Excuse me I did not thank people who called me for my
tests.
Thanks Olivier Taylor and others, Snom.com and
Ingate.com.
Send me mails if you need I call you to test your
configuration.
Regards
harry
___________________________________________________________________________
Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger
Téléchargez cette version sur http://fr.messenger.yahoo.com
Hello all,
How can we configure serweb+sems to work together for
voicemail ?
voicemail plugin search here for greetings here:
# optional parameter: announce_path=<path>
#
# - sets the path where announce files are searched
for
# - the file to be played is searched in the following
order:
# <announce_path>/<domainname>/<username>.wav
#
<announce_path>/<domainname>/<language>/<default_announce>
# <announce_path>/<domainname>/<default_announce>
# <announce_path>/<language>/<default_announce>
# <announce_path>/<default_announce>
# where <language> is taken from the body of
P-Language header
# of the request (if any).
announce_path=/usr/local/lib/sems/audio/
voicemail Tab in serweb allow to upload or download
greetings
How can we record greetings via sems ivr plugin
(record.py) with a phone which put this .wav file in a
shared directory (with NFS) for sems and serweb ?
Regards
Harry
___________________________________________________________________________
Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger
Téléchargez cette version sur http://fr.messenger.yahoo.com
Hello,
I'm currently working on a VOIP network where the phones register to an
OpenSER server - we also have a PSTN Gateway (Asterisk).
To be able to account the PSTN calls of a user we insert a custom header
"X-Account" which stores information about the user account. We can't
just use the From-Header for this as we allow users to e.g. create
unconditional forwards which forward the call to a defined SIP uri when
the user isn't available. If somebody creates a forward to the PSTN it
is still his account which should be accounted and which is therefor put
in the X-Account header.
This part works fine, but I'm still not sure how to handle call
transfers - e.g. when a user wants to transfer his conversational
partner to the PSTN. An unattended transfer in SIP is implemented with a
REFER request which causes the remote phone to send an INVITE to the SIP
uri in question[1]. I'm not sure how to account these transfers. On a
traditional PSTN network the person who caused the transfer would be the
one who is paying for the call. But I see no solution how to implement
this behaviour in SIP (because the old call isn't transfered, but the
conversational partner starts a new call with the INVITE request).
Is it somehow possible to detect if a call was caused by a transfer so
that I can insert the right X-Account header? Do I need a B2BUA like
Asterisk to implement the traditional behaviour? How do other VOIP
providers solve this problem (IMO call transfers shouldn't be such an
uncommon feature).
tia for all suggestions on how to solve this problem
/gst
[1]
http://www.ietf.org/internet-drafts/draft-ietf-sipping-service-examples-08.…
i'm trying to load the db and check if the username for the "to" sip uri is in
the database, but my script is displaying the following errors:
0(0) AVPops - initializing
0(0) ERROR:avpops:parse_check_value: unknown flag <u>
0(0) ERROR:avpops:fixup_check_avp: failed to parse checked value
0(0) ERROR: fix_expr : fix_actions error
ERROR: error -1 while trying to fix configuration
Here's my ser.cfg:
less /etc/ha.d/ser.cfg
/etc/ha.d/ser.cfg: No such file or directory
[root@SER02 ser]# less /usr/local/etc/ser/ser.cfg
#
# $Id: ser.cfg,v 1.27 2005/03/10 14:16:25 andrei Exp $
#
# simple quick-start config script
#
# ----------- global configuration parameters ------------------------
fork=yes
# Uncomment these lines to enter debugging mode
#debug=3 # debug level (cmd line: -dddddddddd)
#fork=no
log_stderror=yes # (cmd line: -E)
