Hi all,
i'm happy to announce a new developer for the kamailio and sip-router project:
Marius Zbihlei.
Marius is a experienced C/C++ developer in our offices in Bukarest (Romania).
He develops since a few years internal applications for the web hosting
plattform of 1&1. Since app. a half year he is part of the team that handles
the development and maintenance of our VoIP backend systems. Most of his work
so far was targeted to our internal repositories, but Marius also already
contributed a bunch of patches to the project. He'll support us with the
maintenance of the kamailio code we contributed to the project in the past,
and also work on new sip-router features in the future.
Best regards,
Henning
--
Henning Westerholt - Development Consumer Products / Consumer Core
1&1 Internet AG, Ernst-Frey-Str. 9, 76135 Karlsruhe, Germany
Hello,
On 05.10.2009 17:29 Uhr, ?? wrote:
>
> hello,
>
>
> I am doing MC simulation at STAR. I met this problem when I compile.
> which mail list should I send?
>
had no idea what "mc simulation at star" is.
Do you use kamailio sip server?
http://www.kamailio.org
If not, you are wrong here and I do not know what mailing list you
should use.
Daniel
> Thanks.
>
>
> xueliang
>
> *************************************************************
> L Xue
> Building 102, Room 102
> Department of Nuclear Physics
> Shanghai Institute of Applied Physics, CAS
> No. 2019, JiaLuo Roud
> Shanghai, 201800
> P. R. China
> Email:xueliang@sinap.ac.cn
> MSN: xliang1984(a)hotmail.com
> **************************************************************
>
>
>
> -----Original Message-----
> From: Daniel-Constantin Mierla [mailto:miconda@gmail.com]
> Sent: Mon 2009-10-5 16:21
> To: ??
> Cc: Users(a)lists.kamailio.org
> Subject: Re: [Kamailio-Users] fs_index.cxx:1: error: bad value
> (x86_64) for -march (mtune)= switch
>
> Hello,
>
> what application do you compile? Is not kamailio, so you probably sent
> to wrong mailing list.
>
> Cheers,
> Daniel
>
> On 04.10.2009 5:57 Uhr, ?? wrote:
> >
> > Hi ,
> >
> >
> > When I try to compile *.cxx on "Quad-Core AMD Opteron(tm) Processor
> 2350"
> > the follow happens:
> >
> > make[1]: Entering directory `/eliza8/rnc/xueliang/MC/StRoot/RTS/src/SFS'
> > g++ -lpthread -lrt -Wall -g -I
> > /afs/rhic//star/ROOT/5.22.00/.sl44_gcc346/rootdeb/include -L
> > /afs/rhic//star/ROOT/5.22.00/.sl44_gcc346/rootdeb/lib -O3 -Wall -pipe
> > -fverbose-asm -march=x86_64 -D_REENTRANT
> > -DRTS_DAQMAN=\""172.16.0.1"\" -DTARGET_SYSTEM=\"LINUX\"
> > -DPROJDIR=\"/RTScache\" -DINSTALL_SUFFIX=\"LINUX/x86_64\"
> > -DRTS_PROJECT_STAR -DRTS_BIG_ENDIAN -DRTS_LOG_COLORED -g -I. -I../
> > -I../../include -I../../trg/include -c -o fs_index.o fs_index.cxx
> > fs_index.cxx:1: error: bad value (x86_64) for -march= switch
> > fs_index.cxx:1: error: bad value (x86_64) for -mtune= switch
> > make[1]: *** [fs_index.o] Error 1
> > make[1]: Leaving directory `/eliza8/rnc/xueliang/MC/StRoot/RTS/src/SFS'
> >
> > How can I solve that
> > Thanks and regards!
