Hi,
I have a SER + Mediaproxy. I have not any problem the call between SIP clients (behind or not the NATs)
but when I try to call to PSTN (via cisco) I only have RTP traffic from SIP client to PSTN.
Summarizing, the path of rtp traffic would have to be from:
up: SIP Client ----> SER ---> GW-PSTN
down: SIP Client <---- SER <--- GW-PSTN
but, really is:
up: SIP Client ----> SER ---> GW-PSTN
down: SIP Client <------------- GW-PSTN
I use the command:
rewritehostport("212.xxx.xxx.xxx:5060");
when I match a geographic number.
The complete scheme is:
SIP Client ---- NAT --------- SER+Mediaproxy -------- SIP Server --- GWPSTN
Some idea?
Thanks,
--
Alberto
Hi,
I've installed 4 RPM files on a Fedora CORE 3.
The RPM are:
ser-0.9.3-0.i386.rpm
php-pecl-classkit-0.4-2.i386.rpm
ser-mysql-0.9.3-0.i386.rpm
serweb-0.9.3-0.i386.rpm
I enabled the httpd and mysql service.
I'm not sure what to do next.
All the Documentation I have is related to source flies and not RPM.
Ser is working but I don't know how to configure the serweb in order for
it to work.
Could you Please help?
Thank you,
Elad.
Hello
We are running SER 0.8.14 and FreeRadius 1.0.4 on
Slackware 10 kernel 2.6.12.3. Yesterday, we had
problems with some functions of SER.
radius_www_authorize & www_challenge
radius_proxy_authorize & proxy_challenge
if(method=="REGISTER")
{
Make sure that user's dont register infinite loops
# (note: does not match with folded lines)
if(search("^(Contact|m):.*@(domain\.com\.br)"))
{
log(1, "LOG\n");
sl_send_reply("476","No Server Address in Contacts
Allowed" );
break;
};
#if (!radius_www_authorize(""))
#{
#www_challenge("","0");
# break;
#};
#if (!radius_proxy_authorize( "")) {
#proxy_challenge("sipproxy.taho.com.br", "0" );
#break;
#};
save("location");
#if (!save("location")) {
# sl_reply_error();
#};
#m_dump();
break;
};
This block ( if(method=="REGISTER"){block} ) has these
functions. When "radius_www_authorize &
www_challenge" or "radius_proxy_authorize &
proxy_challenge" are uncommented, the SER becomes too
slow: it receives around 5 consecutive invites in 7
seconds. After the fifth INVITE, SER answers with a
487 to the User Agent. The call to PSTN is
established after around 30 seconds . When we comment
these functions, the system runs perfectly, fast and
fast. But is strange, because these functions are
called normally in the routes we the INVITE is
received:
route[5]
.............
if (!radius_proxy_authorize(""))
{
proxy_challenge( "domain.com.br", "0" );
break;
};
.............
In the route, they work perfectly. I'd like to know if
I'm doing wrong things in this scripts.
Thanks a lot, friends.
Bruno Machado
__________________________________________________
Faça ligações para outros computadores com o novo Yahoo! Messenger
http://br.beta.messenger.yahoo.com/
Hi,
I am planning to use SER to send some SIP requests. Is it possible to
initiate requests like INVITE from SER? If yes, how can I do that?
I saw the description of "serctl file <file_name" and it is supposed to
be used to read commands from a file. Does it refer to rest of the
serctl commands or SIP requests like INVITE, REGISTER?
Appreciate your help,
Regards,
Bhaskar...
Hello Everyone,
I am new to this list. I would greatly appreciate anyone who can tell me
what I need to do next in order for SER version 0.9.3 to use the Jabber
Module correctly?
This is what I have done so far (on a Red Hat Linux 9 machine):
Installed Jabberd 1.4.4
Installed GNU pth-2.0.4
Installed Expat 1.95.8
Installed Libidn
-I have Modified the config file for SER to load the "Jabber.so" module
after the "pa.so" and "tm.so" modules have been loaded
-(I'm sure this is where I am stuck, what to do with parameters in
config file for SER but I have started to place those in as well, I am
just confused about making the documentation match my settings/variable
names)
-I have partially created a database in MySQL but I have not put tables
or data in it because I have never done this before.
This is what I can/cannot do:
-I can run the jabber server
-I cannot run the SER with the config file which includes the Jabber.so
module load command.
