Hello community,
I want to test Kamailio 3.1.1 with SIPP.
I followed many ideas that I found them on the internet but without no success.
In particular, I followed this link http://www.troubleshootingwiki.org/OpenSER#Stress_Test_The_SIP_Signaling
I can see traffic from the uac --> Kamailio --> uas ans in the other sens but no successful call.
Can you give me any ideas please.
Regards,
Rochdi
Hello community,
I want to test Kamailio 3.1.1 with SIPP.
I followed many ideas that I found them on the internet but without no success.
In particular, I followed this link http://www.troubleshootingwiki.org/OpenSER#Stress_Test_The_SIP_Signaling
I can see traffic from the uac --> Kamailio --> uas ans in the other sens but no successful call.
Can you any ideas please.
Regards,
Rochdi
Hi,
thanks to all responses. I wonder if there is a function to modify the To/From Header stateful. I mean, change the TO header in the INVITE to A and then the 200 OK (for example) reverse the change back to the previous value to B?
but the "magic" $tU is amazing...
Best regards,
Bernhard
----- Original Message -----
From: Olle E. Johansson [mailto:oej@edvina.net]
To: Alex Balashov [mailto:abalashov@evaristesys.com]
Cc: sr-users(a)lists.sip-router.org
Sent: Fri, 07 Jan 2011 11:09:41 +0100
Subject: Re: [SR-Users] Change the SIP-To-User
>
> 7 jan 2011 kl. 10.55 skrev Alex Balashov:
>
> > On 01/07/2011 04:52 AM, Carsten Bock wrote:
> >
> >> the trick, when using "uac_replace_from" is, that it will
> >> "automagically" change subsequest requests. This is required in order
> >> to work with most SIP-Endpoints.
> >> If you just want to change the "From" for the current request, that is
> fine.
> >
> > I understand. Changing To/From by a proxy has long been a thorny issue,
> in the sense that on the one hand those fields are purely cosmetic in
> strict-3261 mode, but on the other hand, changing them breaks 2543
> compatibility, and it's just not something a proxy is supposed to do.
> Well, that changed with a new rfc (put number here) about caller ID updates.
> I do not remember the proxy role, but a UA can now change From/To identities
> mid call, like after a call transfer in a pbx.
>
> /O
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users(a)lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
Hi,
My situation is that I have a number of SIP end points (server) that
support SUBSCRIBE method on various resources they hold. Primarily these
are dialog resources, but include arbitrary other resources with custom
MIME types.
I also have a large number of SIP end-points (clients) that use
SUBSCRIBE to resources held by these end-points. Some of these
end-points are standard phones (SPA-962) monitoring dialog state. Other
end points are custom clients that monitor a variety of custom resources
and MIME types.
I have got to the point where the number of SUBSCRIBE clients is making
it impractical for the SUBSCRIBE sources to support them.
I want to make a SUBSCRIBE proxy server that makes arbitrary SUBSCRIBE
requests to my SIP end points (server) and in turn accepts SUBSCRIBE
requests from my consumer SIP clients (Phones etc). In effect the server
would aggregate SUBSCRIBE requests into a single SUBSCRIBE to a target
resource.
Ideally the SUBSCRIPTION would be dynamic, generated by the SUBSCRIBE
requests from the SIP phones and other clients, but this can be static.
I've read the documentation for the RLS module in Kamailio but I can't
actually figure out what it does. It may do what I want (if only in some
database mode) but I'd appreciate advice as to whether to pursue this as
an option or go to plan-B
Regards
Jeremy
Hi,
ah, thanks. Would be good to have a uac_replace_to. Sounds like a "30min" task to someone familiar with kamailio programming :-)
BR,
Bernhard
----- Original Message -----
From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at]
To: Bernhard Suttner [mailto:bernhard.suttner@winet.ch]
Cc: sr-users(a)lists.sip-router.org
Sent: Fri, 07 Jan 2011 21:59:55 +0100
Subject: Re: [SR-Users] Change the SIP-To-User
> Bernhard Suttner wrote:
> > Hi,
> >
> > thanks to all responses. I wonder if there is a function to modify the
> To/From Header stateful. I mean, change the TO header in the INVITE to A and
> then the 200 OK (for example) reverse the change back to the previous value
> to B?
> >
> > but the "magic" $tU is amazing...
>
> If you use uac_replace_from(), the changes will stateful and fixed for
> every in-dialog requests and responses.
>
> But there is no similar feature for To header manipulation.
>
> regards
> klaus
>
>
>
Hi everyone,
Again stuck at the dispatcher module.Do anyone happen to have a
working config for kamailio with dispatcher list from mysql database ?
if not can someone please guide me on this ?
Do i need to pass module parameters for the column and the table names ?
I have the following config :
loadmodule "dispatcher.so"
if(uri=~"^sip:1[2-9][0-9]{9}@")
{
if(is_user_in("credentials","longdistance"))
{
#route(PSTN);
#exit;
ds_select_dst("1","8");
forward();
}
else{
sl_send_reply("403","No Permissions for Long
Distance calls");
exit;
};
};
--
Thank You
Amit Nepal
Dear all,
I'm trying with the Dispatcher module to terminate a call to three different
gateways serially. For each gw I use the attribute field to get the
attribute and add it as a prefix to the RU.
The problem is that the $avp(dsattrs) is returning always the first value of
the first gw that is using.
example:
destination sip:100.12.12.12:5060 attrs 111
destination sip:100.12.12.11:5060 attrs 222
destination sip:100.12.12.10:5060 attrs 333
In this example the rU will always be 111+number. I tried also to initiate
the avp in the Failure route by giving null value but still does not change
the result.
In Request route:
if(ds_select_domain("$avp(s:disp_dstgrp)", "4"))
{
$avp(s:term_prefix) = $avp(dsattrs); # avp for CDR
purpose
$rU = $avp(dsattrs) + $rU;
t_on_failure("FAIL_NATIONAL");
t_relay();
exit;
}
In Failure route:
if(!ds_next_domain())
{
t_reply("503", "Service Unavailable");
exit;
} else {
$rU = $avp(dsattrs) + $rU;
t_on_failure("FAIL_THIS");
append_branch();
if(!t_relay()) {
t_reply("503","Service Unavailable");
exit;
}
Hi,
what is the best way to replace the userpart of the SIP-To URI with the content of a pseudo variable?
Thanks for every hint.
Best regards,
Bernhard
hello
i want to install Sip Express Router on CENT OS 5.5.
the problem is that after untar the ser-0.9.6.src.tar.gz package my
system doesn't operate MAKE command ,and returns 127 errors containing
message "the system doesn't find the folder...."
is there any solution for this problem...
waiting for your response.
