Hello,
I have downloaded and intalled ser v2 from http://www.iptel.org/ser_2_0_0_release_candidate_1
Followd all the instructions.
In Section Creating your first config.
Run /usr/local/sbin/sercmd and type ctl.listen to see if SER is running and listening. Type help to see other commands available.
I get an error.
[root@localhost serweb-0.9.6]# /usr/local/sbin/ser -f /usr/local/etc/ser/ser.cfg
Listening on
???????????? udp: 127.0.0.1 [127.0.0.1]:5060
???????????? udp: 192.168.1.117 [192.168.1.117]:5060
???????????? tcp: 127.0.0.1 [127.0.0.1]:5060
???????????? tcp: 192.168.1.117 [192.168.1.117]:5060
Aliases:
???????????? tcp: localhost:5060
???????????? tcp: localhost.localdomain:5060
???????????? udp: localhost:5060
???????????? udp: localhost.localdomain:5060
[root@localhost serweb-0.9.6]# /usr/local/sbin/sercmd
ERROR: connect_unix_sock: connect(/tmp/ser_ctl): No such file or directory [2]
can someone tell me why am i getting that error.
Roman
________________________________________________________________________
AOL now offers free email to everyone. Find out more about what's free from AOL at AOL.com.
Hey Everyone,
when i try to load module perl.so, i get this message :
ERROR: load_module: could not open module
</usr/local/lib/openser/modules/perl.so>:
/usr/local/lib/openser/modules/perl.so: undefined symbol: boot_OpenSER
In /usr/local/lib/openser/modules/ i have not only perl.so but also the
lib/perl dir.
openser-1.2.0-tls or openser-1.2.1-tls
Any idea ?
Thanks,
--
Richard Timsit <Richard.Timsit(a)epfl.ch>
EPFL
Running OpenSER v1.1.1 with internal ip for sip client connections
and ext ip to connect to sip providers. I am getting loops in my
calls sometimes... They stop after reaching the max hops trying to
send the BYE message to it self. I am wondering if it is the record-
route that I see that includes the internal IP. The 483 error is
U 2007/07/30 00:49:56.133157 192.168.18.22:5060 -> 192.168.18.22:5060
SIP/2.0 483 Too Many Hops.
From: <sip:15126461516@192.168.18.1:5060>;tag=nu-481d-6c62.
To: <sip:18065436468@192.168.18.22:5060>;tag=6738.
Call-ID: 421ac0c6-1dd2-11b2-8093-df998124da20(a)192.168.18.91.
CSeq: 11269143 BYE.
Via: SIP/2.0/UDP 192.168.18.22;rport=5060;branch=z9hG4bKe6ff.7fe832a5.0.
Via: SIP/2.0/UDP 192.168.18.22;rport=5060;branch=z9hG4bKe6ff.6fe832a5.0.
Via: SIP/2.0/UDP 192.168.18.22;rport=5060;branch=z9hG4bKe6ff.5fe832a5.0.
Via: SIP/2.0/UDP 192.168.18.22;rport=5060;branch=z9hG4bKe6ff.4fe832a5.0.
Via: SIP/2.0/UDP 192.168.18.22;rport=5060;branch=z9hG4bKe6ff.3fe832a5.0.
Via: SIP/2.0/UDP 192.168.18.22;rport=5060;branch=z9hG4bKe6ff.2fe832a5.0.
Via: SIP/2.0/UDP 192.168.18.22;rport=5060;branch=z9hG4bKe6ff.1fe832a5.0.
Via: SIP/2.0/UDP 192.168.18.22;rport=5060;branch=z9hG4bKe6ff.0fe832a5.0.
Via: SIP/2.0/UDP 192.168.18.22;rport=5060;branch=z9hG4bKe6ff.fee832a5.0.
Via: SIP/2.0/UDP 192.168.18.22;rport=5060;branch=z9hG4bKe6ff.eee832a5.0.
Via: SIP/2.0/UDP 192.168.18.91:5060;rport=5060.
Server: OpenSer (1.1.1-notls (i386/linux)).
Content-Length: 0.
Any ideas?
Hi
iam installing openser SVN and Freeradius 1.16
followed the link as mentioned in document
when i run radiusd -X
iam getting below problem
1. rlm_sql (sql): Connected new DB handle, #4
2. Module: Instantiated sql (sql)
3. Module: Loaded Acct-Unique-Session-Id
4. acct_unique: key = "User-Name, Acct-Session-Id, NAS-IP-Address,
Client-IP-Address, NAS-Port"
5. rlm_acct_unique: Cannot find attribute 'NAS-IP-Address' in
dictionary
6. radiusd.conf[1159]: acct_unique: Module instantiation failed.
7. radiusd.conf[1753] Unknown module "acct_unique".
8. radiusd.conf[1747] Failed to parse preacct section.
any suggestiong
ram
Hello everyone,
What is the preferred way of handling PRACKS? simply relay them if
they are in loose_route? should they be authenticated?
