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 log
Could help me in it.
Thanks
Jack
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) <-->
(UDP) Mediant 2000 (ISDN).
When OpenSer sends the message to Kamailio the recourd-routes look like this:
Record-Route: <sip:x.x.248.20;r2=on;lr;ftag=3d9e7d131b>
Record-Route: <sip:x.x.248.20;transport=tcp;r2=on;lr;ftag=3d9e7d131b>
But when the message comes back from Kamailio it is:
Record-Route: <sip:x.x.248.7;lr=on;ftag=3d9e7d131b;did=584.1a683a45>,<sip:x.x.248.20;r2=on;lr;ftag=3d9e7d131b>,<sip:x.x.248.20;transport=tcp;r2=on;lr;ftag=3d9e7d131b>
OpenSER forwards the message to the OCS with Record-route unchanged
and the OCS gets confused and does not reply.
I am not that strong in the SIP RFC's. Is it part of the SIP standard
to compact the record-route into one line?
Is it a bug in Kamailio or is it just a parameter that needs to be changed?
Any help will be much appreciated.
--
Morten Isaksen
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.
Thanks,
Huy Nguyen
www.htk-inc.com
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
column 'xxxxx' in 'where clause'
And concerning this specific issue, where I put 'xxxxx' appears the
src_user. Problem in kamailio.cfg, right?
Thanks!
//Anders
Hi all!
I need to implement a kind of Location Server. The idea behind is to
process the REGISTER messages and analyze the source IP address of the
packet and reply with the "302 Moved Temporarily" message.
What I want to achieve - provide the SIP client with the "nearest"
point to use for registration and further communications.
The possible approach is:
1. make some list of available registrars/SBC's mapped to appropriate
IP (sub)networks.
2. configure SER to process REGISTER messages and reply with "302
Moved Temporarily" having the IP or name of "nearest" point in Contact
field.
Does anybody have any ideas on possible implementations (with SER or
perhaps with some other tools)?
Thanks a lot!
PS. At this moment I can certainly send "302 Moved Temporarily" on
REGISTER with SER, but I have no idea on how to insert correct Contact
header, so my SIP client just stops registering.
Hello,
We are facing a problem after doing the changes that you said to do in the
configuration file. Now we are able to get the server listening on two ports
such as 5060 & 5062 but now the problem is that the user agent which is
registered from 5060 is able to communicate with the other user agent
registered to 5060 and the same is true for the other port 5062 (i.e. the
user agent which is registered from 5062 is able to communicate with the
other user agent registered to 5062.). But there is no communication
happening in case of one user is registered to 5060 and the other user is
registered through 5062. Please help us in this regard.
Thanks and Best Regards
Saurabh Srivastava
On Wed, Oct 13, 2010 at 5:59 PM, Saurabh Srivastava <saurabh(a)atc-labs.com>wrote:
> Hello Andrei,
>
> Wwhere do we need to add this line
>
> "listen=eth0:5060 eth0:5062"
>
> in the ser.cfg file or somewhere else. Please guide me through the steps or
> if it is possible can you please attach an example ser.cfg for listening on
> 2 ports.
>
> Thanks and Best Regards
> Saurabh Srivastava
>
> On Fri, Oct 8, 2010 at 1:28 PM, Andrei Pelinescu-Onciul <
> andrei(a)iptel.org> wrote:
>
>> On Oct 06, 2010 at 11:43, Saurabh Srivastava <saurabh(a)atc-labs.com>
>> wrote:
>> > Hello SER Team,
>> >
>> > We are using SER as a SIP Server for our purpose. We wan to configure
>> SER to
>> > listen on two ports at the same time(i.e. 5060 & some other custom port
>> such
>> > as 5090 or something like that). Can you please assist us how can we
>> achieve
>> > this, means what changes we need to make such that SER listens on two
>> ports
>> > at the same time.
>>
>> Add
>> listen=eth0:5060 eth0:5062
>>
>> The generic format is: listen=[proto:]ip_hostname_or_interface[:port].
>>
>> You can also do it form the command line. e.g.:
>> ser -f yourconfig -l 127.0.0.1:5060 -l 127.0.0.1:5062
>>
>> Andre
>
>
*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 experience
with and Kamailio is ready now for using TLS over TCP with no problem...
My question is, how much TLS/TCP calls Kamailio can handle and does anyone
had an experience with it?
Regards*
Hello,
I am not good at kamailio. Could anyone tell me how can I forward register
messages coming to kamailio to asterisk? I need users to be registered on
kamailio and on asterisk at one time.
I do in kamailio.cfg:
if (is_method("REGISTER"))
{
save("location");
route(FWD_REG);
exit;
}
route[FWD_REG] {
rewritehost("Asterisk_IP:5060");
if(!t_relay())
sl_reply_error();
exit;
}
But it (kamailio ver 3.1) does not forward them.