Dear all,
When we use only one mediaproxy which is on the same machine with SER and disable the dispatcher, this system works perfectly. But now we want to use the dispatcher on the machine A with SER(ip:218.97.252.38), and install a mediaproxy on a separater machine B(ip:218.97.252.40), then mediaproxy just can not start. Call signaling is all OK, but there is no voice. SDP section of invite message is not changed by the mediaproxy.
The log file on machine A shows that:
Nov 8 16:36:10 b2bua proxydispatcher[5188]: command request 8029CEE9-92A3-4E09-91B5-0F1EE955B343(a)218.97.252.39 218.97.252.39:8000:audio 218.97.252.39 sensates.com remote 203.86.79.190 remote X-Lite=20release=201103a info=from:38001@sensates.com,to:3801@sensates.com,fromtag:614201966,totag:
Nov 8 16:36:11 b2bua proxydispatcher[5188]: error: couldn't forward command to 218.97.252.40:25060 (Connection refused)
The log file on machine B shows that:
Nov 8 11:46:10 localhost mediaproxy[4082]: mediaproxy started.
Nov 8 11:46:10 localhost mediaproxy[4082]: Listening for commands on local socket is disabled
Nov 8 11:46:10 localhost mediaproxy[4082]: Mediaproxy error: couldn't create command socket (Cannot assign requested address).
Nov 8 11:46:10 localhost mediaproxy[4082]: exiting.
Nov 8 11:46:10 localhost proxydispatcher[4085]: proxydispatcher started.
Nov 8 11:46:10 localhost proxydispatcher[4085]: disabled from configuration
Nov 8 11:46:10 localhost proxydispatcher[4085]: exiting.
We set mediaproxy.ini (on machine B) as follow:
[Dispatcher]
start = no
socket = /var/run/proxydispatcher.sock
group = ser
[MediaProxy]
start = yes
socket = None
group = ser
listen = 218.97.252.38:25060
allow = 218.97.252.38
proxyIP = 218.97.252.40
portRange = 35000:65000
TOS = 0xb8
idleTimeout = 60
holdTimeout = 3600
forceClose = 0
accounting = off
Thanks a lot.
Hello list,
I'm using openser 1.0 and I'm using auth_radius and avp_radius to retrieve
all my user's preferences via AVPs. Everything works very well except, that
I cannot retrieve AVPs with value of i:0xhex. I'm trying to implement the
ACL example in http://www.voice-system.ro/docs/avpops/ar01s08.html
In my radius config I have the following:
UserName Attribute op Value
Sipuser SIP-AVP = #102#0x08
In my openser.cfg I have the following:
...
if (uri=~"^sip:[0-9]{10}@") {
# Send to PSTN Gateway
xlog("L_N", " DOMESTIC CALL: from $fU to $rU\n");
if (avp_check("i:102", "and/i:0x01")) {
route(4); #NAT Test
route(5); #PSTN Route
} else {
xlog("L_N", " CALL NOT ALLOWED: from $fU to $rU, please
check $fU's ACL\n");
sl_send_reply("503", "Service Unavailable");
};
return;
};
...
This is the debug message that I get from openser:
0(25052) ERROR:auth_radius:extract_avp: invalid AVP numrical value '0x08'
# ------------------------------------------------------------------------
So I tried to cheat, by copying i:102 to a temporary AVP i:302 with the N
flag to convert the string to numerical value, like so:
UserName Attribute op Value
Sipuser SIP-AVP = #102:0x08
Openser.cfg:
if (uri=~"^sip:[0-9]{10}@") {
# Send to PSTN Gateway
if (avp_copy("i:102","i:302/n") && avp_check("i:302", "and/i:0x01"))
{
route(4); #NAT Test
route(5); #PSTN Route
} else {
xlog("L_N", " CALL NOT ALLOWED: from $fU to $rU, please
check $fU's ACL\n");
sl_send_reply("503", "Service Unavailable");
};
return;
};
This is the debug message that I get from openser:
0(25135) ERROR:avpops:copy_avp: cannot convert str to int
# ------------------------------------------------------------------------
And finally, I tried to force it by having the value as string, but try to
use the ANDing op, like so:
UserName Attribute op Value
Sipuser SIP-AVP = #102:0x08
Openser.cfg:
...
if (uri=~"^sip:[0-9]{10}@") {
# Send to PSTN Gateway
if (avp_check("i:102", "and/i:0x01")) {
route(4); #NAT Test
route(5); #PSTN Route
} else {
xlog("L_N", " CALL NOT ALLOWED: from $fU to $rU, please
check $fU's ACL\n");
sl_send_reply("503", "Service Unavailable");
};
return;
};
...
