Hello Everyone,
I have finally got my sip router up and its running well, tested it to be
true -thus far - with asterisk backend w/PSTN. Now, i want to start
offering services, but only have one pstn channel. I was hoping that
someone could advice me on a service provider that is willing to allow me to
route calls to their sip router for a (monthly) fee, and if this is possible
so I can offer services to users without being limited to a single channel
and then I can see how sip simulatenous calls really perform? Advice?
How much is fee for this type of situation to me (approximately range)?
Please can some-1 provide me with recommendations providers? I need this
for me betterment soon.
Help?
jwshel
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Hello,
I am new to the community so please forgive me if I am missing some
obvious answer but till now I have found only some suggestions what
one could or could not do, but is there any known-to-work proven
configuration file which rewrites From?
rr
Hi Folks,
Is there any trick to log ACK method using siptrace module?
I'm trying with:
...
modparam("siptrace", "trace_flag", 22)
modparam("siptrace", "trace_on", 1)
route {
if (is_method("INVITE|ACK|BYE")) {
setflag(22);
};
...
but only INVITE and BYE method has been logged!
I'm using OpenSER 1.1.0-notls.
Thanks for any help
Rodrigo Telles
Hi Folks,
Today I realised something strange in openser-1.1.0-notls.
I'm trying to mach any ACKs to do something (e.g. logging) but its doesn't work at all.
I'm using "method" like this:
If (method=="ACK") {
xlog("L_ALERT","ACK received\n");
};
and I it doesn't work, but...
if (is_method("ACK")) {
xlog("L_ALERT","ACK received\n");
};
works perfectly!
I could not find anything about it in archive mail list neither in doc.
Rodrigo P. Telles
Hi,
I have a radius accounting configuration I would like to share, if
someone needs. Please can I get your feedback guys if this can be done
in a better way. Just share your thoughts if you have done something
similar.
I need to account to my RADIUS the source IP (of the calling UA), the
destination IP (called UA/gw) and the type of the leg - originating or
terminating depending on whether the call is from the UA or to the UA.
To do this I decided to use the cisco vendor-specific radius AVPs
which have similar fields but for h323 calls.
In my dictionary file of radiusclient-ng i added this according to the
cisco specification:
VENDOR Cisco 9
ATTRIBUTE Cisco-AVPair 1 string Cisco
ATTRIBUTE H323-remote-address 23 string Cisco
ATTRIBUTE H323-call-origin 26 string Cisco
and in my openser.cfg i have:
modparam("acc", "radius_extra", "Cisco-AVPair=$avp(s:c_src_ip);
H323-remote-address=$avp(s:c_dst_ip);
H323-call-origin=$avp(s:c_origin); Cisco-AVPair=$avp(s:c_call_id)")
Now in my config i fill in the values of the c_src_ip, c_dst_ip,
c_origin, and c_call_id avps according to the direction of the call.
Here is how i do it for the originating leg, terminating is similar:
route[14] { #ACCOUNT ORIGINATING
if (is_direction("downstream")) {
avp_printf("$avp(s:c_src_ip)", "call-origin-endpt=$si");
avp_printf("$avp(s:c_dst_ip)", "$ru");
avp_subst("$avp(s:c_dst_ip)", "/sip:.*@(.*)/h323-remote-address=\1/g");
avp_subst("$avp(s:c_dst_ip)", "/(.*):.*/\1/g");
avp_printf("$avp(s:c_origin)", "h323-call-origin=originate");
avp_printf("$avp(s:c_call_id)", "call-id=$ci");
}
if (is_direction("upstream")) {
avp_printf("$avp(s:c_src_ip)", "$ru");
avp_subst("$avp(s:c_src_ip)", "/sip:.*@(.*)/call-origin-endpt=\1/g");
avp_subst("$avp(s:c_src_ip)", "/(.*):.*/\1/g");
avp_printf("$avp(s:c_dst_ip)", "h323-remote-address=$si");
avp_printf("$avp(s:c_origin)", "h323-call-origin=originate");
avp_printf("$avp(s:c_call_id)", "call-id=$ci");
}
setflag(2);
}
Now as you can see i take one of the IP addresses from the request-uri
(and strip it with avp_subst), which I am not sure if always contains
IP. Is there a better way - i can call this route after
rewritehostport or lookup so the destination IP will hopefully be in
the r-uri of the message, but maybe there is a better way of getting
it?
