getting this error:
kamailio: ERROR: <core> [modparam.c:151]: set_mod_param_regex: parameter
<trace_to_database> not found in module <siptrace>
parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line
63, column 46: Can't set module parameter
I have set this to modparam("siptrace", "trace_to_database", "0") in my .cfg
/I didn't compile siptrace when I compiled kamailio. I don't know how to
add a module after compiling and installing kamailio. I haven't been
able to find any documentaion on how to add a module after installing
only compiling before installing.
Thanks all for your help
Christopher
/
I am looking for hosted Kamailio Solutions for my VOIP calling app in
android and iphone. I have the app ready and have even setup kamailio
server in our rackspace cloud but there is many problems like one way
calling, noisy calls, no call connection etc. (application VOIP library ,
we are using IMSdroid and Idoubs , based on doubango SIP)
I am looking for a hosted solutions affordable for me - with a support for
300 concurrent calls at a time.
Best Regards,
Satyajeet Thokchom
Business Manager, NGBM Solutions Ltd
Phone:
India:+91-11-25541320
Mobile: 91 7428565264
Website:
www.ngbm.net
Web Support:
www.ngbmsolutions.com
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I'm setting up Homer with kamailio for sip capturing. I've got it all
set up and am getting this error. I have one interface for packet
capture on a mirroired port and one interface for managment. I'm
thinking that I have to only have kamailio listen on the packet capture
interface and not the managment interface. Would that resolve this issue?
loading modules under
/usr/local/lib/kamailio/modules_k/:/usr/local/lib/kamailio/modules/
0(1832) WARNING: <core> [socket_info.c:1406]: WARNING: fix_hostname:
could not rev. resolve 192.168.111.88
Listening on
udp: 127.0.0.1:5060
udp: 192.168.111.88:5060
Aliases:
udp: localhost:5060
config file ok, exiting...
sense it's a local address it's not possible to do a rev. resolve of
that address. Is there a way to turn off the rev. resolve.
I've looked at this in the trouble shooting faq.
How do I avoid Reverse-DNS-Lookups?
A: Simply not use src_ip = "11.22.33.44" and use instead src_ip =
11.22.33.44! We had in our installation some serious trouble with
requests coming from systems, where the hostname could not be reverse
resolved (the DNS-Server responsible for this IP-Range was poorly
configured and would not answer at all).
Hi,
I am trying to get a company who handles our billing and account
provisioning connected to our Kamailio instance. I understand that XMLRPC
is likely the best method for doing this, but I am unable to find direct
commands equivalent to "kamctl add" or "kamctl rm". Is there a resource
that lists the MI or XMLRPC commands, or do I need to create even simple
commands?
Thanks,
Ed
Hi ,
i don't know how to realize this:
A --- INVITE---->proxy ----INVITE----> B
<--- 200 OK
---BYE---->
---INVITE ----->C
<--- 200 OK proxy <--- 200 OK
-------ACK ---> -------ACK --->
When B answer the call, the proxy should immediately hang up the call and
make a new invite. For user A is always the same call.
I tried this :
onreply_route {
################################ connect
if (is_method("INVITE") && status == "200") {
dlg_bye("callee");
drop();
}
}
But it doesn't work. When arrive 200 , the proxy drop the message to the
user a , wait for a while, and then send a cancel to b (it should send a
BYE, instead send a CANCEL, probably because it not have a ACK).
How can I try this?
Thanks
the patch below adds 't' flag to rtpproxy_destroy flags. if 't' flag is
present when rtpproxy_destroy is called, to tag is not included in D
command sent to rtpproxy thus causing destroy of the full call.
't' flag can be used to avoid piling up of unused rtpproxy calls by
making it possible to call rtpproxy_destroy when 200 ok is received from
a branch where rtpproxy is not used.
is it ok for me to commit this patch or do long term rtpproxy users have
a better idea?
it would still make sense to include a timer in rtpproxy that cleans up
un-used calls at an interval that corresponds to sip proxy's invite
timeout.
-- juha
*** /usr/src/orig/sip-router/modules/rtpproxy/rtpproxy.c 2012-10-19 14:31:27.658485534 +0300
--- modules/rtpproxy/rtpproxy.c 2012-10-20 09:52:10.308389743 +0300
***************
*** 1678,1683 ****
--- 1678,1684 ----
str callid, from_tag, to_tag, viabranch;
char *cp;
int via = 0;
+ int to = 1;
int ret;
struct rtpp_node *node;
struct iovec v[1 + 4 + 3 + 2] = {{NULL, 0}, {"D", 1}, {" ", 1}, {NULL, 0}, {NULL, 0}, {NULL, 0}, {" ", 1}, {NULL, 0}, {" ", 1}, {NULL, 0}};
***************
*** 1700,1705 ****
--- 1701,1710 ----
via = 1;
break;
+ case 't':
+ case 'T':
+ to = 0;
+ break;
case 'a':
case 'A':
case 'i':
***************
*** 1736,1742 ****
return -1;
}
to_tag.s = 0;
! if (get_to_tag(msg, &to_tag) == -1) {
LM_ERR("can't get To tag\n");
return -1;
}
--- 1741,1748 ----
return -1;
}
to_tag.s = 0;
! to_tag.len = 0;
! if ((to == 1) && get_to_tag(msg, &to_tag) == -1) {
LM_ERR("can't get To tag\n");
return -1;
}
i tried what happens when i call start_recording() after calling
rtpproxy_manage() and got this kind of error to syslog:
Oct 20 19:57:43 siika mediaproxy-ng[8423]: Failed to properly parse UDP command line '8554_13 R ngsgznnwgzpuovt(a)siika.tutpro.com mgpwe' from 127.0.0.1:51497, using fallback RE
why the error? it is so that mediaproxy-ng does not support recording?
-- juha
You can strip 100rel as long as it's Supported. However, not so good things happen if you strip it from Required. :-)
-- Alex
--
Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard.
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
Tel: +1-678-954-0670
Web: http://www.evaristesys.com/Andrew Mortensen <admorten(a)isc.upenn.edu> wrote:Can I safely strip 100rel support from INVITEs sent by callers registered to my proxy?
We're using proxy-generated 302 redirects for call forwarding in our deployment. In general this works better for us than appending branches, as we rapidly ran into branch limits when dealing with large multiple appearance lines.
However, we've run into a case where the caller, a Polycom 550, seems to get confused when a parallel forked call is redirected, 100rel is in use and the redirection target is also a multiple appearance line.
The caller is correctly responding with PRACKs to all 180s, before and after the redirection. However, hanging up the call on the caller after the redirection, but before any remote party has answered, does *not* cause the caller to send a CANCEL to tear down the call. In fact, the caller doesn't send anything at all. The remote parties continue to ring until the ring timer expires, and the proxy sends CANCELs to all callee endpoints.
That's clearly a Polycom bug, and we'll need to work with them to get it resolved.
In the meantime, I'm looking for workarounds. One quick, crude option is to strip 100rel from the Supported header in the caller's INVITE. I've tested this, and it does solve the problem, but I'm concerned about side effects. I'm aware of the value 100rel has for calls to the PSTN, but in this case the calls never leave the local network. Are there situations where 100rel is critical for a parallel forked call?
andrew
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