This is the debug message that I get from openser:
0(25140) ERROR:avpops:check_avp: value types don't match
# ------------------------------------------------------------------------
It seems to me that the problem lies in auth_radius by not allowing
extraction of AVPs with numerical values in hex. If we can get that work
work, I think the rest should work fine without any alterations.
Thanks for your help in advance!
Lenir
Hello list,
I'm using openser 1.0 and I'm using auth_radius and avp_radius to retrieve
all my user's preferences via AVPs. Everything works very well except, that
I cannot retrieve AVPs with value of i:0xhex. I'm trying to implement the
ACL example in http://www.voice-system.ro/docs/avpops/ar01s08.html
In my radius config I have the following:
UserName Attribute op Value
Sipuser SIP-AVP = #102#0x08
In my openser.cfg I have the following:
...
if (uri=~"^sip:[0-9]{10}@") {
# Send to PSTN Gateway
xlog("L_N", " DOMESTIC CALL: from $fU to $rU\n");
if (avp_check("i:102", "and/i:0x01")) {
route(4); #NAT Test
route(5); #PSTN Route
} else {
xlog("L_N", " CALL NOT ALLOWED: from $fU to $rU, please
check $fU's ACL\n");
sl_send_reply("503", "Service Unavailable");
};
return;
};
...
This is the debug message that I get from openser:
0(25052) ERROR:auth_radius:extract_avp: invalid AVP numrical value '0x08'
# ------------------------------------------------------------------------
So I tried to cheat, by copying i:102 to a temporary AVP i:302 with the N
flag to convert the string to numerical value, like so:
UserName Attribute op Value
Sipuser SIP-AVP = #102:0x08
Openser.cfg:
if (uri=~"^sip:[0-9]{10}@") {
# Send to PSTN Gateway
if (avp_copy("i:102","i:302/n") && avp_check("i:302", "and/i:0x01"))
{
route(4); #NAT Test
route(5); #PSTN Route
} else {
xlog("L_N", " CALL NOT ALLOWED: from $fU to $rU, please
check $fU's ACL\n");
sl_send_reply("503", "Service Unavailable");
};
return;
};
This is the debug message that I get from openser:
0(25135) ERROR:avpops:copy_avp: cannot convert str to int
# ------------------------------------------------------------------------
And finally, I tried to force it by having the value as string, but try to
use the ANDing op, like so:
UserName Attribute op Value
Sipuser SIP-AVP = #102:0x08
Openser.cfg:
...
if (uri=~"^sip:[0-9]{10}@") {
# Send to PSTN Gateway
if (avp_check("i:102", "and/i:0x01")) {
route(4); #NAT Test
route(5); #PSTN Route
} else {
xlog("L_N", " CALL NOT ALLOWED: from $fU to $rU, please
check $fU's ACL\n");
sl_send_reply("503", "Service Unavailable");
};
return;
};
...
This is the debug message that I get from openser:
0(25140) ERROR:avpops:check_avp: value types don't match
# ------------------------------------------------------------------------
It seems to me that the problem lies in auth_radius by not allowing
extraction of AVPs with numerical values in hex. If we can get that work
work, I think the rest should work fine without any alterations.
Thanks for your help in advance!
Lenir
Does anyone have a solution for eliminating multiple INVITE's in the
accounting table (acc) as a result of the caller putting someone on
hold?
Here's my accounting section:
...
modparam("acc", "log_level", 1)
modparam("acc", "log_flag", 1)
modparam("acc", "failed_transactions", 1)
modparam("acc", "report_ack", 0)
...
route[1] {
# -----------------------------------------------------------------
# Default Message Handler
# -----------------------------------------------------------------
t_on_reply("1");
if (method=="INVITE" || method=="ACK" || method=="BYE" ||
method=="CANCEL") {
setflag(1);
};
if (!t_relay()) {
if (method=="INVITE" || method=="ACK") {
end_media_session();
};
sl_reply_error();
};
}
I was interested in getting feedback from current mail group users.
