Hi Friends,
In SER I am subscribing and i am getting 202 Ok also notify but the publish
method data is is not coming to me with notify although after sending
publish i am getting 200 Ok. Can any one suggest me regarding this.
Thanks and Regards,
Ambed
Forgot to CC to the list.
> Hi,
>
>> I never worked with FIFO, and I need exec one command in fifo, the
>> trusted_reload !
>
> OSER_FIFO has to be "/tmp/openser_fifo" in your openserctlrc. Then just
> try "openserctl fifo trusted_reload", and you should get a "200 OK" if
> successful.
>
> Cheers,
> Andreas
>
Hi list,
I never worked with FIFO, and I need exec one command in fifo, the
trusted_reload !
********** openser.cfg **********************
alias="insta.voip.nce.ufrj.br"
mpath="/lib/openser/modules"
children=8
tcp_children=4
debug=3
fork=yes
log_facility=LOG_DAEMON
log_stderror=yes
group="openser"
user="openser"
sock_group="openser"
sock_user="openser"
disable_tcp=no
mhomed=no
server_signature=yes
sock_mode=0600
unix_sock="/tmp/openser.sock"
unix_sock_children=1
reply_to_via=no
sip_warning=no
check_via=no
dns=no
rev_dns=no
disable_core_dump=no
dns_try_ipv6=yes
dns_use_search_list=yes
.......
loadmodule "mi_fifo.so"
modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo")
modparam("mi_fifo", "fifo_mode", 0660)
modparam("mi_fifo", "fifo_group", "openser")
modparam("mi_fifo", "fifo_user", "openser")
modparam("mi_fifo", "reply_dir", "/tmp/")
modparam("mi_fifo", "reply_indent", "\t")
......
**************** openserctl *****************
# $Id: openserctlrc 1827 2007-03-12 15:22:53Z bogdan_iancu $
#
# openser control tool resource file
#
# here you can set variables used in the openserctl
## your SIP domain
SIP_DOMAIN=insta.voip.nce.ufrj.br
## database type: MYSQL or PGSQL, by defaulte none is loaded
DBENGINE=PGSQL
## database host
DBHOST=146.164.247.209
## database name
DBNAME=openser
## database read/write user
DBRWUSER=openser
## database read only user
# DBROUSER=openserro
## password for database read only user
DBROPW=openserro
## database super user
DBROOTUSER="postgres"
## type of aliases used: DB - database aliases; UL - usrloc aliases
## - default: none
ALIASES_TYPE="DB"
## control engine: FIFO or UNIXSOCK
## - default FIFO
CTLENGINE="UNIXSOCK"
## path to FIFO file
OSER_FIFO="/tmp/"
## check ACL names; default on (1); off (0)
VERIFY_ACL=1
## ACL names - if VERIFY_ACL is set, only the ACL names from below list
## are accepted
# ACL_GROUPS="local ld int voicemail free-pstn"
## presence of serweb tables - default "no"
# HAS_SERWEB="yes"
## verbose - debug purposes - default '0'
VERBOSE=1
## do (1) or don't (0) store plaintext passwords
## in the subscriber table - default '1'
# STORE_PLAINTEXT_PW=0
***************************
My doubt is what my conf is correct and what is the command that I use in
the Konsole!?
Thanks in advanced for your dedication to someone that read to here.
Thiago Maluf.
----------------------------------------------------------------
THIAGO MALUF RESENDE
Consultor Voip e Programador WEB (Voip Developer and Web Developer)
Tel: +55 21 86042100
e-mail: malufrj(a)gmail.com
Dear sir,
We install 3 openser 1.1.0 in public internet, and we find a BYE
relay issue to caller SIP UA.
UA(a)-------Proxy1 --------Proxy2 ----Proxy 3 ---------UA(b)
We make call from UA(a) to UA (b), if UA(b) send Bye first, UA(a) can
not get Bye.
