Hey guys,
I have a current setup like this .
PSTN user calls in to SIP boxes via PSTN Gateways
Gateway will send the call to the OPENSER servers and in turn rings the
SIP UA
IF SIP UA tries to forward it out to the PSTN again , the problem i am
getting now is NO ring back on the CALLER . (called party (PSTN) rings
and has no problem starting conversation . Just that no ring back).
I am currently making use of the UAC_Redirect in the openser server to
route based on the 3xx move temporarily message.
Is there a way to send a provisional message to the calling party PSTN
gateway (originating GW) that it is trying / ringing so that the
ORIGINATING GW will generate ring back to the caller ?
Please let me know if you can't understand . THanks for your help guys!
Regards,
Sam
Hi,
I have installed openser 1.2.0 in my federo, however I do not know how to use a tool to add users to database.
Any hint will be appreciated.
Thanks
kerry
Hi
I have two questions.
1. Receiver(a)1.1.1.1 and Receiver(a)2.2.2.2 register with ser.
Client sends an INVITE to ser for Receiver.
ser forks these INVITES in parallel to both Receiver(a)1.1.1.1 and
Receiver(a)2.2.2.2.
As both Receivers are online, they both reply with 200 OK.
SER then relays BOTH 200 OK's back to the Client.
My question: Shouldn't SER send CANCEL to one of the receivers, and
relay only ONE 200 OK back to the client? How would I do this?
It is not the client's responsibility to send a CANCEL to the SER to
relay to one of the receivers, is this correct? This is because the
client only sent one invite, therefore it should only receive one OK. ie
should SER 'shield' the client from the knowledge of the existence of
multiple contact addresses?
2. When I have
tls_verify_client = 1
tls_require_client_certificate = 1
in my cfg file, ser still accepts tcp and udp connections, despite
tls_require_client_certificate=1. I did a search, and found this way:
if (proto != TLS) {
sl_send_reply("403", "Forbidden");
exit;
};
However SER is really still listening on UDP and TCP port 5060. Is there
a way to configure SER such that it only listens and acknowledges TLS
connections?
Thanks
Andrew
Hello
Is there is CLI (Command Line Interface) that enable a user to read or
eventually to change parameters at run time in OpenSER?
Is there a GUI (web, php,...) that enable configuration?
Thanks.
Regards,
Michel.
Hi,
Who's in charge of the agenda in Prague? (Or do we have somebody in
charge? ;-)
The presenters need to be allowed time to prepare for presentations etc.
And we should decide on the length and the format of the meeting. I hope
we can avoid a "death-by-powerpoint" (yes, I include openoffice in this
category) style and rather have short intros and discussions :-)
g-)
Dear All,
I am experimenting with OpenSER on an IPv6 environment. The pdt module
appears not to support using an IPv6 address as destination domain (I
guess that is true of source domain too, but I do not need that
anyway).
I tried to append a pdt row to MySQL using the documented OpenSER FIFO
interface. The message I sent to the FIFO is similar to
:pdt_add:openser_tmp_reply
voip.mycompany.com
9
[::ffff:abcd:1234]
However, when the row is created in MySQL pdt, the [:: is apparently
missing. I guess that is because the FIFO parser mistook that as the
FIFO :-separated message header or something similar?
Testing with an IPv4 address instead as the destination domain, the row
can be created properly, and prefix2domain() worked flawlessly in an
IPv4 environment.
I would like to use IPv4-mapped IPv6 address to forward to an Asterisk
PABX over SIP+IPv4 as Asterisk has no official IPv6 release yet (this is
another question, whether OpenSER will handle v4-mapped address
properly). This is what I would like to test by setting this pdt rule.
I'm using OpenSER 1.2.0-dev19-notls on i386/linux, by the way.
Thanks for any invaluable insights.
Regards,
Bernard Chan.
Hi,
We are happy to announce the release of SEMS 0.10.0 rc2. This release
has a number of bugfixes and stability improvements, but also some
interesting new application and component modules like PIN collect for
scalable conference systems, a new flexible way to script SEMS in
Python, SIP registrar client and UACi authentication, and a new method
to handle jitter.
Check it out at http://iptel.org/sems .
From the Changelog
- new Adaptive jitter buffer as alternative playout method
Contributed by Andriy Pylypenko/Sippy Software
- new PIN collect application with transfer to e.g.
separate conference bridge
- new SIP registrar client for registration at a
SIP registrar
- new UAC authentication component
- new faster announcement application with memory caching for
audio files
- new pre call announcement method using REFER
- stats server can be used for monitoring custom modules/applications
- session specific parameters by default taken from unified
session parameters header
- signature configurable
- install and make system updated
- added documentation
- some security bugfixes (namely fixing possible
buffer overflows)
- ...and a lot of other bug fixes
Thanks go to Andriy and Maxim for the adaptive jitter buffer, Juha,
Alexandr, Alex and all the others for bug reports and contributions.
Regards
Stefan for the semsdev team
--
Stefan Sayer
Media Services Development
stefan.sayer(a)iptego.de
www.iptego.de
iptego GmbH
Am Borsigturm 40
13507 Berlin
Germany
Amtsgericht Charlottenburg, HRB 101010
Geschaeftsfuehrer: Alexander Hoffmann
Dear all,
Would appreciate your advice from the ser-0.9.6 program, if save.c is the program in the SER that reply the 200 OK message for the REGISTER sends from the client?
Thanks.
Regards,
Howard
This email (including any attachment) is subject to the following disclaimer:
http://m1.com.sg/M1/misc/disclaimer
Hello,
I try to use this code :
if (!radius_is_user_in("From", "voip"))
{
sl_send_reply("403", "Forbidden");
exit;
};
Radius server return this :
Reply-Message = "Authorized"
but in openser in debug mode there is :
0(12528) radius_is_user_in(): Failure
Thank for your help.
It is possible to have the SIP provider route calls to an IP address. This
will alleviate the need to register with the provider.
Using a Linksys device may not be the answer either. The Linksys device can
register with openser, but there are limitations on the handling of inbound
calls. The linksys device assumes that the TO Header is the account name.
There is a way to route the inbound DNID using the dial plan. You can
contact Linksys support, they should be able to tell you. I would if I
remembered how.
For asterisk we modified the code (very simple change) to allow Openser to
route calls to asterisk and still use the users peer definition. The change
is a very small change 3 lines of code, that works with both 1.2 and 1.4
branch of Asterisk. We can share this with you.
Gene Willingham
Telasip
-----Original Message-----
Message: 1
Date: Tue, 6 Mar 2007 14:32:56 -0600
From: "Scott Yagel" <syagel(a)packetcall.net>
Subject: [Users] Openser and SIP provider
To: <users(a)openser.org>
Message-ID: <000901c7602e$9fefbf40$4b67a8c0@syagel>
Content-Type: text/plain; charset="us-ascii"
Hello,
Am I correct in my assumption that I can't directly connect Openser to a
ITSP due to Openser not providing registration to the ITSP? This seems to
be what I read in the Openser documentation. This being the case, I have a
Linksys SPA9000 pbx that I can use as a gateway to the ITSP, but attempts to
route 10-digit numbers to the pbx get a response of 403 Forbidden from the
pbx (the pbx has registered to the Openser OK). I see that many use an
asterisk for a gateway, how do they get around this problem with the
asterisk?
Thanks,
Scott Yagel
PacketCall, Inc.
syagel(a)packetcall.net
