Aloha,
Has there been any work on this? Jan, did you ever get the get the code?
Aloha,
Matt
>OK, could you send it to me ? I will ask Raphael or Stefan to integrate
>it with sems.
> Jan.
On 18-07 17:49, Maxim Sobolev wrote:
>> I can participate in the project by providing ITU-T floating point
>> reference implementation cleaned up by me (all global variables removed
>> and easy-to-use encode/decode API built on top of it). It should be
>> trivial to integrate it with SEMS. Its performance quite adequate -
>> allowing to support about 100 channels simulateneously on the modern
>> IA32 hardware.
hello all,
please help a poor newbie: i installed SER as a SIP proxy in front of
asterisk and it works great....followed the instructions in SER howto
to the letter, except i didn't install SRV records (only one server
for now) and i added a handoff to asterisk (asterisk is running on
port 5070 on the same machine) as a PSTN gateway.
everything works fine with x-lite, but the grandstream gets 401 errors
and won't register. Both the x-lite and the grandstream are on the
same NAT with no STUN so that is not the problem.
The config:
username and authorization user are the same
domain/realm: phone.myserver.com
proxy server: phone.myserver.com
outbound proxy: (IP address of server)
this is driving me crazy...if anyone could point me in the right
direction i would appreciate it. sorry if i'm not asking the correct
questions, i don't know what configuration on the x-lite corresponds
to what on the grandstream.
thanks,
yair
p.s. if there is a SER or asterisk or VoIP usergroup in Israel, i
would like to join one, and if not, if there are israeli users on the
list please contact me to start one, to help each other out..
Maybe you should download the source code of ser
0.8.14
from the ftp or CVS (as prefered because it has all
the last changes, don't forget use the stable version)
so you have all the control of the compilation, and
all the freedom to perform changes and probe, and when
you compile the package a ser_mysql.sh and serctl
sripts will be created.
If you nedd any extra info to compile the package fell
free to write me.
Andres Parra
_______________________________
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Shop for Back-to-School deals on Yahoo! Shopping.
http://shopping.yahoo.com/backtoschool
Thanks Jan, I Have probed the
record_route_preset("200.24.99.131")
The server answers with a 200(OK) to the INVITE
Method,
but it doesn't to the BYE method, it responds a
408(Request Timeout).
Let me ilustrate this with a couple of messages sent
by the Sip Server
and Received by the UA:
ACK Method:*****************************
SEND TIME: 1323437
SEND >> 200.24.99.131:5060
ACK sip:3306478@200.21.183.8:5064 SIP/2.0
Via: SIP/2.0/UDP
200.71.97.238:5060;rport;branch=z9hG4bK4CF97589DE6D419A8256E490D84C5B82
From: Andres Parra
<sip:3304076@ipsofactum.com>;tag=2282884754
To: <sip:3306478@ipsofactum.com>;tag=1746857641
Contact: <sip:3304076@200.71.97.238:5060>
Route:
<sip:3306478@200.24.99.131:5060;ftag=2282884754;lr=on>
*is this wright!!
Call-ID:
F6FF9150-1BC0-43B1-A864-F4071E00F9F7(a)200.71.97.238
CSeq: 19778 ACK
Max-Forwards: 70
Content-Length: 0
INVITE Method:*****************************
SEND TIME: 1313281
SEND >> 200.24.99.131:5060
INVITE sip:3306478@ipsofactum.com SIP/2.0
Via: SIP/2.0/UDP
200.71.97.238:5060;rport;branch=z9hG4bK0EF6D4BBBB9746898A187FDDD5153F3D
From: Andres Parra
<sip:3304076@ipsofactum.com>;tag=2282884754
To: <sip:3306478@ipsofactum.com>
Contact: <sip:3304076@200.71.97.238:5060>
Call-ID:
F6FF9150-1BC0-43B1-A864-F4071E00F9F7(a)200.71.97.238
CSeq: 19778 INVITE
Max-Forwards: 70
Content-Type: application/sdp
User-Agent: X-Lite release 1103a
Content-Length: 221
(SDP not Shown)
RECEIVE TIME: 1314343
RECEIVE << 200.24.99.131:5060
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP
200.71.97.238:5060;rport=5060;branch=z9hG4bK0EF6D4BBBB9746898A187FDDD5153F3D