Any input/ideas will be greatly appreciated.
Best,
Dimo
Hello everybody,
I will see many of you at Berlin, next week. The summit has a BoF
session on Tuesday, 15:45-17:00 and continues next days with morning and
afternoon sessions: see http://www.openser.org for more.
Anyone who wants to get in contact before or to have a private meeting,
the can find us at booth 321. All participants registered before
November 1 got full VON conference pass (valid for the three days,
November 6-8, for exhibition and conference, including special events,
like party, etc..).
Cheers,
Daniel
Howdy all!
I am trying to setup the LCS communicating with SER and ASTERISK in
the company I work for. I read the following tutorial:
http://www.voip-info.org/wiki/view/MS+LCS+2005+%252F+SER+%252F+Asterisk+Int…
But there the author points the necessity of 2 LCSs, but I just
couldnt grasp why the need of a second LCS.
Could anyone who has already setup that give me a hint of why it is
necessary a second LCS?
I thought I could route LCS to communicated directly with SER. I tried
to use only one LCS(one that uses TLS) but it didnt communicate with
SER.
Thanks in advance,
Felipe Louback
--
Master Student - Electrical Engineering Department
Computer Engineering and Telecommunications Research Group
Universidade Federal de Minas Gerais - Brazil
"Come to me, all you who are weary and burdened, and I will give you rest."
Matthews 11:28
You do not need to check usrloc t to implement what you want. Define a timer
and when you forward the INVITE to the PBX, you will automatically jump to
the failure_route[] if you do not receive a response from the PBX (PBX
down!) and you can put your logic there.
>From: "sandrinho1976(a)libero.it" <sandrinho1976(a)libero.it>
>To: "serusers" <serusers(a)lists.iptel.org>
>Subject: [Serusers] how to search in usrloc?
>Date: Fri, 3 Nov 2006 01:13:24 +0100
>
>Hi everyone.
>
>I'm facing this problem when configuring a Ser as a proxy and a registrar.
>
>I have the Ser and a IP pbx: the pbx registers itself on the proxy as
>pbx@<pbx_ip> (Contact header).
>I had to manage INVITEs messages for incoming call that are going
>towards the Ser, sent by UACs in the network.
>
>INVITEs are coming in the form <telephone_number>@<ser_ip> (Req-URI):
>Ser had to decide if <telephone_number> is managed by the IP pbx (it has
>this information) and eventually, do a rewrite uri to
><telephone_number>@<pbx_ip> and then forward the message to pbx.
>
>So far, it's all ok.
>
>Unfortunately, i have to handle also this situation: the ip pbx is down.
>At the present, the INVITE is forwarded towards the pbx, the ip pbx does
>not respond and after 10 seconds the requesting UACs sends a CANCEL.
>This latency is not acceptable.
>I'm looking for a way to implement this:
>after the rewrite uri to <telephone_number>@<pbx_ip>, look if I have a
>contact in usrloc with <pbx_ip> (i.e. the pbx is still registered): if
>not, send immediately a response 4xx to UAC.
>The problem is that i'm not able to access the usrloc db from the
>routing script... any hint?
>
>Thanks a lot...
>
>
>
>_______________________________________________
>Serusers mailing list
>Serusers(a)lists.iptel.org
>http://lists.iptel.org/mailman/listinfo/serusers
_________________________________________________________________
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The one you attached, didnt have a semicolon. Anyway, try using a
fresh copy of a ready made ser.cfg.
Bests,
Andrey.
On 11/2/06, Kamal.Mann(a)t-systems.com <Kamal.Mann(a)t-systems.com> wrote:
> Yup I always do the same, Might be copy paste error. Its having semicolon & working fine here.
>
> Regards
> Kamal Mann
>
> -----Original Message-----
> From: Andrey Kuprianov [mailto:andrey.kouprianov@gmail.com]
> Sent: Thursday, November 02, 2006 10:34 AM
> To: serusers(a)iptel.org
> Subject: Re: [serusers]: trusted table(permissions module)
>
> Hi Kamal,
>
> Whenever you modified your ser.cfg did you restart SER after? Did you
> check your ser.cfg for errors using "ser -c" command? I found some
> errors in your ser.cfg code. For instance, inside the main route
> block, your "if", which processes INVITEs, is missing a semicolon at
> the end of the block.
