Hi there,
I'm sorry for bothering all this time with same questions
I still can't solve the problem because nobody reply it.
I really don't have any idea about how SEMS works.
I setup eyebeam to forward call to voicemail number which is 115@domain
for more info, the voicemail address was one of the user in subscriber/aliases table of SER that I added.
When the call forwarded, SER have to check to usrloc for the address. It automatically appear that User not found on the softphone.
Will it be any way that I don't have to create the address for voicemail address but SER can forward the call for SEMS?
Or maybe my softphone forward call setting isn't right?Btw, I'm using eyebeam.
I hope that someone reply for my mail this time.
I'm sorry if my english is bad.
Regards,
Meidiana
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Hi,
thanks to Ancuta Onofrei, a new member of Voice System's team, NATHELPER
module now supports multiple rtpproxies - actually you can define
multiple sets of rtpproxies - nathelper performs load-balancing over a
set and the user can select the set.
For more details, see:
http://www.openser.org/docs/modules/devel/nathelper.html#AEN76
The definition of the rtpproxy socket was extended to allow definition
of multiple sets. See:
http://www.openser.org/docs/modules/devel/nathelper.html#AEN140
Also, a new function was added to set from script what set of rtpproxies
is to be used. See:
http://www.openser.org/docs/modules/devel/nathelper.html#AEN298
Example:
# multiple sets of multiple rtproxies
modparam("nathelper", "rtpproxy_sock", "1 == udp:localhost:12221 udp:localhost:12222")
modparam("nathelper", "rtpproxy_sock", "2 == udp:localhost:12225")
.....
set_rtp_proxy_set("2");
force_rtp_proxy();
.....
regards,
bogdan
Hi,
what happens to the t_lookup_request function of the tm module in the
1.2.0 release? I was using this function in my openser script and no it
disappears. Although the underlying t_check function still exists.
What is the reason?
Thanks,
Wolfgang
Dear friends:
I have installed openser-1.1.0-tls version, and it works fine with mysql, the sip flow is below:
UA OpenSER
invite---------------->
<----------------100 trying
<----------------101 dialog establishement
<----------------180 ringing
.....
My question is, why not have the 407 Proxy Authentication Required in the per call, see my openser.cfg file:
#
# $Id: openser.cfg,v 1.6 2006/02/15 18:23:46 bogdan_iancu 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
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/openser_fifo"
#
# uncomment the following lines for TLS support
#disable_tls = 0
#listen = tls:your_IP:5061
#tls_verify = 1
#tls_require_certificate = 0
#tls_method = TLSv1
#tls_certificate = "/usr/local/etc/openser/tls/user/user-cert.pem"
#tls_private_key = "/usr/local/etc/openser/tls/user/user-privkey.pem"
#tls_ca_list = "/usr/local/etc/openser/tls/user/user-calist.pem"
# ------------------ module loading ----------------------------------
# Uncomment this if you want to use SQL database
loadmodule "/usr/local/lib/openser/modules/mysql.so"
loadmodule "/usr/local/lib/openser/modules/sl.so"
loadmodule "/usr/local/lib/openser/modules/tm.so"
loadmodule "/usr/local/lib/openser/modules/rr.so"
loadmodule "/usr/local/lib/openser/modules/maxfwd.so"
loadmodule "/usr/local/lib/openser/modules/usrloc.so"
loadmodule "/usr/local/lib/openser/modules/registrar.so"
loadmodule "/usr/local/lib/openser/modules/textops.so"
