hello friend,
iam tyring to use the trabas voip-billing-0.0.2
iam following the INSTALL information given
in that .
iam following the fullsteps in that one
till 3 step i executed successfully
i.e
shell> ./bin/genvhost.pl -s=voip.mycompany.com
-e=voip(a)mycompany.com \
/usr/local/dataweb > etc/vhost.conf
now vhost.conf looks like this
# $Id: vhost-tmpl.conf,v 1.4 2002/07/09 04:41:11
hasant Exp $
#
# Copyright (C) 2002 Trabas. All rights reserved.
# Please see the file COPYING for more information
<VirtualHost 202.*.*.19>
ServerAdmin rama(a)rama.kol.net.in
ServerName server.pol.net.in
ErrorLog logs/server.pol.net.in-error.log
CustomLog logs/server.pol.net.in-access.log common
# use this instead if you don't want to log access
#CustomeLog /dev/null common
DocumentRoot
/usr/local/dataweb/voip-billing-0.0.2/web
<Directory /usr/local/dataweb/voip-billing-0.0.2/web>
AllowOverride None
Options ExecCGI FollowSymLinks
Order allow,deny
Allow from all
# uncomment the following lines to take benefit
# from mod_perl (if you have it)
#<IfModule mod_perl.c>
# <FilesMatch "\.(cgi|pl)$">
# SetHandler perl-script
# PerlHandler Apache::Registry
# </FilesMatch>
#</IfModule>
</Directory>
# WARNING: Don't use this for production use!
#Alias /templates/
/usr/local/dataweb//voip-billing/resources/templates/
#<Directory
/usr/local/dataweb//voip-billing/resources/templates>
# Options Indexes
# AllowOverride None
# Order deny,allow
# Deny from all
# Allow from localhost
#</Directory>
Alias /js/
/usr/local/dataweb/voip-billing-0.0.2/resources/js/
<Directory
/usr/local/dataweb/voip-billing-0.0.2/resources/js>
Options None
AllowOverride None
Order allow,deny
Allow from all
</Directory>
Alias /images/
/usr/local/dataweb/voip-billing-0.0.2/resources/images/
<Directory
/usr/local/dataweb/voip-billing-0.0.2/resources/images>
Options FollowSymLinks
AllowOverride None
Order allow,deny
Allow from all
</Directory>
# temporary cgi-bin until we fix the hardcoded
references
# to the script name
ScriptAlias /cgi-bin/
/usr/local/dataweb/voip-billing-0.0.2/cgi-bin/
<Directory
/usr/local/dataweb/voip-billing-0.0.2/cgi-bin>
Options None
AllowOverride None
Order allow,deny
Allow from all
</Directory>
</VirtualHost>
i include the the path of this vhost.conf
with INCLUDE in the httpd.conf
so now i restarted the httpd
and when i press the http://server.pol.net.in
it shows the apache default document than the billing
site.
so where i might be going wrong
please guide me
with regards
rama kanth
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When the caller to the offline user (connected to SEMS) hungup first the BYE message must be sent to SEMS for release the call.
How i can send the BYE to SEMS only if the user that hungup are talking to SEMS ?
Ex:
if ( BYE or CANCEL ) {
if (for established SEMS session) {
vm("/tmp/am_fifo","bye")
} else {
end_media_session();
t_relay();
};
break;
};
Thanks
Ezequiel Colombo
Hi all,
Sorry if this is a veruy stupid question as I am a newbie to sems.
I need to have a different IVR script pointed from ser.cfg to have
difference features, may I know how can I add another script file to call
from the sems.conf using the IVR.so module again but with different scripts.
Please help.
Regards,
Shirley
Hi all,
I am having this problem with ./rtpproxu command.
Before I start this command, my call forward and conference works fine with
PC to PC and PC to PSTN but users behind symmetric NAT have no audio when
connected.
When I started the command, the scenario go the other way round, symmetric
is working but my call forward and conference is not working. I have
attached my cfg file.
Please help and thank you in advance!!!
