Hello.
I'm trying to make an "allow_trusted IP's" call without consult a mysql
database, instead i want to check against a plaintext file. Can i do this
through the permissions module?. In the README it says :
1.1. Overview
1.1.1. Call Routing
The module can be used to determine if a call has appropriate
permission to be established. Permission rules are stored in
plaintext configuration files similar to hosts.allow and
hosts.deny files used by tcpd.
Thanks in advance.
Ricardo Martinez.-
Hello asterisk users,
I you run Debian try to setup mini-dinstall for
updating SER from a local archive .
Regards
Harry
___________________________________________________________________________
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Dear SER Users,
What is the best document to use to help me install SERWEB? Where might I
find these documents? I already have SER running with MySQL.
Leo Papadopoulos
E-mail: leo(a)telecomCTO.com
Web site: www.telecomCTO.com <http://www.telecomcto.com/>
Hi
I have a really strange problem, when I am registering a IP phone, which
is connected to a simple 4 port router, it i submitting multiple
contacts into the db.
Even in the sip messages I can see multiple entries, the IP:port pairs
only differ in the port numbers. Anyone else had this problem,
Iqbal
Hello,
there is a new global parameter that can be used in config file:
'mpath'. The parameter specify the path from where to look for modules.
The value will be prefixed to the parameter of any loadmodule when this
is not absolute. For example:
loadmodule "/usr/loca/lib/openser/modules/textops.so"
loadmodule "/usr/loca/lib/openser/modules/xlog.so"
is the same now with:
mpath="/usr/loca/lib/openser/modules"
loadmodule "textops.so"
loadmodule "xlog.so"
or with:
mpath="/usr/loca/lib/openser"
loadmodule "modules/textops.so"
loadmodule "modules/xlog.so"
This will ease the migration of the config when dealing with many
instances installed with different prefixes.
Cheers,
Daniel
Hi
I have pstn -->ser -->UA
I also have asterisk hanging off ser for voicemail
Now this is all working fine, voicemail and all triggers great on no
answer, BUT to be sure I decided to look atthe sip dialogue, just to see
if all was fine, and so that I could start to clean up my config file.
When a user calls from pstn, then hit the switch, and drop into ser,
which then maps the pstn number to a local alias.
Ip phone rings,
INVITE (pstn) to (ser)
100 trying (ser) to (pstn)
INVITE (ser) to (ua)
100 trying (ua) to (ser)
180 ringing (ua) to (ser)
180 ringing (ser) to (pstn)
so far so good, now if there is no answer, and I forward to asterisk
should there be a cancel to the original INVITE, cause this is what I am
getting:
CANCEL (ser) to (ua)
200 OK (ua) to (ser)
487 request cancelled (ua) to (ser)
then i get
ACK (ser) to (ua) ------where this ACK comes fro I am not sure
200OK (ser) to (pstn) but useragent is no asterisk, hence this makes sense
ACK (pstn) to (ser)
so what I am not clear on is should the CANCEL be there, or not, it
seems to make sense that it is, just want to confirm.
Also since alot of people have the same setup, would it be a good idea
alongside onsip.org and its startup config, if we could post/have a sip
trace of common call scenarios, I know some of these are in the rfc etc,
but someone they dont seem user friendly...
Iqbal
Hello,
there is a new variable in configuration file 'retcode' that can be used
to test against an integer the value returned by last invoked function
or route. In the case of routes that uses return, it is the returned
value. The changes were done yesterday, so the pserver CVS head is now
synchronized.
It is similar to $? from bash ($? can be used also in configuration
file, having the same meaning as 'retcode'). Take care when you are
testing this value, it can be easily misleading. For example:
route[1] {
if(method=="INVITE")
return(1);
return(2);
}
route {
route(1);
if(retcode==2)
{
log("The request is a REGISTER\n");
};
if(retcode==1)
{
...
};
}
In case of REGISTER, the 'log' function is executed and the value of
retcode is changed. The proper scripting is:
route {
route(1);
if(retcode==2)
{
log("The request is a REGISTER\n");
} else if(retcode==1)
{
...
};
}
I have updated the docuwiki for 'return()' and added 'retcode' there, too.
http://www.openser.org/dokuwiki/doku.php?id=openser_core_cookbook#return_inthttp://www.openser.org/dokuwiki/doku.php?id=openser_core_cookbook#retcode
Regards,
Daniel
Hi...
