All very good points. I've largely been able to avoid NAT up until now, so
I'm afraid I'm a bit too green to offer any clarification.
- Jeff
On 9/8/09 1:08 AM, Thomas Gelf tho...@gelf.net wrote:
Thomas Gelf wrote:
Jeff Pyle wrote:
if (client_nat_test(3)) {
force_rport();
Hello,
Could anybody tell me what is the problem with such a configuration:
---
(UAC)192.168.10.1 |=== |192.168.10.10 [OpenSIPS]
192.168.20.20|=== |(UAC)192.168.20.1 |
---
How are you making the packet captures? If tcpdump, did you use -s0?
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I'm using Wireshark
Saúl Ibarra sag...@gmail.com a écrit :
How are you making the packet captures? If tcpdump, did you use -s0?
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The mysterious point is that all packets orginating from the interface
(192.168.20.20) towards UAC(192.168.20.1) have Bad checksum in the UDP
level. This happened with/without RTPproxy, so i'm wondering whether I
should care about this problem or not because the sound is conveyed
and
Dan,
Unfortunately I'm not finding any references to escapaing characters in
Freeradius. I realize this isn't your home turf, but having said that, do
you have any pointers?
Thanks,
Jeff
On 9/7/09 8:02 AM, Dan Pascu d...@ag-projects.com wrote:
On 5 Sep 2009, at 23:01, Jeff Pyle wrote:
2009/9/8 ghaith.alkay...@telecom-bretagne.eu:
The mysterious point is that all packets orginating from the interface
(192.168.20.20) towards UAC(192.168.20.1) have Bad checksum in the UDP
level.
If you're capturing at the machine that has the 192.168.20.20
interface, then everything is ok.
I have been experimenting with Radius authentication on an older version of
OpenSIPS (actually my test
unit has OpenSER v1.3.2, but I don't think it will be much different to 1.5.x
for this). I want to be
able to read several values returned from the Radius server. Is this what the
modparam
Hi John,
The new AAA/RADIUS support is present only in 1.6 (current devel
version). Options are:
1) see if the old RADIUS support does not fit your needs somehow - use
opensips 1.5
2) go for the new AAA/RADIUS support and use 1.6 (even if devel, you can
take the chance to use it - there are
Hi Julien,
It most of the cases, your approach is correct , but to be 100% safe,
use remove_credentials() instead of remove_hf() - remove_credentials
will remove only the credentials/headers that were using on the local
authentication, which is useful when a requests carries more than one
Hi Ashwini,
I do not know about the Call-Control module too much, but can you also
push to it the new FROM value from the script ? or you cannot influence
the info that is sent to CC module ?
Regards,
Bogdan
ASHWINI NAIDU wrote:
Hi Bogdan,
The whole scenario is i want the new
Hi Ashwini,
I see you point - the question is how flexible is the Call Control
module in order to allow you to customize the billing account (and not
to use all the time the FROM URI).
Maybe somebody involved in this module devel can answer you.
Regards,
Bogdan
ASHWINI NAIDU wrote:
Hi
Hi,
I want to use the new AAA/Radius module, but it is for a business critical
service so am reluctant to
move to release 1.6 if it is not quite bedded in yet. What choices are
available?
John Quick
Smartvox Limited
Web: www.smartvox.co.uk
Smartvox is a limited company, registered in England
Hi Uwe,
Uwe Kastens wrote:
Hi Bogdan,
Seems that my question was not very clear.
I would expect that reply messages would be handled automatically, if I
use t_relay.
whatever forwarding function you are using (forward / t_relay), the
replies are automatically routed back by opensips core
Hello,
What impact does authentication have on the b2bua modules? When I put
b2b_init_request(top hiding) into my otherwise functioning script, very
strange things happen.
In order to achieve topology hiding, is it as simple as inserting this init
line at some point, or is there more to it?
-
Il giorno 22/lug/09, alle ore 11:09, Bogdan-Andrei Iancu ha scritto:
Hi Carlo,
you must set the line :
modparam(auth_db, load_credentials, $avp(s:rpid)=rpid)
This line instructs opensips to load at db auth time tht rpid
field into the $avp(s:rpid) variable.
Hi Bogdan,
Last july I
Hello John,
On Tue, Sep 8, 2009 at 5:43 PM, John Quick john.qu...@smartvox.co.ukwrote:
I have been experimenting with Radius authentication on an older version of
OpenSIPS (actually my test
unit has OpenSER v1.3.2, but I don't think it will be much different to
1.5.x for this). I want to be
Hi Bogdan,
I need to route this
replys with an reply_route and forward them explicitly to the pstn gateway.
and how do you do reply routing ?? replies are automatically routed
based on VIA stack and you cannot influence this from script.
...
if (!t_relay()) {
sl_reply_error();
};
Hi Lars,
Any guidance would be appreciated from the community that has done
this already. I installed current revision.