> >
> >
> > xueliang
> >
> >
> > *************************************************************
> > L Xue
> > Building 102, Room 102
> > Department of Nuclear Physics
> > Shanghai Institute of Applied Physics, CAS
> > No. 2019, JiaLuo Roud
> > Shanghai, 201800
> > P. R. China
> > Email:xueliang@sinap.ac.cn
> > MSN: xliang1984(a)hotmail.com
> > **************************************************************
> >
> > ------------------------------------------------------------------------
> >
> > _______________________________________________
> > Kamailio (OpenSER) - Users mailing list
> > Users(a)lists.kamailio.org
> > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
> > http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
>
> --
> Daniel-Constantin Mierla
> * Kamailio SIP Masterclass, Nov 9-13, 2009, Berlin
> * http://www.asipto.com/index.php/sip-router-masterclass/
>
>
>
--
Daniel-Constantin Mierla
* Kamailio SIP Masterclass, Nov 9-13, 2009, Berlin
* http://www.asipto.com/index.php/sip-router-masterclass/
I've just setting SIP Express Router and Mediaproxy in 1 PC. Then, I'm
setting in NAT Router that only activated for MASQUERADE like this:
iptables -A POSTROUTING -s 172.17.2.0/24 -o eth1 -j MASQUERADE
NB:I assume in network 10.100.0.0/24 (eth1) as public IP and
172.17.2.0/24as private IP.
When I calling with x-lite and activated mediaproxy service, the connection
running well with 2 direction (full duplex from 10.x.x.x to 172.x.x.x
conversely).But,when I stopping mediaproxy service (for proving that voice
packet if pass NAT will connect half duplex or maybe broken),apparently this
connection still running well with 2 direction
my question,,
why this happened?
are having trouble with mediaproxy or SIP Express Router?
i'm confusing,,
thanks very much,,
:)
Hello,
Siremis - the web management interface for Kamailio SIP server - has
been released with version 0.9.4. This is a patch release fixing the
issues reported since v0.9.3 related to db connection samples in default
configuration file.
More details at:
http://siremis.asipto.com
Best regards,
Ramona Modroiu
It's very frustrating that some functions don't accept pseudovariables
and there's no workaround.
For example, in this case I need rtpproxy_offer() and rtpproxy_answer()
to accept an IP address argument that is retrieved from a database.
However, this won't work.
rtpproxy_offer("", $var(src_ip)); # Config compilation error
rtpproxy_offeR("", "$var(src_ip)"); # Results in $var(src_ip)
# literally being placed in the
# SDP body.
I understand it is difficult to go back and update all legacy functions
to accept PVs everywhere. But isn't it possible to provide a
wrapper/compatibility function in the core that will parse a PV and
generate as a result something that other functions can see as a string
literal?
--
Alex Balashov - Principal
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Hi ,
When I try to compile *.cxx on "Quad-Core AMD Opteron(tm) Processor 2350"
the follow happens:
make[1]: Entering directory `/eliza8/rnc/xueliang/MC/StRoot/RTS/src/SFS'
g++ -lpthread -lrt -Wall -g -I /afs/rhic//star/ROOT/5.22.00/.sl44_gcc346/rootdeb/include -L /afs/rhic//star/ROOT/5.22.00/.sl44_gcc346/rootdeb/lib -O3 -Wall -pipe -fverbose-asm -march=x86_64 -D_REENTRANT -DRTS_DAQMAN=\""172.16.0.1"\" -DTARGET_SYSTEM=\"LINUX\" -DPROJDIR=\"/RTScache\" -DINSTALL_SUFFIX=\"LINUX/x86_64\" -DRTS_PROJECT_STAR -DRTS_BIG_ENDIAN -DRTS_LOG_COLORED -g -I. -I../ -I../../include -I../../trg/include -c -o fs_index.o fs_index.cxx
fs_index.cxx:1: error: bad value (x86_64) for -march= switch
fs_index.cxx:1: error: bad value (x86_64) for -mtune= switch
make[1]: *** [fs_index.o] Error 1
make[1]: Leaving directory `/eliza8/rnc/xueliang/MC/StRoot/RTS/src/SFS'
How can I solve that
Thanks and regards!
xueliang
*************************************************************
L Xue
Building 102, Room 102
Department of Nuclear Physics
Shanghai Institute of Applied Physics, CAS
No. 2019, JiaLuo Roud
Shanghai, 201800
P. R. China
Email:xueliang@sinap.ac.cn
MSN: xliang1984(a)hotmail.com
**************************************************************
Hello,
here is the roadmap planned to get to Kamailio 3.0:
- this week testing using common git repository
- end of this week have Kamailio 3.0 branched
- 2 weeks testing of kamailio 3.0 branch and fix of discovered missing
features from 1.5, packaging selected modules, name updates, installed
tools, documentation updates, etc
- 26 or 27 Oct day of release
Comments/adjustments are welcome!