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;
};
#!! Nathelper
if (nat_uac_test("3")){
if (method == "REGISTER" || !search("^Record-Route:")){
log("LOG:Someone trying to register from private IP,
rewriting\n");
fix_nated_contact(); # Rewtite contact with source
IP of signalling
if(method== "INVITE"){
fix_nated_sdp("1"); # Add direction-active
to SDP
};
force_rport();
setflag(6);
};
}else{
fix_nated_contact();
};
# 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 (uri==myself) {
if (method=="REGISTER") {
save("location");
break;
};
};
if (uri=~"sip:019[0-9]+@myserver.com") {
strip(1);
rewritehostport("1.2.3.4:5060");
t_relay();
break;
}
if (method=="INVITE") {
fix_nated_sdp("3");
append_hf("P-hint:Invite");
# native SIP destinations are handled using our USRLOC DB
if (!lookup("location")) {
sl_send_reply("", "Connecting");
break;
};
};
if (method=="ACK"){
sl_send_reply("", "ACK");
break;
};
if (isflagset(6)){
force_rtp_proxy();
append_hf("P-hint:Flag6");
}else{
force_rtp_proxy();
t_on_reply("1");
};
# 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;
};
}
# all incoming replies for t_onrepli-ed transactions enter here
onreply_route[1] {
sdp_mangle_port("-12000");
if (isflagset(6) && status=~"2[0-9][0-9]"){
append_hf("First Route");
fix_nated_contact();
fix_nated_sdp("1");
force_rtp_proxy();
} else if (status=~"2[0-9][0-9]"){
if (uri=~"sip:019[0-9]+@myserver.com") {
append_hf("019");
fix_nated_contact();
fix_nated_sdp("3");
}
else{
append_hf("2nd Route");
fix_nated_contact();
fix_nated_sdp("2");
fix_nated_sdp("3");
force_rtp_proxy();
}
};
}
Regards,
Shirley
Hi all,
I'm looking for a way to get the SDP part of a SIP message (for example during
an INVITE) and pass it to an extern application via exec_msg. I don't know how
to do that stuff so if somebody can help me, I'll be grateful.
Thanks
Laurent
Hi Ri,
We use successfully mediaproxy with both stable and unstable branches
of ser on Debian systems. There is no known reason it should crash,
maybe IPTEL guys would have more clues as I saw this problem being
reported several time.
Adrian
--
Hello,
I'm using lates ser CVS stable with mediaproxy (from unstable ser
directory
and also using the mysql create with unstable ser_mysql.sh).
SER crash in the same moment when a sip cliente get register with the
correct user and password, i have attach the syslog.
Am I using the right combination of SER+mediaproxy ? or do i need to use
only the unstable ser (with it's mediaproxy include).
Hi to all
when I try to insert new user with serctl add, I can see this message:
[root@webby ser]# serctl add gino gino gino@localhost
MySql password:
error: overlap with an existing alias
but, the table aliases is empty.
Why?
best regards, Andrea
Guys,
For those relying on Radius accounting generated by SER beware that the
combination of RadAcctId-AcctSessionId is not at all unique. If you
use MySQL for storage of radius accounting you will miss CDRs every now
and then (because is a unique index) and the update queries will
update calls from the past as a result, calls with huge duration are
being generated.
The problem is generated by a combination of factors:
1. Some UAs reuse same call id among multiple calls (Grandstream
systematically show this behaviour)
2. The Acct-Unique-Session-Id is not unique either, it repeats itself
every now and then.
I would appreciate to see reactions for this problem from the
developers.
Regards,
Adrian
Hi,
All,
I have installed and configuered the SER, and it run well except the
voicemail.
When I click the upload greeting in SERWEB, it display:
Warning: Unable to create '/var/greetings/6006.wav': No such file or
directory in /usr/local/serweb/user_interface/voicemail.php on line 50
Another question is:
How I use the Voicemail? Have any user guide document?
Thanks
siduanfeng
Hi Guys,
I have installed serweb and the admin pages are running without a problem.
But I'm having a problem with the user interface pages.
When I try to login into the user pages.. everything hangs at my_account.php Apache log shows a 302 redirect to my_account.php but the page wouldn't load.
If I try to click on the account link from the admin pages, the same thing happens.
Anyone had any issue with my_account.php ?
thanks,
Zak
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