We have mirrored your mail list in a new application that provides a more aggregated and safe environment which utilizes the power of broadband.
Roomity.com v 1.5 is a web 2.01 community webapp. Our newest version adds broadcast video and social networking such as favorite authors and an html editor.
It?s free to join and any feedback would be appreciated.
S.
--------------------------------------------------------------------------------------------------------------------------------------
Broadband interface (RIA) + mail box saftey = <a href="http://SIP_Router_Users_List.roomity.com">SIP_Router_Users_List.roomity.com</a>
*Your* clubs, no sign up to read, ad supported; try broadband internet. ~~1131397872002~~
--------------------------------------------------------------------------------------------------------------------------------------
hi
I had installed ser with media and rtp proxy. Below is the output that i get when I start ser. However, iam not able receive any audio transmission when I connect through sip communicator when iam behind NAT. I am able to register with the server.
0(0) fixing /usr/local/lib/ser/modules/maxfwd.so mf_process_maxfwd_header
0(0) fixing /usr/local/lib/ser/modules/sl.so sl_send_reply
0(0) fixing /usr/local/lib/ser/modules/sl.so sl_send_reply
0(0) fixing /usr/local/lib/ser/modules/mediaproxy.so client_nat_test
0(0) fixing /usr/local/lib/ser/modules/textops.so search
0(0) module - fixing ^Record-Route:
0(0) fixing /usr/local/lib/ser/modules/mediaproxy.so fix_contact
0(0) fixing /usr/local/lib/ser/modules/nathelper.so fix_nated_sdp
0(0) fixing /usr/local/lib/ser/modules/rr.so record_route
0(0) fixing /usr/local/lib/ser/modules/rr.so loose_route
0(0) fixing /usr/local/lib/ser/modules/textops.so append_hf
0(0) fixing /usr/local/lib/ser/modules/textops.so append_hf
0(0) fixing /usr/local/lib/ser/modules/registrar.so lookup
0(0) get_connection(): Connection not found in the pool
0(0) preload_udomain(): Table is empty
0(0) release_connection(): Removing connection from the pool
0(0) fixing /usr/local/lib/ser/modules/auth_db.so www_authorize
0(0) get_connection(): Connection not found in the pool
0(0) release_connection(): Removing connection from the pool
0(0) fixing /usr/local/lib/ser/modules/auth.so www_challenge
0(0) fixing /usr/local/lib/ser/modules/registrar.so save
0(0) get_connection(): Connection not found in the pool
0(0) release_connection(): Removing connection from the pool
0(0) fixing /usr/local/lib/ser/modules/textops.so append_hf
0(0) fixing /usr/local/lib/ser/modules/registrar.so lookup
0(0) fixing /usr/local/lib/ser/modules/sl.so sl_send_reply
0(0) fixing /usr/local/lib/ser/modules/textops.so append_hf
0(0) fixing /usr/local/lib/ser/modules/textops.so search
0(0) module - fixing ^Route:
0(0) fixing /usr/local/lib/ser/modules/sl.so sl_send_reply
0(0) fixing /usr/local/lib/ser/modules/mediaproxy.so use_media_proxy
0(0) fixing /usr/local/lib/ser/modules/tm.so t_on_reply
0(0) fixing /usr/local/lib/ser/modules/tm.so t_relay
0(0) fixing /usr/local/lib/ser/modules/sl.so sl_reply_error
0(0) fixing /usr/local/lib/ser/modules/mediaproxy.so fix_contact
0(0) fixing /usr/local/lib/ser/modules/mediaproxy.so use_media_proxy
0(0) fixing /usr/local/lib/ser/modules/mediaproxy.so client_nat_test
0(0) fixing /usr/local/lib/ser/modules/mediaproxy.so fix_contact
0(0) INFO: udp_init: SO_RCVBUF is initially 110592
0(0) DEBUG: udp_init: trying SO_RCVBUF: 221184
0(0) DEBUG: setting SO_RCVBUF; set=221184,verify=221184
0(0) DEBUG: udp_init: trying SO_RCVBUF: 223232
0(0) DEBUG: setting SO_RCVBUF; set=223232,verify=221184
0(0) DEBUG: setting SO_RCVBUF has no effect
0(0) INFO: udp_init: SO_RCVBUF is finally 221184
0(0) DBG: open_uac_fifo: opening fifo...