I also think that we should look at releasing 1.5.3 since there were few
important updates since 1.5.2.
Cheers,
Daniel
--
Daniel-Constantin Mierla
* Kamailio SIP Masterclass, Nov 9-13, 2009, Berlin
* http://www.asipto.com/index.php/sip-router-masterclass/
Hi,
I've checked out the latest head from the git repository, compiled sip-router and installed it. How can i check how many users are registred at the proxy. In openser/kamailio days I used a simple 'kamctl ul show' and options to get some infos about the proxy 'internals'.
I've read some lines about a ser_ctl command but can't find it my git working copy. I've followed the instructions on http://sip-router.org/wiki/migration/kamailio-3.0-config.
OS: 2.6.18-5-686, debian 5.0.3
thx && br
Andy
--
Neu: GMX Doppel-FLAT mit Internet-Flatrate + Telefon-Flatrate
für nur 19,99 Euro/mtl.!* http://portal.gmx.net/de/go/dsl02
Hello everybody:
I start rtpproxy as the root, on the same computer, rtpproxy 1.1 works well while rtpproxy 1.2 dosen't.
>On 27.09.2009 2:56 Uhr, zhangchao00001 wrote:
>> Hello everyone, i meet a problem when intergrating rtpproxy with
>> kamailio. The two software start well, can communicate with each
>> other, however, the rtpproxy always report "can't create listener" error.
>> Dose anyone meet the same problem?
>is the user that starts the rtpproxy has rights to create sockets? Is
>there a firewall blocking higher ports?
>
>Cheers,
>Daniel
SER Users:
Thanks in advance for any help!
I am running ser 0.9.6 and have a relatively simple cfg file. I have only
four (4) digit extensions, no PSTN, no NAT and I do not care about
authorization so my config file is basically the Hello World ser.cfg with my
system parameters and a few statements in route that allowed me to create
standing conference rooms and an echo application with sems.
Now I need to implement call forwarding and I have been able to setup the
MySQL ser usr_preferences table per chapter 10 of the 'SER - Getting
Started' and the 'mySQL newbie? Problems with mySQL and SER?' documents.
After adding the call forwarding functionality discussed in chapter 10
inside my config file any number dialed to a SIP device rings busy. All SIP
devices register and can call the conference rooms or the echo application
but they can not call each other. I will spare you my WireShark logs but
can provide if that is needed.
I'm looking for trouble shooting suggestions. Is there a way to print to
the std i/o from inside the cfg file? This is the first time I needed to
handle the INVITE message so I've included my INVITE Message Handler and the
Call Forwarding Handler that I added for this effort:
...
if (method=="ACK") {
route(1);
break;
} if (method=="INVITE") {
route(3);
break;
} if (method=="REGISTER") {
route(2);
break;
};
...
route[3] {
# ----------------------------------------------------------------------
# INVITE Message Handler
# ----------------------------------------------------------------------
# Note: We are using this fuction only as a hook into the
# blind call forwarding feature. Simply want to change the
# R-URI and relay the message.
if (avp_db_load("$ruri/username", "s:callfwd")) {
setflag(22);
avp_pushto("$ruri", "s:callfwd");
# Would love to do a printf here to see if this code is being hit!!!
# Wireshark shows INVITE messages are being sent.
# debug/printf("\n\n****** Inside route(3) ********\n\n");
# At this point the a blind call forwarding record was found and the
# new destination was written in the R-URI.
# DEBUG: try just calling route(1).
# route(1);
# Send to Call Forwarding Handler
route(6);
break;
};
route(1);
}
route[6] {
# ----------------------------------------------------------------------
# Blind Call Forwarding Handler
#
# This must be done as a route block because sl_send_reply() cannot be
# called from the failure_route block
# ----------------------------------------------------------------------
lookup("aliases");
if (!is_uri_host_local()) {
if (!isflagset(22)) {
append_branch();
};
route(1);
break;
};
if (!lookup("location")) {
if (uri=~"^sip:[0-9]{4}@") {
route(1);
break;
};
sl_send_reply("404", "User Not Found");
break;
};
# DEBUG: There are no alias' and we are not sending calls to other
# networks. We may just need to call route(1) and not this
# function.
route(1);
}