Hello list,
I'm new to this place, and actually new to the world of SIP and VoIP,
but I'm starting to like it after I've seen what it can do.
I tried to install ser myself on my machine running NetBSD 2.0 on the
amd64 architecture.
Starting from scratch, I tried to install the current releases, but they
fail mostly on modules. So I deciced to give CVS a try.
After compiling, I ended up with a sip_router dir, and it compiled ok
from what I could see. It had a few warnings but I figured that it
couldn't hurt much.
The main configfile didn't leave much to change, so I set debug on and
fork off, and started off:
bash-2.05b# ./ser
Listening on
udp: 10.0.0.30 [10.0.0.30]:5060
udp: 130.89.169.133 [130.89.169.133]:5060
udp: 127.0.0.1 [127.0.0.1]:5060
tcp: 10.0.0.30 [10.0.0.30]:5060
tcp: 130.89.169.133 [130.89.169.133]:5060
tcp: 127.0.0.1 [127.0.0.1]:5060
Aliases:
tcp: localhost.:5060
tcp: localhost:5060
tcp: ascwin.student.utwente.nl:5060
tcp: mobiel30.crib:5060
udp: localhost.:5060
udp: localhost:5060
udp: ascwin.student.utwente.nl:5060
udp: mobiel30.crib:5060
WARNING: no fork mode and more than one listen address found(will use
only the the first one)
stateless - initializing
textops - initializing
ERROR: error -478 while trying to fix configuration
In the /var/log/messages, it shows:
Dec 10 16:16:36 ascent ser: ERROR: lock_set_init (SYSV): semget (...,
55, 0700) failed: No space left on device
Dec 10 16:16:36 ascent ser: ERROR: lock_set_init (SYSV): semget (...,
54, 0700) failed: No space left on device
Dec 10 16:16:36 ascent ser: ERROR: lock_set_init (SYSV): semget (...,
53, 0700) failed: No space left on device
Dec 10 16:16:36 ascent ser: ERROR: lock_set_init (SYSV): semget (...,
52, 0700) failed: No space left on device
Dec 10 16:16:36 ascent ser: ERROR: lock_set_init (SYSV): semget (...,
51, 0700) failed: No space left on device
Dec 10 16:16:36 ascent ser: ERROR: lock_set_init (SYSV): semget (...,
50, 0700) failed: No space left on device
Dec 10 16:16:36 ascent ser: ERROR: lock_set_init (SYSV): semget (...,
49, 0700) failed: No space left on device
Dec 10 16:16:36 ascent ser: ERROR: lock_set_init (SYSV): semget (...,
48, 0700) failed: No space left on device
Dec 10 16:16:36 ascent ser: ERROR: lock_set_init (SYSV): semget (...,
47, 0700) failed: No space left on device
This is the case running as both root and as user.
Any tips/directions? I googled around for the error and couldn't find
anything matching.
Thanks in advance!
Wouter Schoot
hello friends,
as in the mailing list archive about the sems with ivr
i followed the instrucions as mentioned by some
1) in ser.cfg iam loading the module tm.so
2) in sems.conf iam usieng ivr path
3) in ser.cfg iam useing like
if(!vm("/tmp/am_fifo","ivr"))
and my ser.cfg
# ----------- global configuration parameters
------------------------
debug=8 # debug level (cmd line: -dddddddddd)
fork=yes
log_stderror=yes # (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"
loadmodule "/usr/local/lib/ser/modules/vm.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"
# ----------------- 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")
modparam("usrloc","db_url","mysql://ser:heslo@localhost:3306/ser")
# -- rr params --
# add value to ;lr param to make some broken UAs happy
modparam("rr", "enable_full_lr", 1)
#voicemail
modparam("voicemail","db_url","mysql://ser:heslo@localhost:3306/ser")
modparam("voicemail","subscriber_table","subscriber")
modparam("voicemail","email_column","email_address")
# ------------------------- 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;
};
record_route();
if (loose_route()) {
t_relay();
break;
};
if (uri==myself) {
if (method=="REGISTER") {
save("location");
break;
};
if ( method=="ACK" || method=="INVITE" ||
method=="BYE" || method=="CANCEL" )
{
if (!t_newtran())
{
sl_send_reply("500","could not create transaction");
break;
};
t_reply("100","Trying - just wait
a minute !");
if (method=="INVITE")
{
log("in to the invite");
if(!vm("/tmp/am_fifo","ivr"))
{
log("could not contact announcement server");
t_reply("500","error contacting sems");
};
break;
}
else if (method=="BYE" ||
method=="CANCEL")
{
if(!vm("/tmp/am_fifo","bye"))
{
