The new ser stable version, ser 0.8.14 has just been released.
This version is based on the latest rel_0_8_12 branch code and not on
the cvs head. We decided to name it 0.8.14 to avoid confusion (0.8.13
was already taken by the unstable cvs branch).
The new release can be downloaded from:
ftp://ftp.berlios.de/pub/ser/0.8.14 (source code, pre-compiled binaries
for various arhitectures, packages for various operating systems and
linux distributions; for more informations see
ftp://ftp.berlios.de/pub/ser/0.8.14/README ).
If you use cvs, just add -rrel_0_8_14 to your cvs checkout command.
Please read the attached NEWS file for more information.
Andrei
Gents,
I've been going throught the draft-ietf-iptel-cpl-09.txt to try and find information on doing the following:
I want to offer my clients the option to do call forwarding and other smart things with cpl.
Now I can not find information on how I can change the source name of the caller.
Just to explain:
- I am a user and out of the office
- I set a call forward to my mobile
- some calls me and my cpl script is run
- it tells the proxy to route the call to my mobile
Who will now be billed for the call ?
I can not find a way to change the calling-id.
In this case I need to be billed for the outgoing call to the PSTN (mobile).
Am I correct to state this is not possible with CPL ?
regards,
Arne.
Hello,
My 'ser' and 'mysql' are running without any problem !.
Last situation of the "ser.cfg" is:
if (uri=~"mustafa.avrupayazilim.com") {
if (method=="REGISTER") {
if (!www_authorize("mustafa.avrupayazilim.com", "subscriber")) {
www_challenge("mustafa.avrupayazilim.com", "0");
break;
};
My proble is:
I tried to send an email with "serctl mail " command.
I have an user name: "murat", at the database
and with e-mail addres: murat_memet(a)yahoo.com.
I finished the letter with "Ctrl+c ", and see a messege " message send.
Bu t I didnt' see any maseege my malil box at Yahoo.
Dou you have any suggesions?.
One more question:
I can not send any mail to . murat(a)iptel.org.....
Thanks for your help
Hi,
I'm trying to set up a capi gw with sems and ser.
How to write the routing statement from sip phone to pstn if ser and sems
are on the same host ?
Do they communicate via ser_fifo and am_fifo?
thanks,
david
Hi,
when I try to configure ser for the administration of two different domains (with two instancesin two different
ports), I obtain the following error:
connect_db():Too many connections.
db_init():Error while trying ot connect database
then signal 15 is received
Could you tell me if that would be a problem of the mysql configuration, and if true, how can I solve it?
Regards,
Blanca
______________________________________________________
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I have an accounting problem in this simple escenario:
PSTNCaller --> Cisco AS5350 --> SER --> IPPhone
I can see complete radius Accounting log (Start and Stop generated by ser) when the callee IPPhone hung up first, but when the PSTNCaller hung up fisrt just the "start" is generated in radius and the "stop" log is missing, in this case when I watch messages using ngrep in port 5060 I´ve realized that GW sends BYE twice to ser and IPPhone also sends OK twice to Ser...
any idea?
regards
Rafael
PS: relevant as5350 config:
dial-peer voice 2001 voip
description PSTN-->SIPSER
destination-pattern 6604000
session protocol sipv2
session target sip-server
codec g723r63
!
!
sip-ua
sip-server ipv4:200.110.X.Y
!
---------
ser.cfg:
# $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $
/* Uncomment these lines to enter debugging mode
debug=20
fork=yes
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 ----------------------------------
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"
loadmodule "/usr/local/lib/ser/modules/auth.so"
loadmodule "/usr/local/lib/ser/modules/auth_radius.so"
loadmodule "/usr/local/lib/ser/modules/group_radius.so"
loadmodule "/usr/local/lib/ser/modules/acc.so"
# ----------------- setting module-specific parameters ---------------
modparam("rr", "enable_full_lr", 1)
modparam("auth_radius","radius_config","/usr/local/etc/radiusclient/radiusclient.conf")
modparam("auth_radius","service_type",15)
modparam("acc","radius_config","/usr/local/etc/radiusclient/radiusclient.conf")
modparam("acc", "service_type", 15)
modparam("acc", "radius_flag", 1)
modparam("acc", "radius_missed_flag", 3)
modparam("acc", "failed_transactions", 1)
modparam("acc", "report_ack", 0)
# ------------------------- 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", "Message too big");
break;
};
# 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();
# loose-route processing
if (loose_route()) {
t_relay();
break;
};
# 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 (method=="INVITE") {
log(1, "INVITE\n");
setflag(1); /* set for accounting (the same value as in log_flag!) */
};
if (method=="BYE" || method=="CANCEL") {
log (1, "BYE or CANCEL\n");
setflag(1);
};
if (uri==myself) {
if (method=="REGISTER") {
log(1, "REGISTER: Authenticating user\n");
if (!radius_www_authorize("")) {
log(1, "REGISTER: challenging user\n");
www_challenge("", "0");
break;
};
save("location");
break;
};
# native SIP destinations are handled using our USRLOC DB
if (!lookup("location")) {
sl_send_reply("404", "Not Found");
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();
break;
};
}
----------------------
---------------------------------
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Hi,
I am trying to prototype a ser module by first using a perl script in
exec_dset, and I would like to record the calls conference ID. I had
thought that the TID was the conferenceid (As seen on cisco vsa
accounting) but they do not match.
Here is an example:
I get an INVITE request from a Cisco ATA using ser as an outbound
proxy.
The exec_dset script sees this in the environment:
SIP_TID = 6d2796abca7d6e1d65249aa229bc2f53
The cisco 5350 I redirect to shows this in its cisco-vsa accounting:
h323-conf-id = "h323-conf-id=8457403F E38341D8 872CFC52 183D0ADC"
What is happening here? Is the cisco gateway generating the conferencid
for the SIP call? Should it be using the SIP TID?
--
David Stanaway <david(a)dialmex.net>
I had install ser+voicemail(sems), but I don't know how to edit my ser.cfg
file.
could someone post the ser.cfg file.And tell me how to know voicemail is
work?How to test by myself ua?
Is any ua support to use voicemail by sems?
help me!! thanks.
