Good Morning list,
i have kamailio installed in my LAN (192.168.170.101 - machine 1), and i
have eyebeam (192.168.46.101 - machine 2) registred in my kamailio, when
eyebeam make a call i would like to route this call to a gateway
(XXX.XXX.XXX.XXX - machine 3), how can i do that? i would like to read a
practical manual.
thanks
I also have another interesting problem with the aforementioned
configuration (http://pastebin.com/f28051a5).
When I write the dialog profile size into an AVP, it works fine.
When I write it into a script var, i.e. replace $avp(S:dlg_sz) with
$var(dlg_sz), it crashes:
Relaying INVITE from 210.23.22.23 to sip:2122222322@210.23.22.23:5060
[D] Added new dialog for 7709600101
[ONREPLY-ROUTE 1] Provisional reply 100 received.
[ONREPLY-ROUTE 1] 200 OK received for 7709600101
[ONREPLY-ROUTE 1] INVITE/200 is part of dialog for 7709600101
Segmentation fault (core dumped)
GDB reveals:
Program received signal SIGSEGV, Segmentation fault.
0xb7e7f2f8 in strncpy () from /lib/i686/nosegneg/libc.so.6
(gdb) where
#0 0xb7e7f2f8 in strncpy () from /lib/i686/nosegneg/libc.so.6
#1 0x080a4a07 in set_var_value (var=0x8188d98, value=0xbfae5f28,
flags=<value o
at script_var.c:122
#2 0xb7d6c4fe in w_get_profile_size (msg=0x81902b8, profile=0xb5ede180
"\034??\
value=0x818b6b8 "@?\030\b", result=0x818b6f8 "N") at dialog.c:668
#3 0x08054f15 in do_action (a=0x818b850, msg=0x81902b8) at action.c:850
#4 0x08053ed2 in run_action_list (a=0x818b5c8, msg=0x81902b8) at
action.c:138
#5 0x08056365 in do_action (a=0x818bab8, msg=0x81902b8) at action.c:722
#6 0x08053ed2 in run_action_list (a=0x818b178, msg=0x81902b8) at
action.c:138
#7 0x080572c2 in run_top_route (a=0x818b178, msg=0x81902b8) at action.c:118
#8 0xb7de3a64 in reply_received (p_msg=0x81902b8) at t_reply.c:1361
#9 0x08064793 in forward_reply (msg=0x81902b8) at forward.c:507
#10 0x08090d5b in receive_msg (
buf=0x81600e0 "SIP/2.0 200 OK\r\nVia: SIP/2.0/UDP
215.22.22.22;branch=z9hG4
08.52.173.18\r\nVia: SIP/2.0/UDP
198.225.86.10:5060;branch=z9hG4bKragjo0207gm0dc
p:208.52."..., len=814, rcv_info=0xbfae65d4) at receive.c:203
#11 0x080cccfb in udp_rcv_loop () at udp_server.c:449
#12 0x0806b78d in main (argc=1, argv=0xbfae6764) at main.c:693
It would seem to me that there is some sort of buffer overflow issue
that results in the garbage seen above.
Not sure that it makes a difference, but the glibc being linked against
is a Xen-safe one that disables TLS functionality. This is all running
inside a Xen DomU.
--
Alex Balashov
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599
Hello, my name is Scott Seltzer, from ConnectMe (www.connectmevoice.com
<http://www.connectmevoice.com/> )
Connectme is currently in the process of setting up SER for use as a
Proxy and Registrar. We will ultimately need to be able to use an API
to setup new SIP URIs for each new customer. We are not familiar at all
with SER, and would like to find out what support options there may be
out there. We would also be interested in using some consulting
services to help get us started, and support us along the way. Please
let me know if there is anybody that can help us out with this.
Thanks!
Scott Seltzer
Hi forum, I am trying to configure cdrtool with openser and radius, but in the administration console it shows me this message: "Error instantiating db class: ".
I have created the radius database and I see that the openser with the I modulate ACC, he writes data in the chart radacct
any suggestion?,
something that me this forgetting the file global.inc?
best regards
rickygm
Hi, I've stored in "usr_preferences" some entries with:
- attribute: 'bla'
- value: 'alice(a)domain.org' <--- With no "sip:" in front.
I use 'avp_db_load' to get that value and compare using 'avp_check' with the
To uri without protocol (without "sip:").
I get this behaviour by doing:
--------------------------
if avp_db_load("$fu/uri", "$avp(s:bla)") {
$var(To) = $tU + '@' + $td;
if avp_check("$avp(s:bla)","eq/$var(To)/gi") {
[...]
--------------------------
But I would like to avoid using a variable $var(To), something as:
--------------------------
if avp_db_load("$fu/uri", "$avp(s:bla)") {
if avp_check("$avp(s:bla)","eq/$tU + '@' + $td/gi") {
# or:
if avp_check("$avp(s:bla)","eq/$tU@$td/gi") {
[...]
--------------------------
Unfortunatelly it doesn't work, it's not valid. Do I really need to use a
variable (or AVP) for "avp_check"?
Thanks.
--
Iñaki Baz Castillo
Hello all,
I am working on configuring openser as a outbound proxy. User agents register to service provider through openser outbound proxy. I am experiencing issues with retransmissions.
