best Regards
Gustavo Villegas
if a request is forwarded to another destination in failure route by
appending a new branch to it like in this example from the manual:
failure_route[1] {
# forwarding failed -- try again at another destination
append_branch("sip:nonsense@iptel.org");
log(1,"first redirection\n");
# if this alternative destination fails too, proceed to reply_route[2]
t_on_failure("2");
t_relay();
}
then translated request-uri (format o) of accounting record does not
contain this appended new uri, since the appended branch is not checked,
when o-value is selected:
case 'o':
if (rq->new_uri.len) val_arr[cnt]=&rq->new_uri;
else val_arr[cnt]=&rq->first_line.u.request.uri;
ATR(OURI);
break;
is there an existing solution to this problem or should the above code
be fixed?
-- juha
i can easily test in ser.cfg that a request comes from a particular ip
address, but how do i test that a request comes from a particular fifo
or unix socket?
-- juha