That's why I dropped the ideea of having numbered usernames…
On Feb 11, 2011, at 10:45 PM, Dave Singer wrote:
> Adrian,
>
> Probably want to only respond to registers that are to valid user
> accounts, drop the rest, as they start scanning with like 100, 101,
> ., 5000, etc
>
> Dave
>
What is the easiest way to identify traffic with invalid headers?
Specifically, the from and to URIs.
For example, if OpenSIPS is unable to parse a from URI, would $fu be NULL?
Thanks.
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips
Adrian,
Probably want to only respond to registers that are to valid user
accounts, drop the rest, as they start scanning with like 100, 101,
., 5000, etc
Dave
On Fri, Feb 11, 2011 at 6:25 AM, Adrian Vasile wrote:
> Hi Dave,
>
> Yeah, you're right.. Basically allow only REGISTER reques
I have a similar problem, but not solution, my probably is actually occurring
because the originating UA is ignoring a contact header that is sent back
during a 183 progress message. OpenSIPS uses information from that contact
header to figure out where to relay the incoming message (BYE in my
You are right.
I just escalated the scenario to de...@rtpproxy.org
Thank you.
On 11 February 2011 19:15, Ovidiu Sas wrote:
> Please report your crash on the rtpproxy list and provide a way to
> reproduce it.
> Rtpproxy should not crash that easy.
>
> Regards,
> Ovidiu Sas
>
> On Fri, Feb 11
Hi Dave,
Yeah, you're right.. Basically allow only REGISTER requests from anywhere and
the rest check the source ip.
Great ideea.
I will implement it as soon as possible.
Thanks,
Adrian Vasile
y...@opennet.ro
On Feb 10, 2011, at 10:41 PM, Dave Singer wrote:
> Adrian,
>
> I was just thinkin
Please report your crash on the rtpproxy list and provide a way to reproduce it.
Rtpproxy should not crash that easy.
Regards,
Ovidiu Sas
On Fri, Feb 11, 2011 at 12:04 PM, Kamen Petrov wrote:
> Hi Anca,
>
> Ok, I managed it work your way.
>
> The key was not in the rtpproxy_answer but the rtppro
Hi Anca,
Ok, I managed it work your way.
The key was not in the rtpproxy_answer but the rtpproxy_offer :)
Once again thanks to you and Ovidiu for your great help !
So just for the record if someone else face the same issue: segfault in the
rtpproxy on the onreply_route: don't look only the rtpp
On 02/11/2011 03:31 PM, Kamen Petrov wrote:
/onreply_route[1] {
if (!(status=~"183" || status=~"200")) {
drop;
}
rtpproxy_answer("FA");
/
Maybe you could try to use other flags, or renounce at one at a time to
see which one results in segmentation fault
Hi Cris,
On 02/09/2011 02:35 PM, chris wrote:
Want to play back an in call announcement using rtpproxy. This is
available in rtpproxy itself and is accessible through the rtpproxy
module for kamailio but doesn’t seem to be available in the opensips
nathelper implementation.
It is in OpenSIP
And just a followup from what Klaus mentioned here is a link from the
OpenSIPS tutorial page on how you can set up Presence
http://www.opensips.org/Resources/DocsPapPa
On Fri, Feb 11, 2011 at 8:46 AM, Klaus Darilion <
klaus.mailingli...@pernau.at> wrote:
> What kind of presence do you use (confi
What kind of presence do you use (configuration option in xlite)?
end-to-end: that should work out of the box
presence-agent: opensips must be configured as presence server, probably
with proper xcap authorization rules (or disable them)
klaus
Am 01.02.2011 00:07, schrieb ViennaCivicEP2:
>
> H
Also, this is how I am running the rtpproxy:
23414 ?Ss 0:00 /usr/local/bin/rtpproxy -s udp:184.106.168.144
22332 -u root root -p /var/run/rtpproxy/rtpproxy.pid -F -l 184.106.168.144
And here is the nathelper config for both opensips and b2b:
modparam("nathelper", "rtpproxy_sock", "udp:
Anca:
*> There was a problem with the db schema for the b2b_logic table - lots of
wrong NOT NULL constraints there. I have just fixed it. Please take the new
schema from svn and replace the table.*
-- Seems to be fine now, thank you.
*> Are you using the newest version of rtpproxy?*
-- I am runnin
http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-starting-Error-td5994344.html#a5994453
http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-starting-Error-td5994344.html
On Fri, Feb 4, 2011 at 5:08 PM, Venkatesh N wrote:
> When I enter
>
> opensipsctl start
>
>
> INFO:
Then please remove the old core file and make sure that you have the
latest source on both servers.
On Fri, Feb 11, 2011 at 9:27 AM, Kamen Petrov wrote:
> The last core i have is:
> -rw--- 1 root root 43188224 Feb 10 11:49 /core
>
> I did the attached tests 1 or 2 hours ago and the system tim
The last core i have is:
-rw--- 1 root root 43188224 Feb 10 11:49 /core
I did the attached tests 1 or 2 hours ago and the system time now is "Fri
Feb 11 14:29:14 UTC 2011".