0(0) DEBUG: FIFO created @ /tmp/ser_fifo
0(0) DEBUG: fifo /tmp/ser_fifo opened, mode=432
0(0) init_unixsock_socket: No unix domain socket will be opened
2(0) INFO: fifo process starting: 19946
2(0) DEBUG: init_mod_child (-2): tm
2(0) DEBUG: callid: '57b7f7f7-0(a)69.229.23.57'
2(0) DEBUG: init_mod_child (-2): usrloc
2(0) get_connection(): Connection not found in the pool
0(19941) DEBUG: init_mod_child (1): tm
0(19941) DEBUG: callid: '57b7f7f7-19941(a)69.229.23.57'
0(19941) DEBUG: init_mod_child (1): usrloc
0(19941) get_connection(): Connection not found in the pool
1(0) DEBUG: init_mod_child (-1): tm
1(19945) DEBUG: callid: '57b7f7f7-19945(a)69.229.23.57'
1(19945) DEBUG: init_mod_child (-1): usrloc
1(19945) get_connection(): Connection not found in the pool
0(19941) DEBUG: init_mod_child (1): auth_db
0(19941) get_connection(): Connection not found in the pool
1(19945) DEBUG: init_mod_child (-1): auth_db
1(19945) get_connection(): Connection not found in the pool
0(19941) DEBUG: init_mod_child (1): domain
0(19941) get_connection(): Connection found in the pool
0(19941) DEBUG: init_mod_child (1): nathelper
2(19946) DEBUG: init_mod_child (-2): auth_db
2(19946) get_connection(): Connection not found in the pool
1(19945) DEBUG: init_mod_child (-1): domain
1(19945) DEBUG: init_mod_child (-1): nathelper
0(19941) rtpp_test: RTP proxy found, support for it enabled
1(19945) rtpp_test: RTP proxy found, support for it enabled
2(19946) DEBUG: init_mod_child (-2): domain
2(19946) DEBUG: init_mod_child (-2): nathelper
2(19946) rtpp_test: RTP proxy found, support for it enabled
2(19946) SER: open_uac_fifo: fifo server up at /tmp/ser_fifo...
2(19946) DEBUG: register_fifo_cmd: new command (print) registered
2(19946) DEBUG: register_fifo_cmd: new command (uptime) registered
2(19946) DEBUG: register_fifo_cmd: new command (version) registered
2(19946) DEBUG: register_fifo_cmd: new command (pwd) registered
2(19946) DEBUG: register_fifo_cmd: new command (arg) registered
2(19946) DEBUG: register_fifo_cmd: new command (which) registered
2(19946) DEBUG: register_fifo_cmd: new command (ps) registered
2(19946) DEBUG: register_fifo_cmd: new command (kill) registered
2(19946) WARNING: no fifo_db_url given - fifo DB commands disabled!
Let me know if you find any configuration error in SER.
Please let me if you need any more information. I am trying to configure SER from long time. Please help me.
Thanks
Rajesh
---------------------------------
Enjoy this Diwali with Y! India Click here
---------------------------------
Enjoy this Diwali with Y! India Click here
Hello All,
I'm trying (desperately) to use avp_db_load with an avp:
version: ser 0.9.3 (i386/linux)
flags: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535
@(#) $Id: main.c,v 1.197 2004/12/03 19:09:31 andrei Exp $
main.c compiled on 12:28:21 Aug 16 2005 with gcc 3.3
modparam("auth_db","avps_column_str","callscope")
modparam("avpops","avp_aliases","callgroups=i:200")
modparam("avpops","avp_aliases","callscope=i:500")
avp_db_load("$callscope","$callgroups/avps") ;
In watching the debug, I see the avp loaded just fine from the database from the auth_db parameter:
3(1735) DEBUG:avpops:print_avp: p=0xf51c20d8, flags=3
3(1735) DEBUG: name=<callscope>
3(1735) DEBUG: val_str=<200>
but the above avp_db_load line is resulting in:
3(1735) DEBUG:avpops:get_val_as_str: no avp found
3(1735) ERROR:avpops:load_avps: failed to get uuid
I also notice that it doesn't even bother to hit the database after the auth_db hit. So unless the avp table has been cached (Which I have no idea if it has..) how would it know?