The packet is from Proxy2 to Proxy1 as following
BYE sip:1.2.3.4:5060;nat=yes;ftag=ooqjvbi7ql;lr=on SIP/2.0
Record-Route: <sip:5.6.7.8;lr=on;ftag=as1b41fbeb>
Record-Route: <sip:5.6.7.8;lr=on;ftag=as1b41fbeb>
Via: SIP/2.0/UDP 5.6.7.8;branch=z9hG4bKc74c.bd89ce13.0
Via: SIP/2.0/UDP 5.6.7.8;branch=z9hG4bKc74c.ad89ce13.0
Via: SIP/2.0/UDP 100.200.100.200;branch=z9hG4bKc74c.321ef5e3.0
Via: SIP/2.0/UDP 100.200.100.199:5060;branch=z9hG4bK75e8d6f3;rport=5060
From: <sip:0908900000@sip.edu.tw;user=phone>;tag=as1b41fbeb
To: "0909802020 kl" <sip:0909802020@sip.edu.tw>;tag=ooqjvbi7ql
Call-ID: 3c275da915f9-iylvmp2wjuh0@snom360-00041323058B
CSeq: 102 BYE
User-Agent: Gentrice_IPPBX
Max-Forwards: 67
Content-Length: 0
Then bye is looping in proxy1.
BYE sip:1.2.3.4:5060;nat=yes;ftag=ooqjvbi7ql;lr=on SIP/2.0
Record-Route: <sip:5.6.7.8;lr=on;ftag=as1b41fbeb>
Record-Route: <sip:5.6.7.8;lr=on;ftag=as1b41fbeb>
Via: SIP/2.0/UDP 1.2.3.4;branch=z9hG4bKc74c.1714a4e7.0
Via: SIP/2.0/UDP 5.6.7.8;branch=z9hG4bKc74c.bd89ce13.0
Via: SIP/2.0/UDP 5.6.7.8;branch=z9hG4bKc74c.ad89ce13.0
Via: SIP/2.0/UDP 100.200.100.200;branch=z9hG4bKc74c.321ef5e3.0
Via: SIP/2.0/UDP 100.200.100.199:5060;branch=z9hG4bK75e8d6f3;rport=5060
From: <sip:0908900000@sip.edu.tw;user=phone>;tag=as1b41fbeb
To: "0909802020 kl" <sip:0909802020@sip.edu.tw>;tag=ooqjvbi7ql
Call-ID: 3c275da915f9-iylvmp2wjuh0@snom360-00041323058B
CSeq: 102 BYE
User-Agent: Gentrice_IPPBX
Max-Forwards: 66
Content-Length: 0
We used to use default script for proxy1, but the same...
And we do record_route, loose_route relay as well.
Anyone know what happen?
best regards
Thrli
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Hello WeSIPer,
I found a little but annoying Error in WeSIP's C2C application. I send
back a 483 reply on a C2C's first caller-INVITE to WeSIP via openser's
sl-send_reply(). I expected that C2C terminates the call function, cause
C2C has an error handling method for 4xx replies, but nothing happens.
openser as wesip's frontend received definately the reply.
By chance I found, that the reason for this was my reply message.
This doesn't work with WeSIP's C2C application:
sl_send_reply("483", "My message\n");
But this works:
sl_send_reply("483", "My message");
It seems, that C2C's or WeSIPs reply parser has some problems with
linefeeds at the end of reply's message text.
I use openser 1.2.0, WeSIP 0.1.4 and click2call-08_11_06
regards
Helmut
PS: Is there already a fix for WeSIP to reconnect to openser's seas
socket, when openser restarts?
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Version: GnuPG v1.4.5 (MingW32)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org
iD8DBQFGUV2k4tZeNddg3dwRAsPcAKCNqGE+XTK7BvW69x4YT498Bg0dkgCbB3eL
6Me/ZKYaqxfo/qDGB8M3lg4=
=jTel
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Hi guys,
I have OpensER 1.1.0 in fornt of 2 Asterisks servers.