From: Andres Parra
<sip:3304076@ipsofactum.com>;tag=2282884754
To: <sip:3306478@ipsofactum.com>;tag=1746857641
Contact: <sip:3306478@200.21.183.8:5064>
Record-Route:
<sip:3306478@200.24.99.131:5060;ftag=2282884754;lr=on>
Call-ID:
F6FF9150-1BC0-43B1-A864-F4071E00F9F7(a)200.71.97.238
CSeq: 19778 INVITE
Server: X-Lite release 1103m
Content-Length: 0
RECEIVE TIME: 1318750
RECEIVE << 200.24.99.131:5060
SIP/2.0 200 Ok
Via: SIP/2.0/UDP
200.71.97.238:5060;rport=5060;branch=z9hG4bK0EF6D4BBBB9746898A187FDDD5153F3D
From: Andres Parra
<sip:3304076@ipsofactum.com>;tag=2282884754
To: <sip:3306478@ipsofactum.com>;tag=1746857641
Contact: <sip:3306478@200.21.183.8:5064>
Record-Route:
<sip:3306478@200.24.99.131:5060;ftag=2282884754;lr=on>
Call-ID:
F6FF9150-1BC0-43B1-A864-F4071E00F9F7(a)200.71.97.238
CSeq: 19778 INVITE
Content-Type: application/sdp
Server: X-Lite release 1103m
Content-Length: 244
(SDP not Shown)
BYE Method:*****************************
SEND TIME: 1445500
SEND >> 200.24.99.131:5060
BYE sip:3306478@200.21.183.8:5064 SIP/2.0
Via: SIP/2.0/UDP
200.71.97.238:5060;rport;branch=z9hG4bKE6009A2AA4484F558657D94C3E7C7CF7
From: Andres Parra
<sip:3304076@ipsofactum.com>;tag=2282884754
To: <sip:3306478@ipsofactum.com>;tag=1746857641
Contact: <sip:3304076@200.71.97.238:5060>
Route:
<sip:3306478@200.24.99.131:5060;ftag=2282884754;lr=on>
Call-ID:
F6FF9150-1BC0-43B1-A864-F4071E00F9F7(a)200.71.97.238
CSeq: 19779 BYE
Max-Forwards: 70
User-Agent: X-Lite release 1103a
Content-Length: 0
RECEIVE TIME: 1471343
RECEIVE << 200.24.99.131:5060
SIP/2.0 408 Request Timeout
Via: SIP/2.0/UDP
200.71.97.238:5060;rport=5060;branch=z9hG4bKE6009A2AA4484F558657D94C3E7C7CF7
From: Andres Parra
<sip:3304076@ipsofactum.com>;tag=2282884754
To: <sip:3306478@ipsofactum.com>;tag=1746857641
Call-ID:
F6FF9150-1BC0-43B1-A864-F4071E00F9F7(a)200.71.97.238
CSeq: 19779 BYE
Server: Servidor SIP IPSOFACTUM ( Version 0.1
(i386/linux))
Content-Length: 0
Warning: 392 192.168.0.2:5060 "Noisy feedback tells:
pid=12978 req_src_ip=200.71.97.238 req_src_port=5060
in_uri=sip:3306478@200.21.183.8:5064
out_uri=sip:3306478@200.21.183.8:5064 via_cnt==0" *
What is this? i have always wondered
*********************
Ok, sorry about all those messages but i'm about to
finish my query:
The thing is, all the above shows me that the SIP
Proxy is
recieving the BYE but it doesn't process it because a
200(OK)
isn't replayed, why is that?
I thinking that i'd maybe make the server to listen to
the public_IP_Add
with "listen=200.24.99.131", (this could work???), or
adding
"advertised_address = 200.24.99.131" to the ser.cfg.
Maybe you could help me to decide, I'm realy confused.
Thanx
Andres Parra
--- Jan Janak <jan(a)iptel.org> wrote:
> Date: Sun, 12 Sep 2004 20:59:00 +0200
> From: Jan Janak <jan(a)iptel.org>
> To: Andr�s_Parra_L. <apl_1980b(a)yahoo.com>
> CC: serusers(a)lists.iptel.org
> Subject: Re: [Serusers] Record-Route behind NAT
>
> If the SIP server listens on the private IP address
> only then you have
> to force it to use the public IP explicitely because
> it does not know
> it.
>
> You can use record_route_preset("200.24.99.131;lr");
>
> Jan.
>
> On 11-09 11:29, Andr�s Parra L. wrote:
> > I have a little problem, i need to force the UA
> > (outside NAT with public IP address) to send the
> BYE
> > messages
> > to my Sip Server so I can account them in the acc
> > table.
> > I use record_route() to acomplish that but the
> thing
> > is the
> > Record-Route field in the message that an UA
> recieve,
> > the IP address is
> > the SIP SERVER LOCAL IP ADDRESS BEHIND NAT
> > (Record-Route:
> > <sip:5000022@192.168.0.2;ftag=192550680;lr=on>)
> > which means, of course, that the message will not
> pass
> > trough the SIP Server.
> > How Could i force the server to write in the
> > Record-Route field the valid
> > IP address of my Sip Server (200.24.99.131)????
