Hello
This is the output of ldd /usr/local/lib/kamailio/modules_k/rtpproxy.so:
shell-init: error retrieving current directory: getcwd: cannot access
parent directories: No such file or directory
linux-gate.so.1 => (0x00276000)
libkmi.so.1 => /usr/local/lib/kamailio/libkmi.so.1 (0x00b64000)
libkcore.so.1 => /usr/local/lib/kamailio/libkcore.so.1 (0x00768000)
libc.so.6 => /lib/tls/i686/cmov/libc.so.6 (0x00277000)
/lib/ld-linux.so.2 (0x00114000)
Regards
Phillip
Message: 3
Date: Tue, 14 Aug 2012 08:49:13 -0400
From: Alex Balashov <abalashov(a)evaristesys.com>
Subject: Re: [SR-Users] rtpproxy module issue
To: sr-users(a)lists.sip-router.org
Message-ID: <502A4949.2000707(a)evaristesys.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Please post output of:
ldd /usr/local/lib/kamailio/modules_k/rtpproxy.so
On 08/14/2012 08:47 AM, phillman25 wrote:
> Dear List
>
> I have just installed kamailio V 3.3.1
> However, when trying to start kamailio its giving me the below error:
>
>
> ERROR: <core> [sr_module.c:572]: ERROR: load_module: could not open
> module </usr/local/lib/kamailio/modules_k/rtpproxy.so>:
> /usr/local/lib/kamailio/modules_k/rtpproxy.so: undefined symbol:
qm_realloc
> : <core> [cfg.y:3591]: parse error in config file
> /usr/local/etc/kamailio/kamailio.cfg, line 245, column 12-24: failed to
> load module
>
>
> Does anyone know why rtpproxy module cannot be loaded?
>
>
> Thanks
> Phillip
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users(a)lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
--
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/, http://www.alexbalashov.com/
Dear List
I have just installed kamailio V 3.3.1
However, when trying to start kamailio its giving me the below error:
ERROR: <core> [sr_module.c:572]: ERROR: load_module: could not open module
</usr/local/lib/kamailio/modules_k/rtpproxy.so>:
/usr/local/lib/kamailio/modules_k/rtpproxy.so: undefined symbol: qm_realloc
: <core> [cfg.y:3591]: parse error in config file
/usr/local/etc/kamailio/kamailio.cfg, line 245, column 12-24: failed to
load module
Does anyone know why rtpproxy module cannot be loaded?
Thanks
Phillip
Dear List
I am trying to remove specific lines from the following original SDP body:
Content-Type: application/sdp
Content-Length: 406
v=0
o=root 3048 3048 IN IP4 xxx.xxx.xxx.xxx
s=session
c=IN IP4 xxx.xxx.xxx.xxx
b=CT:384
t=0 0
m=audio 11904 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=video 11602 RTP/AVP 34 99
a=rtpmap:34 H263/90000
a=rtpmap:99 H264/90000
a=sendrecv
I need to remove the following lines from the above SDP body as my
International carrier does not support Video capabilities:
m=video 15042 RTP/AVP 34 99
a=rtpmap:34 H263/90000
a=rtpmap:99 H264/90000
I used the following code in Kamailio config:
if(has_body("application/sdp") && search_body("m=video")){
subst_body('#m=video ([0-9]+) RTP/AVP (.*)$# #');
subst_body('#a=rtpmap:34 (.*)$# #');
subst_body('#a=rtpmap:99 (.*)$# #');
subst_body('#a=sendrecv(.*)$# #');
}
Content-Type: application/sdp
Content-Length: 325
P-hint: outbound
v=0
o=root 3048 3048 IN IP4 xxx.xxx.xxx.xxx
s=session
c=IN IP4 xxx.xxx.xxx.xxx
b=CT:384
t=0 0
m=audio 11904 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
As you can see the m=video body has been removed, however, calls are still
failing. Is there something i have missed?
I am using Kamailio v3.2.2
Thanking you in advance!
Phillip
As per the instructions here... http://www.kamailio.org/w/2012/07/websockets
...I downloaded the latest source from Git...
http://www.kamailio.org/w/download/
...but I don't see any mention of Websocket anywhere in the source. I
suspect I didn't actually download the development tree, but the 3.3
branch.
