Hi,
We are migrating from our openser 1.4 to kamailio 4.0.3 and have run into a
couple of issues. One of them is described below and we need to know if it
is a known bug or not.
Kamailio receives an INVITE from a gateway and finally responds with 200
OK. Then an ACK is generated by the gateway towards the Kamailio that some
how does not get relayed to the next hop.
The only special aspect here is that the ACK does not contain a Contact
header. Is that something that could confuse kamailio and make it unable to
relay?
According to SIP RFC, a contact header in ACK is not a must have. Our case
is the one where ACK is sent by the UAC after receiving a 200 OK final
response. Everything else about the ACK looks OK e.g. ACK has a branch
different from the the branch value in the UAC INVITE.
Is it a a known bug in 4.0.3?
In the configuration, we handle ACK as below:
if (is_method("ACK")) {
if (t_check_trans()) {
t_relay();
exit;
} else {
exit;
}
Br
Adnan
Hello,
Using 4.0.4:
I'm getting the following error (removed the values part of the query):
3(2931) ERROR: db_unixodbc [dbase.c:133]: db_unixodbc_submit_query(): rv=-1. Query= insert into Dialogs (HashEntry,HashID,CallID,FromURI,FromTag,ToURI,ToTag,CallerSock,CalleeSock,StartTime,State,Timeout,CallerCSeq,CalleeCSeq,CallerRouteSet,CalleeRouteSet,CallerContact,CalleeContact,SFlags,ToRouteName,req_uri,xdata,iflags )
3(2931) ERROR: db_unixodbc [con.c:220]: db_unixodbc_extract_error(): unixodbc:SQLExecDirect=42S22:1:207:[FreeTDS][SQL Server]Invalid column name 'req_uri'.
The dialog.c file infact doesn't contain the req_uri_column, x_data_column and the iflags_column in static param_export_t mod_params[],static int mod_init(void).
The dlg_db_handler.c files does contain them in the beginning of the file and in static int select_entire_dialog_table(db1_res_t ** res, int fetch_num_rows).
I haven't checked the other remaining files, but I think the only problem is the renaming of these columns in the kamailio script.
Just wondering if anyone else has noticed this?
Problem with signalling - RTP gets lost!
Rtpproxy not working properly?
I am absolutely stuck ... cause this happens in a live environement.
I have the following situation
A calls B over carrier 1 - number is not valid and I get back error 404 from carrier and now freeswitch should play a message saying: "number not valid".
But from carrier 1 I get back an RTP stream that is useless [1] - and if the correct streams opens from freeswitch - this does not get back to A [2].
I tested with rtpproxy on Kamailio - and all the rtp streams arrive at the Kamailio - but they cannot be "connected" correctly.
I guess the problem is the 183 I get back from carrier 1 - after whitch rtp is opened.
Or there is a wrong sdp singallisation if the "correct" stream arrives [3].
Sorry - I cannot get a solution - but I could provide various tcpdumps and pcaps.
A Kamailio Carrier 1 Freeswitch
INVITE
--------------------->
100 Your call is important
<---------------------
INVITE
------------------------>
100 Trying
<------------------------
183 Session Progress SDP
<------------------------
183 Session Progress SDP
<------------------------
RTP
<=================
RTP
<=================
RTP
=================>
RTP
=================> [1]
404 not found
<---------------------------
ACK
---------------------------->
INVITE
--------------------------------------------------------->
100 Trying
<--------------------------------------------------------
200 OK SDP
<-------------------------------------------------------- [3]
200 OK SDP
<---------------------------
RTP (Announcment - number not valid") [2]
<===================================
ACK
---------------------------->
ACK
---------------------------------------------------------->
INFO
---------------------------->
INFO
---------------------------------------------------------->
200 OK
<--------------------------------------------------------
200 OK
<---------------------------
BYE
---------------------------->
BYE
---------------------------------------------------------->
200 OK
<--------------------------------------------------------
200 OK
<---------------------------
Dear Sir,
well i would like to know,is there any web interface for kamailio to
configure complete configuration part,database,modules and upgrades similar
like Astrix server ,Kamailio Sip server is Really Rock Solid server,when i
first visited your site,i saw the name called Rock Solid ,i thought it just
the call like that ,but after testing it ,it satisfied my needs,that is
Video calling 2 way ,in excellent manner,well i would like to is Kamailio
Sip server is Peer to Peer or not,if it is Peer to Peer,i have tested it ,i
removed the connectivity from Server between Clients when call
established,it worked for 20 mins and after it discards the call ,so i was
bit confusing that kamailio sip server is Peer to Peer or not,pls confirm
me,if it is peer to Peer,pls send me the step by step instruction how to
configure the Peering module ,But i request you if there is any Web
interface Like Astrix it would be great to control all the elements in
Kamailio
Best Regards
T.NAVEEN CHAND
I am running Kamailio in CentOS. I ran tcpdump and noticed that we are getting attacked from IP 188.138.32.72. I have already blocked it on IPtables, but he keeps on attacking the server. If I look at "/var/log/secure" there are no SIP messages. My question is where is the log file for Kamailio and how can I prevent this type of attacks in the future.
Thanks,
Hi all
Based on my problem reported with subject "error handling<http://sip-router.1086192.n5.nabble.com/error-handling-tp123240.html>" I have some other questions.
I think it is a conceptual question - and I do not see any solution.
I would like to handle Kamailio with carrierroute / carrierfailureroute module different errors.
Like 404, 403, busy - or whatever sip error occurs.
Some of them need to be sent to a freeswitch playing an announcement (like "this number is blocked", "no more credit", ...).
So if an error occur (lets say 403) then the call is routed by carrierfailurroute to fresswitch playing message for 403.
If I am listening the whole message - I get back error 403 at the end and the call is logged in missed calls as error 403 sent from the freeswitch - everything ok.
If I cancel the listening by hanging up - 487 is stored in missed_calls - cause I terminated the call before getting error 403 back from freeswitch. ==> so I loose this important information
How can I get back error 403 - play an announcement and make sure, it is logged as 403 in database?
Thanks for helping ...
Oli
I am having the exact same problems as issue as this user. There is
nothing in the siremis/log/ folder when this happens. I'm running
Siremis 4.0.0 with Kamailio 4.0.4. Has this issue been fixed?
Hello,
On 5/18/13 3:22 AM, Alexander Albert wrote:
> Hi there, i hope you can help me, i am having trouble with the
> register page of siremis, strange thing is, i can open the register
> page, but only after i have logged my self in, if not, it redirects me
> directly to the login page.. how can i setup the register page to be
> accessible without the need of logging in ?
> Also i am getting the info after trying to register, that "Public
> registration is not enabled!"
> but i have set siremis/modules/ser/config/common.Main.php to
> $cfg_siremis_public_registrations = true;
> What is the problem, please help.
can you look in siremis/log/ folder, there are some files printing debug
messages, maybe you get hints about what is wrong.
Cheers,
Daniel
