Dear all
I am facing problems installing OpenSER 1.3 from branch on a Debian
4.0R2 running on a SUN UltraSPARC IIi server (sparc64).
After retrieving all required files and executing # make mode=debug, the
second file's compilation fails already:
Compiling blacklist.c
....
Error: Architecture mismatch on "membar".
(Requires v9|v9a|v9b; requested architexture is sparclite.)
Does anyone know what could be the reason for that. On i386 I never
faced problems, and as OpenSER supports Solaris, it should also support
Debian on sparc64 architecture or not?
Thanks a lot in advance.
Best regards
Sebastian
I need to append the callerid as tech prefix to all outgoing calls for
complience with my billing Server. I'm
using carrierroute model for routing. Can i do this or do i have to resort
to do it from within the openser script
--
TC
Hi, imagine I need to log two extra fields in accounting:
- Source-IP
- Proxy-IP
"Source-IP" will be dynamic containing the value of $si.
"Proxy-IP" is a fixed string value = "88.88.88.88", but... how to log this
value?:
* modparam("acc", "radius_extra", "Source-IP=$si; Proxy-IP=88.88.88.88")
Of course the above line fails since the IP string should be a text so should
be between " ":
* modparam("acc", "radius_extra", "Source-IP=$si; Proxy-IP="88.88.88.88"")
This fails obviously.
* modparam("acc", "radius_extra", "Source-IP=$si; Proxy-IP=\"88.88.88.88\"")
This fails too.
* modparam("acc", "radius_extra", "Source-IP=$si; Proxy-IP='88.88.88.88' ")
This fails too.
Is it impossible? any suggestion?
PD: Yes, I know I can do a bit "dirty" trick:
modparam("acc", "radius_extra", "Source-IP=$si; Proxy-IP=$var(proxy-ip)")
...
$var(proxy-ip) = "88.88.88.88";
but I'd prefer not to do it.
--
Iñaki Baz Castillo
Hi friends,
Could you please help me how i can make dialplan to call local each subscriber using some special prefix?
I tried the following but it did not work.
if (uri=~"sip:00[1-9][0-9]+@.*") {
strip(2);
rewritehostport("pstncarrier:5060");
route(1);
exit;
}else if (uri=~"sip:[1-9][0-9]+@.*")
rewritehostport("pstncarrier:5060");
route(1);
exit;
}else if (uri=~"sip:01[1-9][0-9]+@.*")
strip(2);
if (!lookup("location")) {
sl_send_reply("404", "Not Found");
exit;
};
route(1);
exit;
};
The first two above condition is working fine but when we trying to call using 01 prefix + subscriber ID the call not going and the message comming 404 Not Found.
Regards,
www.Go4Calls.Com
VoIP Forums
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Hi,
is it possible to check message contents if they contain some sting in case
of content type text/plain??
will it work if I use search('some string'); from textops module?
Regards
Tomasz
Thanks for the reply Alex! I really appreciate it.
I've been reading manuals & sip specs all day, and I think that I found my answer, and it really is an easy one. here's my own answer to my question, can you let me know if this sounds right?
When a message is send from a user agent (on the public internet) to my gatweway & then to the proxy (on a private network), the gateway will add a via on to the message. that's how open SER (or any other proxy) can send a response message back to user agent on the public internet via the gateway.
It's really the same thing as setting up a registrar, or any other kind of proxy/user agent that's going to communicate via a SBC of any kind. So, it that right? is that how the SIP via header is used?
I understand the part about registrations. We're not currently doing that in our network, we're using all static IP addressing. I plan on using registrations in the future. the gateway that I'm using can act as a session border controller, but it can also act as a class 4/5 switch (TDM or IP). so it's some serious overkill for what I want to do here. :o)
Thanks again for the reply,
--Doug
----- Original Message ----
From: Alex Balashov <abalashov(a)evaristesys.com>
To: Doug McLetchie <dougmcletchieatwork(a)yahoo.com>
Cc: users(a)lists.openser.org
Sent: Wednesday, February 6, 2008 1:23:51 AM
Subject: Re: [OpenSER-Users] newbie questions
Doug,
Doug McLetchie wrote:
> For inbound calls (calls coming from another carrier to my big expensive
> gateway & destined for a specific PBX), I'd like my gateway to send the
> call to the proxy, who will determine which PBX to send the call to, and
> then send the call to to correct PBX via the Gateway. I don't
> understand how to set up OpenSER to send the call via the Gateway. If I
> provision the static ip address of the PBX in the proxy, won't it try to
> send directly to that IP address instead of sending it to the Gateway?