What other files do I need to modify or download? What information do I
need? I'm not very familiar with linux, so please assume I know basic
commands/knowledge... Any comments/suggestions/websites/help is GREATLY
appreciated!
Thanks,
Saleh
Is there any good, DETAILED information on AVPops usage that, say, doesn't
read like a developers API reference page? Currently, I'm cleaning up portions
of my ser.cfg which have horrid, ugly hacks in them that I put in to handle
this case or that case...
For instance, I use the calls_forwarding table to handle call blocking and
call forwarding info (since it seems a little more reasonable than
usr_preferences as it simply has more relevant tables). For call blocking, I
can say give a purpose of 'blocking' and an action of relay or reply... with
parameters set to what kind of reply or where to relay (voicemail server?
Alternate server? etc). For forwarding, I can give an action of forwarding
with a parameter of where to forward and another to see if he has permission
to forward to PSTN numbers. Seems like a very handy table for it.
Now, my current method of doing this is with this horrid mixture of external
scripts, stripping portions off the SIP_HF_FROM header (since it comes with
additional info like tags and brackets and comments), and feeding the new
stuff back into an SQL query to determine whether or not the call needs to be
blocked, forwarded, and if either, where it goes.
It's a mess, I'll admit, but it's because I've been poring over AVPops stuff
and have yet to figure out what I'm actually DOING with any of it. I get that
it uses little i:XX constructs as variables (which is bizarre, but whatever --
I imagine I could alias them to something human-readable). And... well...
that's about as far as I've gotten deciphering the API docs and making them
into something usable.
I imagine I would what... set the avp_table to be calls_forwarding, and then
create some schemes for it (?)... but after that, actually doing things like
taking just the sip:bob@fred.com portion of an incoming uri (which would look
along the lines of "Ima Doofus"
<sip:bob@fred.com:5060;user=tel>tag=something;tag2=something else ) and then
trying to determine if that user is on the list of blocked users and, if so,
what should be the response action... or taking the callee info, and if the
callee isn't available, checking to see if he has a forwarding number,
checking to see where it goes, etc, etc.
It all SEEMS like it should be elementary stuff, but I'm simply not getting
it. Anyone have any good references, well-detailed documentation, long,
in-depth books on the topic they could send me toward?
I'd be your friend for, like, ever. :)
N.
You could look at the Getting Started document on www.onsip.org
because we use the AVPOPS module in several examples and we give a
much deeper description of what the functions do.
However, we only use a subset of the AVPOPS library.
Regards,
Paul
On 9/22/05, sip <sip(a)arcdiv.com> wrote:
> Yeah... the problem with this documentation is that it was clearly written by
> someone who doesn't WRITE documentation as a matter of course, but needed
> something to be out there just to say there's documentation.
>
> It's a little like a developer writing docs for his own code or API. VERY
> understandable to the developer himself, but not so crystal to anyone else.
> This is one of the primary reasons I never let my developers write their own
> API docs. They get with a tech writer, and the tech writer asks questions
> about what the API does, and then writes documentation in such a way that
> monkeys (such as myself) can understand.
>
> I was looking for something a little more... lucid.
>
> If it doesn't exist, cool. I'll keep muddling away and perhaps I'll hit that
> eureka point of understanding somewhere, but I was hoping.
>
> N.
>
>
> On Thu, 22 Sep 2005 09:29:31 -0400, Paul Hazlett wrote
> > The offical documentation for AVPOPS (for SER-0.9.x) can be found here:
> >
> > http://www.voice-system.ro/docs/avpops/0.9.0/
> >
> > Regards,
> > Paul
> >
> > On 9/22/05, sip <sip(a)arcdiv.com> wrote:
> > > Is there any good, DETAILED information on AVPops usage that, say, doesn't
> > > read like a developers API reference page? Currently, I'm cleaning up portions
> > > of my ser.cfg which have horrid, ugly hacks in them that I put in to handle
> > > this case or that case...
> > >
> > > For instance, I use the calls_forwarding table to handle call blocking and
> > > call forwarding info (since it seems a little more reasonable than
> > > usr_preferences as it simply has more relevant tables). For call blocking, I
> > > can say give a purpose of 'blocking' and an action of relay or reply... with
> > > parameters set to what kind of reply or where to relay (voicemail server?
