On 18.05.2013 22:26, Ján Hrnko wrote:
Hi, I have a problem when using uri_db module. When I leave everything
on default and not specify modparam "uri_db", "db_url", then it will not
connect to the database. In the var/log/syslog I have seen that it tries
to connect to database okmaialio instead
Because there is no logic implement removing the header automatically.
So you have to call remove_hf("Content-Type") afterwards.
regards
Klaus
On 18.05.2013 08:00, Konstantin M. wrote:
Hello,
I'm using a very ugly code to test remove_body():
request_route
{
remove_body();
$r
Hello,
I looked at the asterisk config, looks like you have not configured that.
You need to add a dialplan in the extensions config and trunks in the sip
confiig.
If you want to make the dialplan realtime as well, you should add the dialplan
to the db.
Success.
Gertjan
-Original messa
Advanced Training, San Francisco, USA - June 24-27, 2013
* http://asipto.com/u/katu *
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I would appreciate some help on the following questions I have:
- If I use TLS mutual authentication, do I still need a subscriber password or
the TLS successful mutual session setup will assume that the client is
"trusted" so it can register what it is asking to register?
- For large deploym
Hello,
maybe the commit log helps a bit:
-
http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=commit;h=eb321e3ffeb7ff474693bc035c8af0915a745b4f
Cheers,
Daniel
On 5/20/13 7:06 PM, Jignesh Gandhi wrote:
Hello,
I read in the documentation that Kamailio SCTP supports multi homing.
D
Hello,
I read in the documentation that Kamailio SCTP supports multi homing.
Does anyone know how I can configure this?
Thanks in advance,
--Jignesh Gandhi
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Hello,
if you want to send all calls that arrive to kamailio having the prefix
01 to freeswitch:
if($rU =~"^01") {
$ru = "sip:" + $rU + "@__FREESWITCHIP__";
route(RELAY);
exit;
}
Be sure calls are authenticated at that point and, if needed, the call
is not actually coming from fr
Hello,
On 5/20/13 12:47 PM, johnc wrote:
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Hi,
I updated this config to work with version 4. See below. I have calls
working over TLS between jitsi sip clients registered to the same proxy
of either one of two proxies I have built using this updated
Hello,
On 5/20/13 6:12 AM, Jignesh Gandhi wrote:
Hello,
I recently started using SCTP relay feature of Kamailio and am
receiving SCTP-SIP and
relaying it UDP-SIP and vice versa.
What happens , if an SCTP association is broken from the distant end
in the middle of the call,
is there a way
This works for me fine:
loadmodule "msilo.so"
modparam("msilo","db_url",DBURL)
modparam("msilo","from_address","sip:regist...@sip.stribogkonsult.com")
modparam("msilo","contact_hdr","Contact:
regist...@sip.stribogkonsult.com:5060;msilo=yes\r\n")
modparam("msilo","content_type_hdr","Content-Type:
On 20 May 2013 04:07, zhengyw wrote:
> Hi Barry:
>This issue have not been resolved after following by your method that
> modifed the video1_sipregs table struct,attachment is table info and
> asterisk log.
> Can you help me with this problem? thank you very much!
>
>
I can see that your sipr
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Hi,
I updated this config to work with version 4. See below. I have calls
working over TLS between jitsi sip clients registered to the same proxy
of either one of two proxies I have built using this updated config. I
can make calls to/from clients o
Dear All,
Is there any update regarding the aforementioned issue?
Best regards.
Dear All,
We would like to store SIP text Messages when the destination Subscriber is Offline.
We have insert into kamailio cfg file the configuration lines below.
Unfortunately storing message
Hi
Version Kamailio v4.0 + Siremis installed on Debian Wheezy via apt-get
install
I want to use Kamailio as a proxy edge register to our network.
I have installed Kamailio and freeswitch.
I can register on Kamailio but I can't route a call from my sip client from
Kamailio to freeswi
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