> I think that the feature that I'm looking for is something like an
> outbound proxy, for the proxy. (does that make sense?)
OpenSER can certainly do what you are trying to accomplish.
SIP routing is done by URI, which consists of a "user" part and a
"domain" part. The "user" part is the number (or alphanumeric
identification, in the case of pure-VoIP peering) and the "domain" part
is the IP "place" at which the "user" part is reachable.
When you route a call to some URI, what you are really saying is, "Here,
domain, you must know what to do with this 'user' part - i.e. have
reachability information for it (a SIP contact bound from a
registration, for example)."
A proxy by itself isn't enough. If you need to reach these PBXs, you
clearly need to know how to reach them. This requires a SIP registrar
somewhere, so that the PBXs can register against it and say, "Here, you
can reach me at such and such IP and port." Or, I suppose, you can
define these contacts statically with a database interface from the
proxy, in which case you don't need to use a registrar.
I don't know what this Big Expensive Gateway is, but if it's something
like a Session Border Controller, it should be able to forward SIP
REGISTER requests to your proxy/registrar. Or do they register against
the gateway?
Either way, you can perform this resolution with OpenSER.
The problem you *might* run into is sending the same logical call leg
back to the Gateway. Depending on what it is, it may not like that and
perceive a call routing loop, because the call that went through it to
the proxy is the "same" call (in terms of SIP Call-ID, and other things
that make up a logical call "leg") that is now being sent back around to
it. This problem is often solved with the introduction of a
back-to-back user agent which can re-originate a different call leg.
--
Alex Balashov
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599
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Good Day All,
I am trying to get accrue AcctSessionTime of calls. I am using Openser+Asterisk but some time asterisk CDR are not accoutered, record missing + if some one call 15 min it will come 2 hrs in cdr.
Is Mediaproxy will be best choice to get trust able AcctSessionTime?
If Mediaproxy AcctSessionTime is trustable, i have to use ACC module of Openser or only dbaccounting configuration will be enough to store records direct from Mediaproxy into MySQL backend?
Thank You
Abdul Latef
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Hi,
I'm using SEAS module to interact with WeSIP Application Server. I'm using
the version wesip-02-04-07-beta-0.1.4.tar.gz of WeSIP.
I've managed to configure it and I'm also seeing that the messages are
received by WeSIP. I'ld like to know:
1. Are there any special files to which WeSIP writes the received messages?
2. In case, some one wants to modify the received message then is there a
possibilty to send the modified message back to OpenSER?
3. I have used the above tar file, to me it appears that this file only
contains the binary form of WeSIP. So where can I get the complete source
code of WeSIP?
Many thanks,
Rajat
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Hi all,
the database modules and the supporting infrastrucure in the OpenSER core are
heavily reworked in the trunk version at the moment.
Common functionality that is present in all database modules were integrated
in the core, new functionality have been added. Its planned to unify the
overall behaviour and structure of all modules that are interfacing with
databases, to make them more maintainable, more safe and also more bug-free.
Important foundations for this changes have already been created.
As part of this process the database modules names were changed. All these
module names are now prefixed with 'db_'. The new names of the modules are
now db_mysql, db_postgres, db_unixodbc, db_berkeley, db_text, db_flatstore.
If you use one database module with the trunk version, you must change some
parts of the configuration.
Old module configuration:
loadmodule "mysql.so"
modparam("mysql", "ping_interval", 300)
modparam("mysql", "auto_reconnect", 1)
New module configuration:
loadmodule "db_mysql.so"
modparam("db_mysql", "ping_interval", 300)
modparam("db_mysql", "auto_reconnect", 1)
The database URLs for modules that connect to a database, e.g. usrloc don't
need to be changed, if you use mysql, postgres, unixodbc or flatstore DBs.
For the dbtext and db_berkeley modules the new database URLs are:
modparam("usrloc", "db_url", "text://my-database-path")"
modparam("usrloc", "db_url", "berkeley://my-database-path")"
(you need to adapt the path to your setup, of course).
If you encounter any bugs that were introduced during this change, please open
a bug report on the tracker.
Best regards,
Henning