> > > Alternate server? etc). For forwarding, I can give an action of forwarding
> > > with a parameter of where to forward and another to see if he has permission
> > > to forward to PSTN numbers. Seems like a very handy table for it.
> > >
> > > Now, my current method of doing this is with this horrid mixture of external
> > > scripts, stripping portions off the SIP_HF_FROM header (since it comes with
> > > additional info like tags and brackets and comments), and feeding the new
> > > stuff back into an SQL query to determine whether or not the call needs to be
> > > blocked, forwarded, and if either, where it goes.
> > >
> > > It's a mess, I'll admit, but it's because I've been poring over AVPops stuff
> > > and have yet to figure out what I'm actually DOING with any of it. I get that
> > > it uses little i:XX constructs as variables (which is bizarre, but whatever --
> > > I imagine I could alias them to something human-readable). And... well...
> > > that's about as far as I've gotten deciphering the API docs and making them
> > > into something usable.
> > >
> > > I imagine I would what... set the avp_table to be calls_forwarding, and then
> > > create some schemes for it (?)... but after that, actually doing things like
> > > taking just the sip:bob@fred.com portion of an incoming uri (which would look
> > > along the lines of "Ima Doofus"
> > > <sip:bob@fred.com:5060;user=tel>tag=something;tag2=something else ) and then
> > > trying to determine if that user is on the list of blocked users and, if so,
> > > what should be the response action... or taking the callee info, and if the
> > > callee isn't available, checking to see if he has a forwarding number,
> > > checking to see where it goes, etc, etc.
> > >
> > > It all SEEMS like it should be elementary stuff, but I'm simply not getting
> > > it. Anyone have any good references, well-detailed documentation, long,
> > > in-depth books on the topic they could send me toward?
> > >
> > > I'd be your friend for, like, ever. :)
> > >
> > > N.
> > >
> > > _______________________________________________
> > > Serusers mailing list
> > > serusers(a)lists.iptel.org
> > > http://lists.iptel.org/mailman/listinfo/serusers
> > >
>
>
Hello All,
Immediately I start SER, I get the following errors using SER with Audiocodes. Is it SER working with Audiocodes, if not which devices are recommended ?
MP-104 errors
Version ID: 4.60A.016.003
1d:23h:23m:11s ( sip_stack)(653 ) !! [ERROR] SIPStackMngr:
Message was not Parsed correctly !
1d:23h:23m:11s ( sip_stack)(654 ) !! [ERROR] Last parsed line
was :
1d:23h:23m:11s ( sip_stack)(655 ) !! [ERROR] The error was =
Unexpected input
1d:23h:23m:11s ( sip_stack)(656 ) !! [ERROR] The Line of the
error was = 1
1d:23h:23m:11s ( sip_stack)(657 ) !! [ERROR] The Column of the
error was = 2
Thanks,
Ggtel
Hi,
I'm using the append_rpid_hf() command to add the remote party id.
I've implemented it like this:
modparam("auth", "rpid_prefix", "<sip:")
modparam("auth", "rpid_suffix", "@xxx.xxx.xxx.xxx;user=phone>;party=calling;id-type=subscriber;screen=yes;privacy=off")
[...]
# first the caller needs to be authenticated
if (uri=~"^sip:[0-9]*@(xxx.xxx.xxx.xxx|(voip1\.)?test\.com)") {
if (!(src_ip==xxx.xxx.xxx.xxx | method==ACK | method=="CANCEL" | method=="BYE")) {
if (!proxy_authorize("xxx.xxx.xxx.xxx", "subscriber")) {
proxy_challenge( "xxx.xxx.xxx.xxx","0");
break;
} else if (method=="INVITE" & !check_from()) {
log(1, "LOG: Spoofed from attempt\n");
sl_send_reply("403", "Use From=id next time");
break;
};
};
};
append_rpid_hf();
What I noticed is that sometimes ser sends it and sometimes not. I have a value in the rpid column of the user...
Has anyone of you an idea why that happens??
Thanks!
Sebastian
Hi,
Will SER able to determine if I put a whole network or subnet in
the trusted table? So that I will not need to put all the ip addresses
of my network in the trusted table.
Thanks,
--
Ryan Pagquil
Infodyne Inc. - PhilOnline.com
3603 Antel Global Corporate Center
Doña Julia Vargas Ave.
Ortigas Center Pasig City
Tel: 687-0715
Web: www.philonline.com