# Uncomment this if you want digest authentication
# mysql.so must be loaded !
loadmodule "/usr/local/lib/openser/modules/auth.so"
loadmodule "/usr/local/lib/openser/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")
# -- rr params --
# add value to ;lr param to make some broken UAs happy
modparam("rr", "enable_full_lr", 1)
# ------------------------- 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");
exit;
};
if (msg:len >= 2048 ) {
sl_send_reply("513", "Message too big");
exit;
};
# 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
if (!method=="REGISTER")
record_route();
# 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");
route(1);
};
if (!uri==myself) {
# mark routing logic in request
append_hf("P-hint: outbound\r\n");
# if you have some interdomain connections via TLS
#if(uri=~"@tls_domain1.net") {
# t_relay("tls:domain1.net");
# exit;
#} else if(uri=~"@tls_domain2.net") {
# t_relay("tls:domain2.net");
# exit;
#}
route(1);
};
# 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) {
if (method=="REGISTER") {
# Uncomment this if you want to use digest authentication
if (!www_authorize("localhost", "subscriber")) {
www_challenge("localhost", "0");
exit;
};
save("location");
exit;
};
lookup("aliases");
if (!uri==myself) {
append_hf("P-hint: outbound alias\r\n");
route(1);
};
# native SIP destinations are handled using our USRLOC DB
if (!lookup("location")) {
sl_send_reply("404", "Not Found");
exit;
};
append_hf("P-hint: usrloc applied\r\n");
};
route(1);
}
route[1] {
# send it out now; use stateful forwarding as it works reliably
# even for UDP2TCP
if (!t_relay()) {
sl_reply_error();
};
exit;
}
============================
thanks.
Mike
---------------------------------
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Hi
I was looking for a softswitch and found that SER could be usefull for
what I am going to implement.
I want to have a Signalling GW (I think it will be SER) who will receive
calls and then direct them according to his routing table.
SER will transfer calls to Asterisk, and then Asterisk (together with
A2billing) will make the accounting for that call.
Is it the right choise to use SER?
Actually, I cannot find any forum concerning SER to check if there are
already someone who is using SER as I am planning to use.
Also, concerning WEB interface - it seams that SERWeb gives only
user-management interface and not administrator interface - am I right?
Please advise.
gabrielyans
Dear all:
avpops module' exported function avp_db_delete() can do delete one record?
example:
I use "test" as default table that has two fields.
username | number
-------------------
bob | 1
alice | 1
bob | 2
my configuration:
modparam("avpops", "avp_url", "mysql://ser:heslo@localhost/ser")
modparam("avpops", "avp_table", "test")
modparam("avpops", "username_column", "username")
modparam("avpops", "avp_aliases", "num=i:55")
modparam("avpops", "db_scheme", "number_scheme:table=test;value_col=number;value_type=string")
if (method=="INVITE"){
avp_db_load("$from/username", "$num/$number_scheme"); //load from table "test" ,number column as string value into AVP name '$num'
avp_db_delete("$from/username", "s:test") //delete all bob' record
};
if bob calling then bob'record has deleted
username | number
--------------------
alice | 1
But i want delete a record.
ex: delete number value is 1 for bob
username | number
--------------------
alice | 1
---------------------
bob | 2
avp_db_delete() can do delete one record?
Another question is avp_db_store() can do insert new data in table?
ex: insert into test (username,number) values ("mary","3")
username | number
-------------------
alice | 1
-------------------
bob | 2
-------------------
mary | 3
avp_db_store() can do it?
sorry,I still can't understand what it mean about avp_db_store() and avp_db_delete().
Can tell me?
Thanks a lot in advance
Chungyu
Hi,
I have a problem,
I use the ser 0.8.14, and I update to 0.10.99 dev 30
I have a module that send BYE to all part of comunication using fifo.
but this don't work in 0.10.99 version
can you help me ?
best regards
Lou
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Greetings,
I'm looking for a (IPv6 capable) SIP UA that functions basically as a
Music-on-Hold type client.
This will be used in a lab, and the basic desired functionality is to
have this UA register its extension, and then have any of our
soft/hard-phones be able to dial the extension and hear the music
playing (WAV, MP3, whatever) - note that there is no need for it to be
able to handle simultaneous calls... a SIP softphone that allowed
auto-answer and auto-wav-playback would be exactly what I'm looking
for...
I really just want to remove the need to have a live person on the
other end of a call to answer and speak to demonstrate VoIP
functionality during lab demonstrations and quality experiments...
Thanks,
~Aaron
Some INVITE messages to my SER include an additional header "X-Remote-Info" that
I want to be able to parse and pass to an external program using exec_msg
What is the best way to go about this?
The well known headers are passed as ENV variables but other than re-writing the
module to include a SIP_X-REMOTE-INFO how can I pass the entire INVITE to an
external script?
TIA
--
Graham Lewis
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