What, if any, are the potential risks of not authenticating PRACKS?
Thanks in advance.
--
Zahid
Hi,
I'm facing an issue with openser. I checked the latest SVN
code and it seems my issue is not solved even in the svn version.
My UA is sending an SDP with 2 m lines: one for audio
and one for video. In a re-INVITE, I want to disable video
and I'm sending m=video 0 RTP/AVP..
My openser is currently modifying the port number for the
video line even if it's 0 which means it's disabled...
This behavior would of course happen even for audio lines
that ones wants to disable.
I was not able to find a clean fix in nathelper.c and hope
to get some help for proposing one... I guess it would be
something close to:
if (oldport==0)
skip altering ip/port for media and continue;
but I don't want to insert any mem leak or lose a session...
tks,
Aymeric MOIZARD / ANTISIP
amsip - http://www.antisip.com
osip2 - http://www.osip.org
eXosip2 - http://savannah.nongnu.org/projects/exosip/
Dear sir:
First,I am a Chinese,and my English is poor.^_^
We are testing kinds of applications on IPV6 including VOIP. And we use Linphone as our UA. If I don't use domain name,how can I add a account? On IPV4,I add account " # openserctl add test test test@ipv4_address" .And in the file "openser.cfg" ,I directly write my SIP server ipv4_address . Can I directly replace "ipv4_address" by "ipv6_address"? I has tried,and the resault is bad.
I am very sorry that I am amateurish for VOIP ,because Our work is only to test.
Thank you!
Li Zhonglei
-------------------------------------------------------------------
吉贝克 ——商业智能的领军者( http://d1.sina.com.cn/sina/limeng3/mail_zhuiyu/2007/mail_zhuiyu_20070723.ht… )
===================================================================
注册新浪2G免费邮箱( http://mail.sina.com.cn/chooseMode.html )
Hi there,
I'm doing sipp performance measurements with openser 1.2.
I run OpenSER on machine A, then run one SIPp on machine B, one
SIPp on machine C.
For the SIPp on machine C, I start it using UAS mode, ie., ./sipp -sn uas
For the SIPp on machine B, if I start is using the following command:
./sipp -sn uac C -rsa A -m 1 -r 1
Everything is correct. The OpenSER can relay messages correctly.
But if I start SIPp on machine B using the following command:
./sipp -sn uac -rsa A -m 1 -r 1 -d 5000
Then the SIPp client on machine B will crash because of receiving unexpected
"ringing 180" message. Note: the only difference is I add a pause 5 seconds
using "-d 5000" option.
What I found is: the OpenSER server will keep sending "INVITE" to C
during the '5 second pause', so C will keep reponding with "180 Ringing" to
B.
Does anybody know why OpenSER keeps sending "INVITE" to C while B only
issues one "INVITE" request?
BTW, the configuration file for OpenSER is very simple:
#
# $Id: openser.cfg 1676 2007-02-21 13:16:34Z bogdan_iancu $
#
# simple quick-start config script
# Please refer to the Core CookBook at
http://www.openser.org/dokuwiki/doku.php
# for a explanation of possible statements, functions and parameters.
#
# ----------- global configuration parameters ------------------------
debug=3 # debug level (cmd line: -dddddddddd)
fork=yes
log_stderror=no # (cmd line: -E)
children=32
log_facility=LOG_LOCAL0
disable_tcp=yes
disable_dns_blacklist=yes
disable_dns_failover=yes
# Uncomment these lines to enter debugging mode
#fork=no
#log_stderror=yes
#
listen=udp:192.168.2.102:5060
# ------------------ module loading ----------------------------------
loadmodule "modules/tm/tm.so"
modparam("tm", "wt_timer", 2)
# ------------------------- request routing logic -------------------
# main routing logic
route{
t_relay();
}
Thanks a lot!
-Bo Zhang
Hi, in URI as "ping(a)domain.org" I need to change URI to just "domain.org"
(without the username@). But I have multidomain so can't rewrite uri directly
and must use the requested URI domain ($rd).
So I try:
if (uri=~"sip:ping[@]+.*") {
rewriteuri("sip:$rd");
xlog("L_INFO", "-- New URI = $ru\r\n");
...
but I get:
-- New URI = sip:$rd
so it seems that I cannot use variables for making function parameters, is it
true? so then, how should I do it? maybe I need using AVP's for that?
Thanks for all.
--
Iñaki Baz Castillo
ibc(a)in.ilimit.es
Hi, I'd like to understand how TM module filters "sometimes" some calls. For
now, I just think it there isn't response and the call is repeated sometimes
then TM module filteres the following calls.
I find no info about how it works in the doc, could somebody give me a link or
so to a document with the explanation of this issue?
Thanks a lot.
--
Iñaki Baz Castillo
ibc(a)in.ilimit.es