My OpenSER act as load balancer for outbound calls and for registrar
server for inbound calls coming from Asterisks PSTN GW to Users
Some users, using AVM Fritz Fon Boxes cannot register on the proxy, they
get a "483 Too Many Hops".
They say it happens only if they configure the Fritx using FQDN of our
proxy, if they put its IP addess in it they are having no problems...
(it sounds very strange to me but I captured some trafic and had
confirmation of what they're saying)
Here is a SIP dialog with the error message
Request-Line: REGISTER sip:sip.advancemedia.it SIP/2.0
Via: SIP/2.0/UDP
192.168.1.121:5060;branch=z9hG4bKC0F2D683E6BE8AC5F1A4A260BE120
Route: <sip:sip.advancemedia.it;lr>
From: <sip:cirifax@sip.advancemedia.it>;tag=1898319928
To: <sip:cirifax@sip.advancemedia.it>
Call-ID: 9ADCCD22187BDEC93B5953DD93E0F(a)192.168.1.121
CSeq: 1 REGISTER
Max-Forwards: 70
User-Agent: AVM FRITZ!Box Fon ata 1020 11.04.01 (Jan 25 2006)
Supported: 100rel, replaces
Allow-Events: telephone-event, refer
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, UPDATE, PRACK, INFO,
SUBSCRIBE, NOTIFY, REFER, MESSAGE
Accept: application/sdp, multipart/mixed
Accept-Encoding: identity
Content-Length: 0
Status-Line: SIP/2.0 483 Too Many Hops
Message Header
Via: SIP/2.0/UDP
192.168.1.121:5060;received=80.67.115.96;branch=z9hG4bKC0F2D683E6BE8AC5F1A4A260BE120
From: <sip:cirifax@sip.advancemedia.it>;tag=1898319928
To:
<sip:cirifax@sip.advancemedia.it>;tag=b36cb132c8240044969e91db40b97d00.d5d1
Call-ID: 9ADCCD22187BDEC93B5953DD93E0F(a)192.168.1.121
CSeq: 1 REGISTER
Server: OpenSer (1.1.0-notls (i386/linux))
Content-Length: 0
Warning: 392 213.92.79.131:5060 "Noisy feedback tells: pid=13326
req_src_ip=213.92.79.131 req_src_port=5060
in_uri=sip:sip.advancemedia.it out_uri=sip:sip.advancemedia.it via_cnt==71"
It doesn't happen if the user put the ip of the proxy server in the
configuration of the Fritz
I guess it's a problem with loose routing but I cannot understand if
it's a buggy implementation of SIP in the fritz or my misconfiguration
Here is my openser.cfg, tnx in advance for help.
Edoardo
# $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $
# ----------- global configuration parameters ------------------------
check_via=yes # (cmd. line: -v)
dns=no # (cmd. line: -r)
rev_dns=no # (cmd. line: -R)
fifo="/tmp/ser_fifo"
# ------------------ module loading ----------------------------------
loadmodule "/usr/lib/openser/modules/sl.so"
[...]