Last time I messed with any VoIP code was in 2008 (OpenSER and
Asterisk) and now I'm starting all over again. I apologize for the
dumb newbie question, but how do I test the Websocket code?
--
Juan Carlos Castro y Castro
Instant Solutions - Telefonia Gerando Resultado
http://www.instant.com.br
Principais capitais: 4063-6100
Demais regiões: (11)4063-6100
Hello.
After upgrade kamailio and db, lcr module bad routing if entered *any
value*to from_uri column in lcr_rule table:
mysql> select * from lcr_rule where id=5;
+----+--------+--------+-------------------------------+-------------+---------+---------+
| id | lcr_id | prefix | from_uri | request_uri |
stopper | enabled |
+----+--------+--------+-------------------------------+-------------+---------+---------+
| 5 | 1 | yyyy | sip:x[xx]\d{2}@xxx.xxx.xx.xx | NULL |
0 | 1 |
+----+--------+--------+-------------------------------+-------------+---------+---------+
and load_gw(...) skip this rule, in log:
INFO: lcr [lcr_mod.c:1493]: skipping disabled <gw/rule> = <1/5>
If remove value from 'from_ui' - rule enabled and all work.
Why is turned off rule ?
Linux Debian squeeze, upgraded from "deb
http://deb.kamailio.org/kamailiosqueeze main"
--
"В связи с отсутствием интереса общественности, конец света отменяется" (с)
?
С Уважением, Механошин Алексей
Hi,
I've followed the tutorial on kab.asipto.com for presence using built-in
xcap server. http://kb.asipto.com/kamailio:presence:k31-made-simple
I'm using Kamailio version 3.3.1 and did minor changes in modparams and
rtpproxy function calls and the kamailio accepted the configurations file
posted on the page and started.
But the problem is that I don't get the presence status of the contacts
still. Please suggest what to look for and how to troubleshoot this.
I get this error on Jitsi - image attached.
ERROR: "http://ip.of.server/xcap-root/xcap-caps/global/index resource can
not be read"
Regards,
Sammy
Hello, guys.
What is the best tool for crafting sip messages for testing purpose? For
example, simple REGISTER or INVITE message. I'm new to kamailio (not sip)
so it would be great for me to use debbuger module for step-by-step digging
into cfg.
Hi,****
** **
I encountered a problem. While doing some avp's settings I notice that one
avp that I use is changing to <null> without me doing anything.****
** **
Here is the part of the cfg file:****
………….****
………….****
Route[SET]{****
…………..****
$avp(Sfeaturetype)=$avp(next_feature_type);****
…………****
Route(DIDSRV);****
}****
** **
route[DIDSRV] {****
xlog("L_CRIT","$C(rg) SCRIPT: in DIDSRV type = $avp(Sfeaturetype), index =
$avp(SfeatureIndx) $C(xx)\n");****
if($var(srvcount)>3){****
xlog("L_NOTICE","$C(rg)Too many services loops $C(xx)\n");****
update_stat("nts_subs_srv_loop", "+1");****
$avp(TRMCS)="10";****
t_reply(403,"too many loops");****
exit;****
}****
$var(srvcount)=($var(srvcount)+1);****
xlog("L_CRIT","$C(rg) SCRIPT: in DIDSRV before switch to $avp(SfeatureType)
or $avp(next_feature_type) $C(xx)\n");****
switch ($avp(SfeatureType)){****
case "0" :****
……….****
………****
** **
On the route "SET" I put the value from the $avp(next_feature_type) in
$avp(Sfeaturetype).****
After that I call the route "DIDSRV" and use switch($avp(Sfeaturetype)).
