Hi @all,
I have following problem. I have registered all my phones on OpenSER
with the full number. e.g. 43 11555 1007
When one employee from one site wants to call an employee from the same
site I want them only to dial 1007 not
The whole number. I have following perl script, but when I use the
"canonical" function, it always adds "+" in front of the number.
I get following message:
canonical number in 'sip:1777@server.com' is '+43115551007'
After the execution of the perl script I want to check, if the user with
43115551007:
It looks like this:
if (uri=~"sip:[0-9]+@.*")
{
perl_exec("canonical");
if (does_uri_exist())
{
route(2);
} else {
#route to LCR-Module
route(3);
};
}
But it does not run well.
The script looks like this:
-------------------------------------------------
use OpenSER::Constants;
use OpenSER::Utils::PhoneNumbers;
sub canonical {
my $m = shift;
if ($m->getMethod() eq "INVITE") {
my $p = new
OpenSER::Utils::PhoneNumbers(publicAccessPrefix => "",
internationalPrefix => "",
longDistancePrefix => "",
countryCode =>
"43",
areaCode =>
"11555",
pbxCode => "");
my $u = $m->getRURI();
if ($u =~ m/(.*)sip:([+0-9]+)\@(.*)/) {
my $c = $p->canonicalForm($2);
OpenSER::log(L_INFO, "canonical number in '$u'
is '$c'\n");
} else {
OpenSER::log(L_INFO, "Not a POTS number.\n");
}
}
return 1;
}
Can anyone provide me some help?
Thanks
martin
hi:
I want to get a sip proxy that support Call , MESSAGE , NOTIFY ,SUBSCRIBE.
When the destination ua is offline the message is storing.
The storing message is send when it is online.
The two ua can call each other.
What statble version i can use?
How can i config the ser.cfg?
Thanks very much !
2007-09-25
程千
Hello
Im trying to implement call forwarding unconditional redirect to a PSTN
destination.
Call flow: PSTN-->SIP-CallFW-->PSTN
On my PSTN side I have a cisco AS5300. We succesfuly forwarded to a PSTN
destination.
We have changed calling party (for billing purposes) and called party to
complete call forwarding. In other words we generate a new call attempt.
Everything looks ok but if calling party (PSTN originating side) finishes
the call before called party answers (PSTN destination side), cisco GW
receives a CANCEL message and it answers with message 481 Call
leg/Transaction does not exist to the Proxy.
The call still ringing and goes to voice mail on PSTN destination side that
means cisco GW never receives a BYE regarding to the original INVITE
message.
Can you give me a hint on how to handle this issue ?
Thanks
Captured attached
Ariadne L. Ramos Solís
Depto. Ingeniería
Aprovisionamiento y Señalización VoIP
Galaxy Communications Corp.
Tel. 2000117
e-mail aramos(a)clarocom.com
I'm looking for someone who can assist with an OpenSER/MySQL/Radius
setup which needs to perform LCR from MySQL, write proper CDRs (and
fail calls if they can not be written), and allow for routing a
specific group of inbound IPs to specific gateways. Essentially, I'm
looking to create a multi-tenant session border controller for
wholesale VoIP to VoIP.
If you have experience in this and are interested in doing hourly
contracting work, both to assist with the initial setup as well as
ongoing support, please contact me.
Daryl G. Jurbala
Director of Network Operations
Global Convergence Solutions
Phone: +1 732-853-0513
Hi!
Openser 1.2: I put some routing info into the record-route header. If a
transaction gets redirected to another destination in failure route, I
have to adopt the parameter and add a record-route header with the new
parameter.
Currently, I do add_rr_param and record_route again in failure route,
but this leads to 2 Record-Route headers in the outgoing request. Is
there a way to reset RR in failure route - or am I just doing something
wrong?
thanks
klaus
Hi,
I m facing the same problem while configuring openxcap. I have installed
openxcap in FC4 Linux and also the prerequisites packages.
Can someone please let me know how to start the daemon. Thanks in
advance.
Thank you,
Regards,
Harini Dhanasekaran.
------------------------------------------------------------------------
------------------------------------------------------------------------
------------------------------------------------------------------------
--------------------------
KevinKinnan kinnan2224 at hotmail.com
<mailto:users%40openser.org?Subject=%5BOpenSER-Users%5D%20OpenXCAP%20ins
tall%20and%20config%20problem%3F&In-Reply-To=> wrote:
Thu Sep 6 17:47:14 CEST 2007
I am not sure whether or not my procedure is right as server doesnt work
at the moment, but I will keep working on it.
Here is the Fedora installation
[1] Fedora 6 linux install;
[2] pre-dependencies package;
Because I installed OpenIMSCore and Openser Presence module before, so
some dependencies was already installed.
In this case I just installed clearsilver from Add&Remove Package, and
copy lxml folder(1.3.4 download from lxml site)
to /usr/lib/python2.5/site-packages/.
[3] install OpenXCAP, followed installation guide/Debain unstable
installation
By now, that's it. Coz no daemon can be run from /etc/init.d/openxcap.
Also, I haven't modified anything in the config.ini
and openxcap files
Cheers,
Kevin
The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments.
WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email.
www.wipro.com
Hi Everyone
I have tested openser 1.1 with the Mysql module and now I want to test
1.2with mysql, rtpproxy and the dispatcher module. When I'm compiling
openser
should I do anything differently? I saw rtpproxy listed in APT and the
dispatcher modules was in /usr/local/lib/openser/modules last time I had
openser 1.1 installed. should I only change my opernser.cfg in order for it
to load?
--
Thank You
forgot to cc the list ...
-------- Original-Nachricht --------
Betreff: Re: domain name in route header
Datum: Mon, 17 Sep 2007 18:58:33 +0200
Von: Klaus Darilion <klaus.mailinglists(a)pernau.at>
An: Juha Heinanen <jh(a)tutpro.com>
Referenzen: <18154.37441.743333.215104(a)tutpro.com>
Juha Heinanen schrieb:
> klaus,
>
> if you want to support domains in route headers, then you need to do
> naptr and srv lookups for the domain when handling loose routing. it is
> not enough to compare the domain to the ones domain module knows about.
> domain in route header may namely be anything as long as it resolves to
> your proxy. for this reason, i don't like the idea of domains in route
> headers. it hurts performance far too much.
I also do not like to configure openser all possible hostnames as alias
- but it is not uncommon that the SIP domain is in the pre-loaded route
- at least most exosip based clients do that. Thus, having support for
those clients is IMO useful.
regards
klaus