Seems like error #1: no avp found? Seems like it doesn't like my syntax but according to the avpops documentation, it should be right:
source = (sip_uri|avp_alias|str_value) ['/'('username'|'domain'|'uri'|'uuid')]
note:
avp_db_load("$callscope/uuid","$callgroups/avps") ;
gives me:
0(0) ERROR:avpops:fixup_db_avp: source/flags "callscope" unknown!
Look, what I really want to do is use the authentication credentials to load an avp. Isn't that was avp is for? I don't want the FROM or the TO or the [R]URI, but the actual AUTHentication username. Any ideas??
Thanks!!
BTW, I've tried the newest SER DEVEL CVS, and OPENSER 1.0 with basically the same results. Some custom modules I have would also make me want to stick with 0.9.0.
-Brett
Hi,
I want to use voip for a medium size subscriber, i suggest 10.000
i want to;
subscribers talk each other with no charge
some one can get 2 or 4 subscribing for faks or something else
i want voicemail option for people who can not answer calls
maybe conference call
Call logs for incoming and outgoing
Phonebook maybe
first i thougt that ser can do this but i dont know if it can
i use suse 9.3 /mysql/php on an intel desktop pc (not a server) with p.4
3ghz 1 Gbit DDR2-533
there are some programs i did not understand their roles;
Radius?
Asteriks?
Sim?
Which programs i should use to do them?
Please give information,
Thanks
Toygun
hi,
i run an SER which supports multiple realms. all works fine,
authentication, calls etc. except one thing:
we are in the rare sitiuation of having the same number-block in two
different cities. i have aliases for my users (username is areacode
+number) which is only the number (without area-code) to support
dialing without area-code. but in the case with the two same blocks
in different area-codes the alias handling gets a bit confused:
# serctl alias show 88999998
<sip:0471288999998@04712.sip.domain.de>;q=1;expires=459311264
<sip:0471188999998@04711.sip.domain.de>;q=1;expires=459311264
so i have two aliases for the same number. any of the numbers is in
different SIP_DOMAINS, namely 04712.sip... and 04711.sip..., but even
when i state the SIP_DOMAIN to use for the lookup stuff i get:
# SIP_DOMAIN=04712.sip.domain.de serctl alias show 88999998
<sip:0471288999998@04712.sip.domain.de>;q=1;expires=459311207
<sip:0471188999998@04711.sip.domain.de>;q=1;expires=459311207
and of course the other way round with 04711. the realm/domain is
stored within the alias, so there are two records:
88999998 04711.sip.domain.de sip:0471188999998@04711.sip.domain.de ...
88999998 04712.sip.domain.de sip:0471288999998@04712.sip.domain.de ...
with different domains. now, what happens is: when a user from the
04711 area-code calls another user (for him in that same area), he/
she might end up reaching a user with the 04712 area-code. even
though the INVITE messages and all state, that the particular (any)
user is from the 04711 domain, or 04712 for that matter ...
i also tried to name the alias 88999998(a)04712.sip... and so on, but
then, it doesn't even find an alias at all.
what did i miss?
oh, and yes, both realms are defined via an alias-directive in
SER.cfg so he treats both as local (or at least it should).
any hint would be appreciated.
brgds,
g. feiner
I have installed openser v1.0 and it is working perfectly. But I have
a problem with the xlog module. Many times I've got the following
error:
Nov 7 11:28:34 localhost openser[1736]: xl_printf: buffer overflow --
increase the buffer size...
The line that produces this error is this (it is inside a case stament):
xlog("L_ERR","Has been selected the 1 index");
I have tried to increase the buffer size in the following form:
modparam("xlog","buf_size",8192)
No matters the size that I put on the line buf_size, I always get the
same error.
I thing that a buffer size of 8kb should be enough for this log
message, isn't it?
Thanks.
--
Salu2:
***************************************
José María González
***************************************