> >
> > Example message sent by the Sip Proxy:
> > RECEIVE TIME: 1420109
> > RECEIVE << 200.24.99.131:5060
> > SIP/2.0 180 Ringing
> > Via: SIP/2.0/UDP
> >
>
200.71.103.253:5060;rport=5060;branch=z9hG4bK725C92055C654839934F043974F39E1F
> > From: Andres Parra
> > <sip:3304076@ipsofactum.com>;tag=192550680
> > To: <sip:5000022@ipsofactum.com>;tag=2391513217
> > Contact: <sip:5000022@68.38.237.35:32805>
> > Record-Route:
> > <sip:5000022@192.168.0.2;ftag=192550680;lr=on>
> > Call-ID:
> >
> 31517648-DF5F-4A12-BE74-5B1026B4C39D(a)200.71.103.253
> > CSeq: 29855 INVITE
> > Server: X-Lite release 1103m
> > Content-Length: 0
> >
> > Ser.cfg:
> >
> > #
> > # $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15
> andrei
> > Exp $
> > #
> > # simple quick-start config script
> > #
> >
> > # ----------- global configuration parameters
> > ------------------------
> >
> > #debug=3 # debug level (cmd line:
> -dddddddddd)
> > #fork=yes
> > #log_stderror=no # (cmd line: -E)
> >
> > /* Uncomment these lines to enter debugging mode
> > debug=7
> > fork=no
> > log_stderror=yes
> > */
> >
> > check_via=no # (cmd. line: -v)
> > dns=no # (cmd. line: -r)
> > rev_dns=no # (cmd. line: -R)
> > #port=5060
> > #children=4
> > fifo="/tmp/ser_fifo"
> >
> >
> > # ------------------ module loading
> > ----------------------------------
> >
> > # Uncomment this if you want to use SQL database
> > loadmodule "/usr/local/lib/ser/modules/mysql.so"
> >
> > loadmodule "/usr/local/lib/ser/modules/sl.so"
> > loadmodule "/usr/local/lib/ser/modules/tm.so"
> > loadmodule "/usr/local/lib/ser/modules/rr.so"
> > loadmodule "/usr/local/lib/ser/modules/maxfwd.so"
> > loadmodule "/usr/local/lib/ser/modules/usrloc.so"
> > loadmodule
> "/usr/local/lib/ser/modules/registrar.so"
> >
> > # Uncomment this if you want digest authentication
> > # mysql.so must be loaded !
> > loadmodule "/usr/local/lib/ser/modules/auth.so"
> > loadmodule "/usr/local/lib/ser/modules/auth_db.so"
> > loadmodule "/usr/local/lib/ser/modules/acc.so"
> > loadmodule "/usr/local/lib/ser/modules/textops.so"
> >
> > # Uncomment this if you want to use SQL database
> >
> >
> > # ----------------- setting module-specific
> parameters
> > ---------------
> >
> > # -- usrloc params --
> >
> > #modparam("usrloc", "db_mode", 0)
> >
> > # Uncomment this if you want to use SQL database
> > # for persistent storage and comment the previous
> line
> > modparam("usrloc", "db_mode", 2)
> >
> > # -- auth params --
> > # Uncomment if you are using auth module
> > #
> > modparam("auth_db", "calculate_ha1", yes)
> > #
> > # If you set "calculate_ha1" parameter to yes
> (which
> > true in this config),
> > # uncomment also the following parameter)
> > #
> > modparam("auth_db", "password_column", "password")
> >
> > # -- rr params --
> > # add value to ;lr param to make some broken UAs
> happy
> > modparam("rr", "enable_full_lr", 1)
> >
> > # -- acc params -
> > modparam("acc", "log_missed_flag", 3)
> > modparam("acc", "log_level", 1)
> > modparam("acc", "log_flag", 1)
> > modparam("acc", "db_flag", 1)
> > modparam("acc", "db_missed_flag", 3)
> >
> >
> >
> >
> >
> > # ------------------------- request routing logic
> > -------------------
> >
> > # main routing logic
> >
> > route{
> >
> > # initial sanity checks -- messages with
> > # max_forwards==0, or excessively long requests
> > if (!mf_process_maxfwd_header("10")) {
> > sl_send_reply("483","Too Many Hops");
> > break;
> > };
> > if ( msg:len > max_len ) {
> > sl_send_reply("513", "Mensaje demasiado
> grande");
> > break;
> > };
> > # prevents private ip space from being used
> > if (search("^(Contact|m):
> > .*(a)(192\.168\.|10\.|172\.16)")) {
> > if (method=="REGISTER") {
> > log(1, "LOG: Someone trying to register
> from
> > private IP\n");
> > sl_send_reply("479", "Por favor no utilice
> > direcciones IP privadas" );
> > break;
> > };
> > };
> > # loose-route processing
> > if (loose_route()) {
> > t_relay();
> > break;
> > };
> > # labeled all transaction for accounting
> > setflag(1);
> >
> > # we record-route all messages -- to make sure
> that
> > # subsequent messages will go through our proxy;
> > that's
> > # particularly good if upstream and downstream