Actually I have SER 0.8.14 up and running on Linux Redhat 9.0 and I´m trying to do something about digits manipulation.
I have registered on it diferent users from diferent regions of the country and I need to process local calls thru PSTN just with 7 digits even though all the users are registered with 10 digits (3 for Area Code + 7 for phone number)
For example, User from Region #1, that has 241 as area code, wants to make a call thru PSTN to a user with the same area code but only dialing 7 digits.
Here my question:
Can SER take the first 3 digits (Area Code) from the ANI (Ser user) and add them to the destination number (DNIS) before send to PSTN Gw ?
Example:
Ser user: 2417790000(a)ser.net-uno.net
Local number dialed by SER User: 9850000
Number to send to PSTN: 2419850000
I have no idea how I can do something like that
Best Regards,
Daniel Mizrachi
I have pulled this from my working OpenSER configuration file, so I can
verify that the logic works.
The following example takes advantage of the recent marriage of AVPs
with the TEXTOPS subst_uri() function.
The problem is to dynamically create a call to 1 (area code) 555-1212
for a user that dials 411 on his/her phone.
The area code (also known as an NPA) is assigned to a user in the
usr_preferences table. A simple change to the value in the table will
generate a call to a different "local directory assistance operator".
The ability to dynamically associate an NPA with a customer can be used
to give the customer a "virtual local calling area". If only 7 digits
of a phone number are entered by a user, OpenSER can easily prepend a 1
plus the users dynamic NPA. I remember the days when a 1 + area code
were optional for calls to your local calling area. This example is a
first step in that direction.
An important off-shoot of this will be to perhaps assist in one aspect
of the FCC's 911 decision. If the NPA (and perhaps the NXX) can be
pulled from a location database and associated to an end user in the
usr_preferences table, (did someone say LERG) then when a call to 911 is
made, the generated phone number could be routed to a specific PSAP
operator. If the end user changes his/her location, a web interface can
be used to copy the new NPA/NXX to the usr_preferences table.
Database Entries:
Table: usr_preferences
Username: user1
Attribute: npa
Value: 123
OpenSER.cfg
route[1] {
#---------------------------------------------------------------------------------------
# 411 Local Directory Assistance
#---------------------------------------------------------------------------------------
# check the called number to see if it is 411
if (uri=~"^sip:411@.*") {
# check the AVP database for a match with the "user", not
"user@domain" part of the From: header
# checking for user@domain should also work.
# if the "user" has a database entry with an "npa" attribute, load
the AVP.
if (avp_db_load("$from/username","s:npa")) {
# rewrite the uri by replacing the 411 with 1 plus the AVP "npa"
plus 5551212
subst_uri('/^sip:411@(.*)$/sip:1$avp(npa)5551212@\1/');
# delete the AVP npa to release resources
# the release of AVP resources is supposed to happen automatically
so the delete
# may not be strictly required.
avp_delete("s:npa");
} else {
# the AVP database didn't find a match with the "user" so we rewrite
# the uri with a default call to "local directory assistance".
rewriteuri("sip:1xxx5551212@example.com");
};
break;
};
}
Regards,
Norman Brandinger
norm at goes dot com
Hi,
we have been running from several months a single server with SER 0.8.14
with no problems. Now, in order to have some kind of redundancy, we are
setting up a backup SER server and we are replicating the information from
the primary server to the secondary with the t_replicate command
if (method=="REGISTER") {
if (!radius_www_authorize("tnet.it")) {
www_challenge("tnet.it", "0");
break;
};
if (!save("location")) {
sl_reply_error();
} else {
if (!src_ip==backup.foo.bar) {
t_replicate("backup.foo.bar", "5060");
};
};
break;
};
It seems to work fine but after a few minutes running the server goes down
and we get the following message on /var/log/messages:
Jun 29 19:19:10 dns2 ser[5648]: child process 5649 exited by a signal 11
Jun 29 19:19:10 dns2 ser[5648]: core was not generated
Jun 29 19:19:10 dns2 ser[5648]: INFO: terminating due to SIGCHLD
Jun 29 19:19:10 dns2 ser[5652]: INFO: signal 15 received
Jun 29 19:19:10 dns2 ser[5650]: INFO: signal 15 received
Does anybody have any idea about how to solve this issue?
Thanks in advance.
Regards,
Conrado Camacho
ccamacho(a)tnet.it
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