Before the switch I xlog the value of the avp twice.****
As you can see in the log below, on the first xlog print, the value is 0
(as I expect). On the second print it resets to <null>****
WHY?****
** **
Log:****
Aug 12 13:38:50 net-ivr-KamIN-Test kamailio[29309]: <core>
[mem/q_malloc.c:413]: qm_malloc(0x7fe6f9aff000, 72) returns address
0x7fe6fcc4d330 frag. 0x7fe6fcc4d300 (size=72) on 1 -th hit****
Aug 12 13:38:50 net-ivr-KamIN-Test kamailio[29309]: CRITICAL: <script>:
#033[0;31;42m SCRIPT: before DIDSRV type = 0, index = 8888888889
#033[0;39;49m****
Aug 12 13:38:50 net-ivr-KamIN-Test kamailio[29309]: CRITICAL: <script>:
#033[0;31;42m SCRIPT*: in DIDSRV type = 0*, index = 8888888889
#033<8888888889%C2%A0%20%23033>
[0;39;49m****
Aug 12 13:38:50 net-ivr-KamIN-Test kamailio[29309]: <core>
[mem/q_malloc.c:369]: qm_malloc(0x7fe7ff765010, 848) called from <core>:
rvalue.c: rval_new_empty(236)****
Aug 12 13:38:50 net-ivr-KamIN-Test kamailio[29309]: <core>
[mem/q_malloc.c:413]: qm_malloc(0x7fe7ff765010, 848) returns address
0x7fe7ff8f25b0 frag. 0x7fe7ff8f2580 (size=1232) on 1 -th hit****
Aug 12 13:38:50 net-ivr-KamIN-Test kamailio[29309]: <core>
[mem/q_malloc.c:437]: qm_free(0x7fe7ff765010, 0x7fe7ff8f25b0), called from
<core>: rvalue.c: rval_destroy(141)****
Aug 12 13:38:50 net-ivr-KamIN-Test kamailio[29309]: <core>
[mem/q_malloc.c:461]: qm_free: freeing frag. 0x7fe7ff8f2580 alloc'ed from
<core>: rvalue.c: rval_new_empty(236)****
Aug 12 13:38:50 net-ivr-KamIN-Test kamailio[29309]: CRITICAL: <script>:
#033[0;31;42m SCRIPT*: in DIDSRV before switch to <null>* or 0 #033[0;39;49m
****
Aug 12 13:38:50 net-ivr-KamIN-Test kamailio[29309]: <core>
[mem/q_malloc.c:369]: qm_malloc(0x7fe7ff765010, 928) called from <core>:
rvalue.c: rval_new_empty(236)****
Aug 12 13:38:50 net-ivr-KamIN-Test kamailio[29309]: <core>
[mem/q_malloc.c:413]: qm_malloc(0x7fe7ff765010, 928) returns address
0x7fe7ff8f25b0 frag. 0x7fe7ff8f2580 (size=1232) on 1 -th hit****
Aug 12 13:38:50 net-ivr-KamIN-Test kamailio[29309]: <core>
[mem/q_malloc.c:437]: qm_free(0x7fe7ff765010, 0x7fe7ff8f25b0), called from
<core>: rvalue.c: rval_destroy(141)****
Aug 12 13:38:50 net-ivr-KamIN-Test kamailio[29309]: <core>
[mem/q_malloc.c:461]: qm_free: freeing frag. 0x7fe7ff8f2580 alloc'ed from
<core>: rvalue.c: rval_new_empty(236)
Dears,
Am trying to apply Kamailio LCR feature lab test, after am finally able to
run Kamailio with no errors, am not able to reach my target clarified in
this following scenario:
Call 961312345 - route the call to 192.x.x.15 in first place and if failed
route the call to 192.x.x.10. I applied needed configuration in lcr tables
(attached image) and the routing script is:
route{
if (!mf_process_maxfwd_header("10")) {
sl_send_reply("483","Too Many Hops");
exit;
};
if (msg:len >= 2048 ) {
sl_send_reply("513", "Message too big");
exit;
};
if (!method=="REGISTER")
record_route();
if (loose_route()) {
append_hf("P-hint: rr-enforced\r\n");
route(1);
};
if (!uri==myself) {
append_hf("P-hint: outbound\r\n");
route(1);
};
if (is_method("INVITE"))
{
if (!load_gws(1)) {
sl_send_reply("503", "Unable to load gateways");
exit;
}
}
if (uri==myself) {
if (method=="REGISTER") {
save("location");
exit;
};
lookup("aliases");
if (!uri==myself) {
append_hf("P-hint: outbound alias\r\n");
route(1);
};
# native SIP destinations are handled using ourUSRLOC DB
if (!lookup("location")) {
sl_send_reply("404", "Not Found");
exit;
};
append_hf("P-hint: usrloc applied\r\n");
};
route(1);
}
route[1] {
if (!t_relay()) {
sl_reply_error();
};
exit;
}
BUT when attempting a call am getting the message 'Call failed: Not found .'
Any help to achieve the successful call using above scenario is highly
appreciated.
Thanks in advance.
F.Chahrour