Could be helpfull to know what you want to do with opensips :-)
BR
Uwe
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Replies are automatically routed only if they are statefully routed.
Uwe Kastens wrote:
Hi Bogdan,
I need to route this
replys with an reply_route and forward them explicitly to the pstn gateway.
and how do you do reply routing ?? replies are automatically routed
based on VIA stack
Hi,
Replies are automatically routed only if they are statefully routed.
Statefull = t_relay() ?
BR
Uwe
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Uwe Kastens wrote:
Replies are automatically routed only if they are statefully routed.
Statefull = t_relay() ?
Yes.
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Uwe Kastens wrote:
Hi,
Replies are automatically routed only if they are statefully routed.
Statefull = t_relay() ?
Keep in mind, you can't really mix stateful and stateless handling for
what are hopefully obvious reasons. If you statelessly direct a reply,
the stateful transaction
Hello,
Ok. I need to have that forward() in my configuration the get answers
like 404, 486, 487 back to my asterisk. Reading your statements this
should not be possible since I use t_relay for the requests and the
replys should be routed by default.
I will make a trace and post it to the list.
Uwe,
forward() is a function exclusivly used for REQUESTS - for replies,
nothing needs to be done as OpenSIPS will do it automatically:
1) if the requests was statefully forwarded (via t_relay() ), the
transaction will contain all the info to route back the reply
2) if the requests was
Hi,
you do not need to do any routing in onreply route at all, in none of
the case (stateless or statefull)
I will make a trace and post it to the list. One with forward and the
other without.
make a trace and opensips logs.
I have attached opensips.log with debug=9.
Thank you for the response UK. We are looking at using OpenSIPS on Linux to act
as a SIP Router to handoff SIP calls to other carriers that are also using SIP,
and explore the use of the agent registration server functionality. I have
already previously installed and debugged Asterisk on CentOS
Hi,
Opensips is a SIP router not a media gateway. So far you will need
something that will take care of comvert TDM/PSTN to sip.
There should be lot of examples for this kind of setup.
BR
Uwe
Kemp, Larry schrieb:
Thank you for the response UK. We are looking at using OpenSIPS on Linux
Bogdan-Andrei Iancu wrote:
2) if the requests was statelessly forwarded (via forward() ), the VIA
stack (in received reply) will contain all the info to route back the reply
I think the question is whether stateless forwarding can be used to
override default processing of Via and route the
Hi,
Good question and not easy to answer. ACME is expensive AND you will
need somebody to configure it in a way you will need it. So as an
redundant option your talking about 100-150K.
To buy a big name won't prevent you from implementing, bugsearching.
My personal opinion: Take less money,
Can OpenSIPS be used as a Session Border controller sitting at my edge passing
and receiving SIP traffic to others I SIP peer with? If not, what other
open-source would anyone suggest to act as SBC's? I too would rather do it via
open-source and x86 or 64bit chip, less costly. Thanks.
Lars
Kemp, Larry wrote:
Can OpenSIPS be used as a Session Border controller sitting at my edge
passing and receiving SIP traffic to others I SIP peer with? If not, what
other open-source would anyone suggest to act as SBC's? I too would rather do
it via open-source and x86 or 64bit chip, less
Personally I prefer the Sonus GSX9000 gear for a PSTN gateway. I even put
one in my kid's room.
- Jeff
On 9/8/09 1:30 PM, Uwe Kastens ki...@kiste.org wrote:
Hi,
Yes, this could be an option. But a very expensive one :-)
BR
Uwe
Kemp, Larry schrieb:
So I would use OpenSIPS
Hi,
Yes, this could be an option. But a very expensive one :-)
BR
Uwe
Kemp, Larry schrieb:
So I would use OpenSIPS behind say like an Acme Packet
http://www.acmepacket.com/ Session Border Controller or a MetaSwitch
http://www.metaswitch.com/ connecting to the PSTN, then use OpenSIPS to
Certainly. If I just wanted to pass my SIP to other carriers or have them
connect to my SIP customers could I use OpenSIPS for that alone, or would I
still need some other sort of session border controller?
Larry Kemp
-Original Message-
From: users-boun...@lists.opensips.org
First time compiling OpenSIPS; installing on CentOS. Downloaded source as well
as RPM's from http://centos.leurent.eu. I was wondering if any of you that have
already done this successfully and had any resources or notes from your
deployment that you might be willing to share, detailed how-to
Hello everybody, dose anyone know where the log file is?___
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Hi Bogdan,
That problem is solved by using the diverter_avp modparam in call control
module. thanks for replying
On Tue, Sep 8, 2009 at 8:37 PM, Bogdan-Andrei Iancu
bog...@voice-system.rowrote:
Hi Ashwini,
I do not know about the Call-Control module too much, but can you also
push to it
By default the logging of opensips will be done in */var/log/syslog* in
debian systems and */var/log/messages* in redhat based systems
2009/9/9 zhangchao1 zhangchao...@163.com
Hello everybody, dose anyone know where the log file is?
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