#memlog=5 # memory debug log level
#log_facility=LOG_LOCAL0 # sets the facility used for logging (see syslog(3))
alias="SER02"
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"
fifo_db_url="mysql://ser:heslo@10.1.201.107/ser"
#user=ser
#group=ser
#fifo_user=ser # owner of the ser fifo
#fifo_group=ser
#fifo_mode=0660 # fifo's permissions
#disable_core=yes #disables core dumping
#open_fd_limit=1024 # sets the open file descriptors limit
#mhomed=yes # usefull for multihomed hosts, small performance penalty
#disable_tcp=yes
#tcp_accept_aliases=yes # accepts the tcp alias via option (see NEWS)
#
# ------------------ 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/textops.so"
loadmodule "/usr/local/lib/ser/modules/xlog.so"
loadmodule "/usr/local/lib/ser/modules/exec.so"
loadmodule "/usr/local/lib/ser/modules/avpops.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", 1)
modparam("usrloc", "timer_interval", 10)
# -- db_url params --
modparam("acc|auth_db|domain|group|permissions|speeddial|uri_db|usrloc|xdz_tools",
"db_url","mysql://ser:heslo@10.1.201.107/ser")
# -- auth params --
modparam("auth_db", "calculate_ha1", yes)
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)
# -- tm params --
modparam("tm", "fr_inv_timer", 90)
# -- avp params --
modparam("avpops", "avp_url", "mysql://ser:heslo@10.1.201.107/ser")
modparam("avpops", "avp_table", "location")
#modparam("avpops", "use_domain", "1")
#modparam("avpops", "uuid_column", "callid")
modparam("avpops", "username_column", "username")
modparam("avpops", "domain_column", "domain")
#modparam("avpops", "attribute_column", "attribute")
modparam("avpops", "value_column", "contact")
#modparam("avpops", "type_column", "type")
#modparam("avpops", "avp_aliases",
"voicemail=i:500;calltype=i:700;fwd_no_answer_type=i:701;fwd_busy_type=i:702")
# To use more than one tables example
#modparam("avpops", "db_scheme",
"scheme1:table=location;username_col=username;domain_col=domain;value_col=contact;value_type=string
")
# ------------------------- 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;
};
# 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==myself)
{
xlog("L_INFO", "%rm New URI = %ru from %ct\n\n");
if (method=="REGISTER")
{
# digest authentication
if (!www_authorize("localhost", "subscriber")) {
www_challenge("localhost", "0");
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
xlog("L_INFO", "pre-rr(): %rm to URI %ru\n\n");
# record_route();
record_route_preset( "10.1.201.189:5060" );
xlog("L_INFO", "post-rr(): %rm to URI %ru\n\n");
# subsequent messages withing a dialog should take the
# path determined by record-routing
if ( loose_route() )
{
# mark routing logic in request
append_hf("P-hint: rr-enforced\r\n");
xlog("L_INFO", "loose_route(): Looking up %rm URI %ru from
%ct\n\n");
lookup("location");
xlog("L_INFO", "loose_route(): t_relay() %rm to URI %ru\n\n");
route(1);
break;
};
if ( uri =~
"sip:[0-9]+@(10.1.101.(159|250)|192.168.10.31|64.77.239.90)" )
{
if ( method == "INVITE" || method == "ACK" || method ==
"CANCEL" || method == "REFER" || method == "BYE" )
{
xlog("L_INFO", "\n%rm came from a VoIP phone
(%ct)\n\n");
# Making an outbound cal, which starts with a "9", to a
PSTN phone
if( uri =~ "^sip:9[0-9]+@*" )
{
append_hf( "P-hint: OUTBOUND\r\n" );
route(2);
rewritehostport("10.1.101.152:5060");
# forward( localhost, 5061 );
xlog("L_INFO", "URI is a PSTN phone...sending
URI (%ru) to 10.1.101.152\n\n");
# xlog("L_INFO", "URI is a PSTN phone...sending
URI (%ru) to Asterisk\n\n");
route(1);
break;
}
# Making an outbound call to another VoIP phone
else
{
# Redirect call to voicemail when destination
is not registered
append_hf( "P-hint: USRLOC\r\n" );
if( !lookup("location") && !(uri =~
"^sip:11111111111@*") )
{
# prefix("*");
xlog("L_INFO", "\n\n%ru is not online
(in usrloc)\n");
# route(3);
route(4);
}
else
{
setflag(5);
};
# forward( uri:host, uri:port );
# forward( localhost, 5061 );
if ( isflagset(5) )
