greetings, I have a customer that is sending the REGISTER with 2 ports in the Contact header:
Contact: <sip:44435661000448181@201.xxx.xxx.xxx:19778:5060>
After some troubleshooting we found that the wireless router is the one inserting the 2nd port. It is not the SIP UA itself. Is it possible to "sanitize" the REGISTER at the beginning of the configuration logic with the TEXTOPS module so that we can leave one port and then process the REGISTERwith the typical: if (!www_authorize("$fd", …
[View More]"subscriber")){...} Or will this only applies to messages that are proxied/forwarded (example: received malformed INVITE, before calling route[1] to send it to the destination sanitize it with the TEXTOPS functions)I guess my question is: after removing the extra port at the begining, when I call the "if (!www_authorize("$fd", "subscriber"))",will it process the modified version or the original one? txs a lot fborot
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Take a look at Homer.
Reda Aouad <reda.aouad(a)gmail.com> wrote:
>Hello,
>
>Anyone knows a tool/software (NOT CDRTool) to graphically visualize the SIP
>traces recorded in database by Kamailio's siptrace module?
>
>Thanks
>Reda
>
>_______________________________________________
>SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>sr-users(a)lists.sip-router.org
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Hello,
Anyone knows a tool/software (NOT CDRTool) to graphically visualize the SIP
traces recorded in database by Kamailio's siptrace module?
Thanks
Reda
Hi,
I'm just wondering if there has been any type of work to implement any
part of RFC 5626, particularly the registrar server code?
Lumicall now generates a UUID at install time, and sends that as
sip.instance in the Contact header. It always sends reg-id=1 (only
trying one proxy at the moment).
Although this is not a full implementation of the spec, it seems to be
sufficient for eliminating extra Contact records in the repro
registration server (from reSIProcate). However, I've noticed …
[View More]Kamailio
doesn't seem to understand this parameter yet.
Regards,
Daniel
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hello,
I am running kamailio 3.2.2 on Solaris 10 64 bit and I am getting a core
dump on this statement:
$ru = "sip:" + $rU + "@" + $sel(cfg_get.registrar1.server_ip)
+ $sel(cfg_get.registrar1.server_port);
Here is how the variable is defined:
registrar1.server_ip = "xxx.xxx.xxx.xxx" desc "Registrar server IP address"
registrar1.server_port = "5080" desc "Registrar server Port"
Here is the output from the logs with debug:
10(25373) ERROR: *** …
[View More]cfgtrace:
c=[/opt/kamailio-3.2/etc/kamailio/kamailio.cfg] l=1072 a=65 n=assign
10(25373) DEBUG: <core> [select.c:424]: Calling SELECT 1004b2bc0
10(25373) DEBUG: <core> [select.c:424]: Calling SELECT 1004b4da0
10(25373) ERROR: *** cfgtrace:14(25377) : <core> [pass_fd.c:293]: ERROR:
receive_fd: EOF on 22
14(25377) DEBUG: <core> [tcp_main.c:3555]: DBG: handle_ser_child: dead
child 10, pid 25373 (shutting down?)
14(25377) DEBUG: <core> [io_wait.h:617]: DBG: io_watch_del (1003738b8,
22, 0, 0x0) fd_no=18 called
14(25377) DEBUG: <core> [tcp_main.c:3316]: DBG: handle_tcp_child: dead
tcp child 0 (pid 25373, no 10) (shutting down?)
14(25377) DEBUG: <core> [io_wait.h:617]: DBG: io_watch_del (1003738b8,
24, 1, 0x0) fd_no=17 called
0(25363) ALERT: <core> [main.c:751]: child process 25373 exited by a
signal 11
0(25363) ALERT: <core> [main.c:754]: core was generated
0(25363) INFO: <core> [main.c:766]: INFO: terminating due to SIGCHLD
14(25377) INFO: <core> [main.c:817]: INFO: signal 15 received
Here is the full route code:
route[TO_REGISTRAR] {
# check if Registrar server's IP is defined
if (strempty($sel(cfg_get.registrar1.server_ip))) {
xlog("SCRIPT: REGISTRAR - registrar1.server_ip not
defined\n");
return;
}
$ru = "sip:" + $rU + "@" + $sel(cfg_get.registrar1.server_ip) + ":"
+ $sel(cfg_get.registrar1.server_port);
route(RELAY);
exit;
} # End of TO_Registrar Route
I am trying to setup the proxy server to route all registers requests to
another kamailio server
Thanks
Nathaniel
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Hello!
I get step by step to my multihomed setup and have now problems that
kamailio didnt rewrite
the 200 OK to my public ip address.
I have attached an ngrep trace and my kamailio 3.2.2 config.
Would be great if someone can give me an hint!
222.222.222.222 public ip-address from kamailio
172.20.100.74 private ip-address from kamailio
217.777.777.777 public ip-address from the UAC
172.20.100.103 private ip-address from the UAC
172.20.100.61 sip ip-address from the IVR …
[View More]behind kamailio
172.20.100.71 rtp ip-address from the IVR behind kamailio
--
Mit freundlichen Grüßen
*Karsten Horsmann*
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Dear Daniel ,
I thank you for your reply , I have a server having the Astersisk ip
address (192.168.10.15) , and rtp + kamailio is installed on an
another pc have the following ip (192.168.10.17) which is linked to
the Astersik , and on the same pc , another network card exists having
the ip address (192.168.20.3 ) which is linked to a client pc having
the ip address ( 192.168.20.4) .
I tracked the call and i can see SIP ACK nd BYE between 20.3 and 20.4
but there is no audio conversation …
[View More] this is my configuration file for
kamailio attached above .
P.S : testing without NATING as described in the above setup .
I thank you alot again for all your help .
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