Case 1.
UA sends an INVITE to Proxy and proxy is down. Openser sends 100 Trying to UA and retransmits INVITE messages to proxy. After 8 retry attempts sends 408 Request time out to handset. All as expected.
Case 2:
Openser received INVITE for UA from proxy. UA is down. Openser sent 100 Trying to proxy and routes reqest to UA. No response from UA and ** Openser did not send any retransmissions of INVITE to UA**. After a while CANCLE arrived at openser and cleared transaction.
why openser does retransmissions in Case 1 and why not in case 2?
Can openser proxy be operated in stateful outbound proxy to take care of retransmissions?
I am using default openser config file. Is there any thing I am doing wrong?
Any help is greatly appreciated.
Thanks
Sid
Does Kamailio match 'uri == myself' if the domain is the same as the
canonical hostnames/aliases but the port is different than the one
Kamailio knows itself to be listening on?
-------- Original Message --------
Subject: Re: [Sip-implementors] REGISTER R-URI with port parameter
Date: Tue, 28 Oct 2008 16:46:04 -0500
From: Dale.Worley(a)comcast.net
To: sip-implementors(a)lists.cs.columbia.edu
References:
<00D42150952F70458C66072322F7FE2502EDA940(a)saturn2.aculab.com>
<200810241652.m9OGqx6s027493(a)dragon.ariadne.com>
<200810241902.43581.ibc(a)aliax.net>
From: =?iso-8859-1?q?I=F1aki_Baz_Castillo?= <ibc(a)aliax.net>
El Viernes, 24 de Octubre de 2008, Dale.Worley(a)comcast.net escribió:
> sip:xxx@host:5060
> sip:xxx@host:15060
> sip:xxx@host:25060
> sip:xxx@host:35060
>
> In that case, the request-URI of the REGISTER needs to contain the
> proper port number.
Well, not totally needed. A UA could construct a REGISTER like this:
REGISTER sip:xxx@host SIP/2.0
and sent it to host:15060.
You can't depend on that working -- It's never been settled what must
happen if a SIP message arrives at a UAS which isn't the UAS that the
RFC 3263 rules would send it to. It's possible that the UAS will
forward it based on the request-URI.
It's safer to include the port, if the destination port is not 5060.
For example Twinkle does it if you set the registrar in a port
different than 5060.
I wouldn't depend on that working.
Dale
_______________________________________________
Sip-implementors mailing list
Sip-implementors(a)lists.cs.columbia.edu
https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors
--
Alex Balashov
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599
Guys,
When attempting to pass REGISTER requests up stream I get the following
error messages on the console (below). This also brings up the question of
"what is the best way to simply pass REGISTER requests on to an upstream
registrar?"
Thanks for the help.
-Daniel
REGISTER: NATed client, enabling NAT
Oct 29 18:03:58 [48175] ERROR:tm:t_forward_nonack: no branch for forwarding
Oct 29 18:03:58 [48175] ERROR:tm:w_t_relay: t_forward_nonack failed
Oct 29 18:03:58 [48175] ERROR:tm:t_forward_nonack: no branch for forwarding
Oct 29 18:03:58 [48175] ERROR:tm:w_t_relay: t_forward_nonack failed
Oct 29 18:03:59 [48175] CRITICAL:tm:t_should_relay_response: pick_branch
failed (lowest==-1) for code 401
Here's the relevant parts of my ser config;
in route[1]
route[1]
{
...
if (method == "REGISTER")
{
route(2);
}
route(1);
}
route[2]
{
# Check to see if the UAC is trying to UNREGISTER. If not test
# for NAT. If NAT is present, we mark it as such before we save()
# the location, that way flag 6 is always set for NATed UAs.
if (!search("^Contact:[ ]*\*") && nat_uac_test("19")) {
xlog("L_NOTICE", "REGISTER: NATed client, enabling NAT\n");
setflag(6);
fix_nated_register();
force_rport();
};
# We must handle replies for registrations, for caching and location
# tracking purposes.
t_on_reply("1");
t_on_failure("1");
# Check for digest
#if (radius_www_authorize(""))
#{
# xlog("L_NOTICE", "REGISTER: No Digest, sending
challenge\n");
# www_challenge("", "0");
# exit;
#}
# Digest was good if we get here
#save("location");
# Relay register to porta
t_relay("216.151.143.69");
}
onreply_route[2]
{
if (status =~ "2[0-9][0-9]")
{
# save("location");
};
}
> Hello everyone,
> I'm currently working on the implementation of RFC4474 in SER.
> I didn't manage to set up the auth_identity module in SER.
> Is there someone able to tell me what do I have to write in ser.cfg to make this possible?
> An example of an entire ser.cfg will be great.
> I hope you will help me.
> Thanks & regards,
>
> Jérôme HERVE
> tél. 02 96 05 27 41
> mob. 06 76 15 18 49
>
>
Hello everyone,
I'm currently working on the implementation of RFC4474 in SER.
I didn't manage to set up the auth_identity module in SER.
Is there someone able to tell me what do I have to write in ser.cfg to make this possible?
An example of an entire ser.cfg will be great.
I hope you will help me.
Thanks & regards,
Jérôme HERVE
tél. 02 96 05 27 41
mob. 06 76 15 18 49