{
xlog("L_INFO", "URI is a VoIP
phone...forwarding URI (%ru) to destination\n\n");
# xlog("L_INFO", "URI is a VoIP
phone...forwarding URI (%ru) to Asterisk\n\n");
resetflag(5);
route(1);
};
break;
};
};
route(1);
break;
};
# route(1);
}
#---------------------------------------------------------------------
route[1]
{
# send it out now; use stateful forwarding as it works reliably
# even for UDP2TCP
xlog("L_INFO", "\n\n%ru is being t_relay()ed\n");
if (!t_relay()) {
xlog("L_INFO", "\n\n%ru t_relay() error\n");
sl_reply_error();
break;
};
}
#---------------------------------------------------------------------
# All PSTN calls require a prefix of "0010"
route[2]
{
if ( uri =~ "^sip:0010[0-9]+@*" )
{
xlog("L_INFO", "URI has 0010 prefix already\n\n");
}
else
{
xlog("L_INFO", "URI does not has 0010 prefix...attaching
prefix\n\n");
strip(1);
prefix("0010");
};
}
#---------------------------------------------------------------------
# Check database for actual user address
route[4]
{
if ( avp_db_load("$to/username","s:") &&
avp_check("s:username","eq/$to/username/g") )
{
xlog("L_INFO", "\n\n%ru is online (in db)\n");
setflag(5);
break;
};
xlog("L_INFO", "\n\n%ru is not online (in db)\n");
sl_send_reply( "404", "User does not exist" );
}
Can someone tell me what i'm doing wrong? Thanks.
Jack
____________________________________________________
Sell on Yahoo! Auctions no fees. Bid on great items.
http://auctions.yahoo.com/
Hi,
please (CC) the mailing list in your replies.
Try enable logging in serweb. Set
$config->enable_loging = true;
$config->log_level = "PEAR_LOG_DEBUG";
in config file and see the log file.
If it will not help you, send the logfile and content of your
subscriber table to me.
Karel
han yibing napsal(a):
> Hi
>
> I installed SERWEB 0.9.3 version
> and i login mysql to check subscriber and
> admin_privileges tables ,all status is ok.
>
>
>
> --- Karel Kozlik <karel(a)iptel.org>写道:
>
>
>>Hello,
>>which version of serweb are you useing?
>>
>>Please make sure your username, password and domain
>>values are exactly
>>matched the values of your subscriber table.
>>
>>regards Karel
>>
>>han yibing napsal(a):
>>
>>> "bad username or password" display when i have
>>>default username and password (admin/heslo) login
>>>SERWEB,my domain is local IP address.
>>>
>>>would you link to tell why and how to do it ?
>>>
>>>thanks.
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>
> ___________________________________________________________
>
>>>雅虎免费G邮箱－中国第一绝无垃圾邮件骚扰超大邮箱
>>>http://cn.mail.yahoo.com/?id=77071
>>>
>>>_______________________________________________
>>>Serusers mailing list
>>>serusers(a)lists.iptel.org
>>>http://lists.iptel.org/mailman/listinfo/serusers
>>
>
>
>
>
>
>
>
> ___________________________________________________________
> 雅虎免费G邮箱－中国第一绝无垃圾邮件骚扰超大邮箱
> http://cn.mail.yahoo.com/?id=77071
>
I have searched high & low but can't seem to find any info on how to
use fifo_server.php for fifo relay. I have setup serweb (0.9.3) on a
different box than ser, and it works fine except for the fifo stuff.
Can anyone give me a clue on how to configure this properly.
Thanks!
- Daryl
As I anm struggling to build a configuration with OpenSER which
operates a multi-domain proxy, sends calls to SEMS for voicemail after
a timeout and uses a third-party SIP gateway for PSTN in/out; I wonder
if someone has a configuration for this somewhere already. It does
seem like a rather "normal" config...
Does anyone have such a configuration? It would help me tremendously!
Thank you,
A.
--
Adam Sherman
Technologist
I'm attempting to configure OpenSER to feed SEMS the correct
information for voicemail, namely the email address.
SEMS documentation referes to the avp.so module, which I understand is
no longer relevant.
If all I need to do is pull the email address out of the subscribers
table and feed it to SEMS, what openser config do I need?
Thanks,
A.
--
Adam Sherman
Technologist