> > entities
> > # use different transport protocol
> >
> > # record-route INVITES to make sure BYEs will
> visit
> > our server too
> > if (method=="INVITE") record_route();
> >
> >
> >
> > # if the request is for other domain use UsrLoc
> > # (in case, it does not work, use the following
> > command
> > # with proper names and addresses in it)
> > if (uri=~"ipsofactum.com" ){#||
> > !(uri=~"^sip:(192\.168\.|10\.|172\.16)")) {
> >
> > if (method=="REGISTER") {
> >
> > # Uncomment this if you want to use digest
> > authentication
> > if (!www_authorize("ipsofactum.com",
> "subscriber"))
> > {
> > www_challenge("ipsofactum.com", "0");
> > break;
> > };
> > setflag(3);
> > save("location");
> > break;
> > };
> >
> >
> >
> > # native SIP destinations are handled using our
> > USRLOC DB
> > if (!lookup("location")){ #&&
> > !lookup("subscribers")) {
> > # call invitations to off-line users are
> reported
> > using the
> > # acc_request action; to avoid duplicate
> > reports on request
> > # retransmissions, request is processed
> > statefuly (t_newtran,
> > # t_reply)
> > if ((method=="INVITE" || method=="ACK") &&
> > t_newtran() ) {
> > t_reply("404", "Usuario no
> registrado!,
> > contacte el directorio de usuarios registrados");
> > acc_db_request("404 Not
> > Found","missed_calls");
> > break;
> > };
> > # all other requests to off-line users are
> > simply replied
> > # statelesslyeth0 and no reports are
> issued
> > #sl_send_reply("404", "Usuario no
> existente!,
> > contacte el directorio de usuarios suscritos");
> > #break;
> > } else {
> >
> >
> >
> > # user on-line; report on failed
> transactions;
> > mark the
> > # transaction for reporting using the same
> > number as
> > # configured above; if the call is really
> > missed, a report
> > # will be issued
> >
> > setflag(3);
> > # forward to user's current destination
> > t_relay();
> > break;
> > };
> > };
> >
> >
> >
> > # forward to current uri now; use stateful
> > forwarding; that
> > # works reliably even if we forward from TCP to
> UDP
> > if (!t_relay()) {
> > sl_reply_error();
> > };
> >
> > }
> >
> >
> >
> >
> > _______________________________
> > Do you Yahoo!?
> > Shop for Back-to-School deals on Yahoo! Shopping.
> > http://shopping.yahoo.com/backtoschool
> >
> > _______________________________________________
> > Serusers mailing list
> > serusers(a)lists.iptel.org
> > http://lists.iptel.org/mailman/listinfo/serusers
>
__________________________________
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Does anyone have a good SEMS how-to? I've successfully loaded SER in Fedora
C2, which works great. But SEMS, I can't even download it, much less,
install it.
Any help would be appreciated :-)
___________________________________________
Ricardo Alvarado
e ricardo(a)iq-networking.com
___________________________________________
Hi
does anyone knows if iptel SIP stack and SER are running on Montavista
linux?
If yes, how much of an effort was it.
Any info, pointers would be greatly appreciated.
Thank you
Dean
Hi, I need configuration the SJPhone program to iptel services, I create
account in iptel.org buy don't understand make configuration in the program ?
That other programs work with the service of iptel?
Santiago Hoyos Restrepo
Calipso Comunicaciones S.A.
www.calipso.com.co
Dear maintainers:
When I start ser, I got an error message below:
ERROR: load_module: could not open module
</usr/lib/ser/modules/mysql.so>: /usr/lib/libmysqlclient.so.10: symbol
errno, version GLIBC_2.0 not defined in file libc.so.6 with link time
reference?
How can I solve the problem?
Thanks in advance. ^^
--
Chi-Yen Yeh
Providence University
Computer Science and Information Management
Did anybody ever trying to get SER running on the open source Linksys
router? I have a WRT54G router loaded with a firmware that supports SSL
forked from the Linksys code. Look to me that we may be able to squeeze in
some SER functionality on that Linux. I agree that not all modules will be
fitting.
For more info on the GPL linksys: http://www.linksys.com/support/gpl.asp
Also did anybody ever dreamed on adding other VOIP protocol support to SER?
(MGCP, H323, etc...)
MarcG.