>
> if (method=="INVITE")
> {
> # sl_send_reply("404", "INVITE ");
> route(3);
> break;
> } <----- missing semicolon here
>
>
> If you are comfortable with your current ser.cfg, that's ok, but I'd
> suggest you start with a fresh one, make your trusted table work, and
> then add a "REFER" processing, like in your currect ser.cfg. Just
> download one fresh ser.cfg from iptel.org site and add your
> allow_trusted() "if" clause there.
>
> Regards,
>
> Andrey.
>
>
> On 11/2/06, Kamal.Mann(a)t-systems.com <Kamal.Mann(a)t-systems.com> wrote:
> > Hi
> > Please find ser.cfg enclosed.
> >
> > Trusted table entry:
> > +---------------+-------+--------------+
> > | src_ip | proto | from_pattern |
> > +---------------+-------+--------------+
> > | 10.25.119.156 | any | ^sip:.*$ |
> > +---------------+-------+--------------+
> > 1 row in set (0.00 sec)
> >
> > Thanks in anticipation
> > Kamal Mann
> > -----Original Message-----
> > From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Andrey Kuprianov
> > Sent: Wednesday, November 01, 2006 7:03 PM
> > To: serusers(a)iptel.org
> > Subject: Re: [serusers]: trusted table(permissions module)
> >
> > Hi Kamal,
> >
> > I understand your setup. Nevertheless, it will be difficult to figure
> > out your problem, w/o your ser.cfg file at hand. Also, send a trusted
> > table query result (i.e. that row which contains entry for your
> > SIP-AS).
> >
> > Regards,
> >
> > Andrey.
> >
> > On 11/1/06, Kamal.Mann(a)t-systems.com <Kamal.Mann(a)t-systems.com> wrote:
> > > Hi All
> > > In my scenario I need ser to communicate with my SIP-AS. This SIP-AS is having an application xyz running on it. This XYZ sends an invite to URI "A(a)xcv.de" which is a subscriber in SER and this is in TO header of Invite msg & FROM header contains another URI test(a)xcv.de but this URI (test(a)xcv.de) isn't registered / subscribed to SER. I need SER trust all packets from SIP-AS ip. BUT SER is sending 407 reply of INVITE to XYZ application! IP in trusted table is of SIP-AS on top of which XYZ application is running.
> > >
> > > Thanks in anticipation
> > > Kamal Mann
> > >
> > > -----Original Message-----
> > > From: Maciej Żwirski [mailto:mzwirek@poczta.fm]
> > > Sent: Wednesday, November 01, 2006 3:20 PM
> > > To: serusers(a)lists.iptel.org
> > > Subject: Re: [serusers]: trusted table(permissions module)
> > >
> > > Kamal.Mann(a)t-systems.com wrote:
> > >
> > > > Hi All
> > > >
> > > > I made an entry /"10.25.119.156, any, ^sip:.*$"/ into trusted table so
> > > > that any packet sent from 10.25.119.156 would be trusted by SER and
> > > > don't ask for its credentials. But SER is replying *407- proxy* auth
> > > > needed!! After this I tried with permissions.allow file:
> > > >
> > > Hi,
> > > I had the same issue while interconnecting SER and Asterisk. It turned
> > > out to be an issue on Asterisk side (I had the same user added for SER
> > > and Asterisk, so the poor thing couldn't authorize either :)) So you
> > > could check if the 407 you're getting is from SER or from the remote host.
> > >
> > > Regards,
> > > Maciej Zwirski
> > >
> > > ----------------------------------------------------------------------
> > > Jestes kierowca? To poczytaj! >>> http://link.interia.pl/f199e
> > >
> > > _______________________________________________
> > > Serusers mailing list
> > > Serusers(a)lists.iptel.org
> > > http://lists.iptel.org/mailman/listinfo/serusers
> > > _______________________________________________
> > > Serusers mailing list
> > > Serusers(a)lists.iptel.org
> > > http://lists.iptel.org/mailman/listinfo/serusers
> > >
> >
> >
> >
>
Hello,
does anybody knows the state of T.38 with MediaProxy or
RTPProxy/NATHelper? Is it supported in MediaProxy v1.7.2 or the current
RTPProxy(NATHelper release?
Regards
Bastian