modparam("usrloc", "db_mode", 2)
modparam("usrloc", "db_url", "mysql://user:pass@192.168.252.5/openser")
modparam("usrloc", "timer_interval", 120)
modparam("auth_db", "calculate_ha1", 0)
modparam("auth_db", "db_url", "mysql://user:pass@192.168.252.5/db")
modparam("uri_db", "db_url", "mysql://user:pass@192.168.252.5/openser")
modparam("rr", "enable_full_lr", 1)
modparam("registrar", "nat_flag", 6)
modparam("registrar", "max_expires", 3600)
modparam("registrar", "min_expires", 60)
modparam("registrar", "append_branches", 0)
modparam("registrar", "desc_time_order", 1)
modparam("nathelper", "natping_interval", 20) # Ping interval 20 s
modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT
modparam("dispatcher", "force_dst", 1)
# ------------------------- request routing logic -------------------
# main routing logic
route{
if (!mf_process_maxfwd_header("10")) {
sl_send_reply("483","Too Many Hops");
exit;
};
if ( msg:len > max_len ) {
sl_send_reply("513", "Message too big");
exit;
};
if ( (method=="OPTIONS") || (method=="SUBSCRIBE") || (method=="NOTIFY") ) {
sl_send_reply("405", "Method Not Allowed");
exit;
}
# loose-route processing
if (loose_route()) {
# mark routing logic in request
append_hf("P-hint: rr-enforced\r\n");
t_relay();
exit;
};
if (!method=="REGISTER") {
record_route();
};
if ((src_ip==ip.of.asterisk.1) || (src_ip==ip.of.asterisk.1)) {
if (!lookup("location")) {
sl_send_reply("486", "Busy here");
exit;
};
t_on_reply("1");
if (!t_relay()) {
sl_reply_error();
};
exit;
};
if (nat_uac_test("3")) {
if ((method=="REGISTER") || (method=="INVITE") || (method=="OPTIONS")) {
fix_nated_contact();
force_rport();
setflag(6); # Mark as NATed
}
}
if (method=="REGISTER") {
if (!proxy_authorize("exorsa", "openser_view")) {
proxy_challenge("exorsa", "0");
exit;
}
if (!check_to()) {
sl_send_reply("403", "Digest username and URI username do NOT match!
Stay away!");
exit;
}
save("location");
exit;
};
if (method=="INVITE") {
if (!proxy_authorize("exorsa", "openser_view")) {
proxy_challenge("exorsa", "0");
exit;
}
if (!check_from()) {
sl_send_reply("403", "Digest username and URI username do NOT match!
Stay away!");
exit;
}
}
if (!uri==myself) {
# mark routing logic in request
append_hf("P-hint: outbound\r\n");
route(1);
exit;
};
append_hf("P-hint: usrloc applied\r\n");
route(1);
}
route[1]
{
# !! Nathelper
if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)" &&
!search("^Route:")){
sl_send_reply("479", "We don't forward to private IP addresses");
exit;
};
# NAT processing of replies; apply to all transactions (for example,
# re-INVITEs from public to private UA are hard to identify as
# NATed at the moment of request processing); look at replies
t_on_reply("1");
if ((src_ip!=ip.of.asterisk.1) && (src_ip!=ip.of.asterisk.2)) {
ds_select_dst("1", "0");
}
if (!t_relay()) {
sl_reply_error();
};
}
# !! Nathelper
onreply_route[1] {
# NATed transaction ?
if (isflagset(6) && status =~ "(183)|2[0-9][0-9]") {
fix_nated_contact();
# otherwise, is it a transaction behind a NAT and we did not
# know at time of request processing ? (RFC1918 contacts)
} else if (nat_uac_test("1")) {
fix_nated_contact();
};
}
Hi,
I need a b2bua to use with openser. I also don't want to use radius. what options do I have? thanks in advance.
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Hi,
At the moment I am using SVN version of OpenSER (2:2253M) also I have
been using the 1.2 release. The problem has been the NOTIFY message
which has been dropping time to time, or like in current svn release, I
do not get the NOTIFY message at all anymore. I think my openser.cfg is
having something wrong, but haven't been able to figure out what is the
reason.
I do use XCAP_Lite 1.0.0 with presence & presence_xml modules. The
force_active is set to "0", because I want to have reactive buddy
authorization. The XCAP side works nicely, also watchers and
active_watchers are populated by OpenSER correctly. But once the
SUBSCRIBE is received from client, no NOTIFY is sent to any clients
which are active in watchers table.
After reading the log file I found this message from it:
0(4415) PRESENCE: publ_notify: Could not get subs_dialog from database
Thank you for tips,
-Mika Saari