t_reply("500","error
contacting sems");
};
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();
};
}
and my sems.conf
[root@server1 sems]# cat sems.conf
##################################
# global parameters #
##################################
fork=yes
stderr=no
loglevel=1
fifo_name=/tmp/am_fifo
ser_fifo_name=/tmp/ser_fifo
plugin_path=/usr/local/lib/sems/plug-in
smtp_server=localhost
smtp_port=25
##################################
# module specific parameters #
##################################
config.voicemail=inline
rec_file_extension=wav
announce_path=/usr/local/lib/sems/audio/
default_announce=default_en.wav
max_record_time=30
accept_delay=0
email_template_path=/usr/local/etc/sems/
config.voicemail=end
config.announcement=inline
announce_path=/usr/local/lib/sems/audio/
default_announce=default_en.wav
config.announcement=end
config.ivr=inline
ivr_script_path=/root/answer_machine/plugin/ivr/script
ivr_script_file=ivr.py
tts_caching=y
tts_cache_path=/tmp/wavs
config.ivr=end
config.conference=inline
default_announce=/usr/local/lib/sems/audio/first_participant.wav
config.conference=end
config.number_reader=inline
number_path=/usr/local/lib/sems/audio/
prolog_file=welcome_to_number_reader.wav
epilog_file=thanks_calling_number_reader.wav
config.number_reader=end
and the output iam getting are as followed
(1254) ERROR: reply (AmRequest.cpp:335):
AmRequestUAS::reply: 500 command 't_reply' not
available
my aim is to run the ivr.py and to get the
dtmf recorded in to a variable .
when i run the command serctl fifo which
it doesnot shows t_relay porcess in that
so any hints to go from here
your help will be highly appreciated
with regards
v r kanth
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Hi
Im trying to make SIP to H.323 calls, redirecting from SER to Asterisk and then to GNUGK
I tried hard to do it but the documentation is not very explicit.
As SIP Im using a ZyXel Prestige 2000P and X lite, and as H.323 im using an Aethra.
What do I need in the SER configuration (ser.cfg) to redirect the H.323 calls to asterisk?
I tried the rewritehostport:
rewritehostport("Asterisk_ip : Asterisk_port");
t_relay();
.... but Asterisk didnt received the calls.
Can anyone tell me how to redirect H.323 calls to Asterisk?
Thanks in advance
Joao Pereira
Hi:
I use the following rule to add "user=phone" parameter in the
"INVITE" request;
if(method == "INVITE")
{
subst_uri('/^sip:([0-9]+)@(.*)$/sip:\1@\2;user=phone/i');
};
But I can't work! how should I do in order to make it?
Thanks for your instructions and help!
Best Regards
Sun Zongjun
Hi:
I encounter one problem. When some ua want to call PSTN phone, I want to
place "user=phone" into the REQUEST uri and From ,To header. How can
get that??
Thanks for your instructions
Best Regards.
Sun Zongjun
Hi,
I am using ser in a prepaid call system. I need ser to drop a call when
the balance is used up. But I don't know how ser can drop a
call(disconnect a call). Can some one help me?
Thanks in advance.
Michael
Michael(a)polylink.net
Hello All,
I recenlty installed ser stable version (0.8.14) and I use it with a
couple of Cisco IP Phones and a Cisco 36xx voice Gateway. I installed it
using mediaproxy because some phones are behind a nat and it works fine
for "basic calls" (IP-IP, IP-PSTN and PSTN-IP).
However I have a problem when I try to transfer (using attended
transfer) a call coming from PSTN to a phone that is behind nat. The
problem is the following:
The phone originating the transfer sends correctly the REFER method to
the Voice gateway through SER server containing the refer-to:
number@publicIPaddress but the Gateway sends the INVITE method "directly
to the publicIPaddress of the NAT", and doesn't pass through the SIP
server (that "manages" the messages for the NATed phones) even if I
configured a dial peer voip matching the voip strings and specifying the
server where I have to send the INVITE requests.
My question is: is there any way to configure SER changing the REFER
message in order to "force" the gateway to send the INVITEs through the
server?
Or, anyone knows if something more than dial-peers has to be configured
on Cisco gateways?
Thanks a lot in advance,
Raffaele.