I guess there is no new core :(
On 11 February 2011 16:23, Ovidiu Sas wrote:
> Please get a gdb trace from the core
Hi Kamen,
On 02/11/2011 03:31 PM, Kamen Petrov wrote:
Feb 11 12:49:06 sms /root/opensips-1.6.4-tls/opensips[21621]:
ERROR:db_postgres:db_postgres_store_result: 0x7b9360: ERROR: null
value in column "e3_sid" violates not-null constraint#012
There was a problem with the db schema fo
Please get a gdb trace from the core file.
Thanks,
Ovidiu
On Fri, Feb 11, 2011 at 8:31 AM, Kamen Petrov wrote:
>> Ok guys,
>>
>> Few issues still (after updating from trunk).
>>
>> As suggested, I removed the engage_rtp_proxy from the b2b opensips
>> instance.
>>
>> I noticed the following debug
Just an update on this: it's ridiculously hard.
We've done some major surgery on the route logic, and at this point I have the
strange condition where opensips seems to be sending multiple ACKs to the
carrier on a single reINVITE. The carrier should be sending us two invites -
one for each leg
When I enter
opensipsctl start
INFO: Starting OpenSIPS :
ERROR: PID file /var/run/opensips.pid does not exist -- OpenSIPS start
failed
I checked /var/log/messages and got following
Feb 4 18:02:23 ubuntu kernel: [ 6747.275248] intel ips :00:1f.6: CPU
power or thermal limit exceeded
Feb
Will do Dave - thanks for following up!
Sent from my iPhone 4
On Feb 4, 2011, at 15:57, Dave Singer wrote:
> Tyler,
>
> Just went through the OpenSIPS default script webminar =>
> http://www.opensips.org/html/docs/video/webinar005/
> And while the audio at the beginning is bad (and very end),
Guys I a newbie to OpenSIPS
I have installed opensips and mysql on ubuntu following some instructions.
I have also installed x-lite. Now how to register a user in opensips and to
use it with the client ? I am stuck, please let me know
Regards
Ricky
___
What are you trying to do ?
On Wed, Feb 2, 2011 at 1:28 PM, Robin Malhotra wrote:
> Step 3: Create MySQL tables using the opensipsdbctl shell script. The
> syntax for
> this utility follows:
>
> opensipsdbctl create
>
>
>
> I'm getting the following error for the above syntax
>
> bash: syntax e
It's very simple setup a Conference server using OpenSIPS and Asterisk.
So use asterisk.
Regards,
s
Il 27/01/2011 17:39, Anca Vamanu ha scritto:
Toyima,
I am sorry, I don't have experience in setting up conference systems,
so I can not make a recommendation.
Regards,
___
Hi,
i´m new to the Opensips community. I started a few days ago and i´m now at
the point to post my first question, because i cant fiddle out my mistake in
configuration.
This is what i´ve done so far.
- Setting up 3 Virtual Machines (1x Debian Lenny Server, 2x Windows XP Host
with X-Lite Client
Hi All,
Need help with a nagging issue:
UA->Opensips 1->Opensips 2->PSTN
UA sends an invite on Opensips 1, and is routed via do_routing() to Opensips
2, Opensips 2 uses do_routing to get to the PSTN, call starts ringing.
UA cancels call before answer, but now t_check_trans fails and
>
> Ok guys,
>
> Few issues still (after updating from trunk).
>
> As suggested, I removed the engage_rtp_proxy from the b2b opensips
> instance.
>
> I noticed the following debug from the opensips:
> Feb 11 12:49:06 sms /root/opensips-1.6.4-tls/opensips[21621]:
> ERROR:db_postgres:db_postgres_stor
COOL Adrian...many thanks for your kindly answers...by the way, i've checked
on the rfc that the client must use NAPTR and SRV to resolve domains!
2011/2/11 Adrian Georgescu
> If your SIP device support dialing only phone numbers, you need a
> translation mechanism, this you can implement in the
If your SIP device support dialing only phone numbers, you need a translation
mechanism, this you can implement in the SIP proxy. You can use standard ENUM
(http://www.faqs.org/rfc/rfc3764.txt), local database lookups, configuration
logic to translate the number into a fully qualified SIP addre
create...got it...that means that if i have a phone registered in proxy A,
and i want to call userB, A has no idea where B resides, at all...how does A
know the domain of B? he must put in the RURI of the invite
userB@domain_of_b, right? how does A knows the domain of B? does A must
press in the ph
SIP routing works exactly like email. How did you know to email this list?
Adrian
On Feb 11, 2011, at 1:42 PM, Toyima Dias wrote:
> Thanks Adrian...
> So...how does ALice now that bob is in the biloxi.com domain? per the rfc
> 3263 section 4 (client usage) the ua must use DNS to determine wher
Thanks Adrian...
So...how does ALice now that bob is in the biloxi.com domain? per the rfc
3263 section 4 (client usage) the ua must use DNS to determine where to send
a call...but i have a softphone righ now, and i'm trying to make a call like
this:
234...@proxy2.com (inserted by me), not just pu
The SIP proxy lookups up the domain part, what appears after the @ sign before
any parameters separated by ; if is an IP address like in your example you do
not perform a DNS lookup you just send the packet there.
In the request URI you must put the address of the remote end, not your own
addr
Adrian, i'm checking the rfc...but even i have a question...when UA sends an
INVITE to it's proxy to a phone for example (obviously not registered on the
proxy), the proxy will check the RURI of this invite and it will se the
following:
user A sends the invite to its proxy : INVITE
sip:26451238097
Thanks Adrian...reading the RFC3263!
Thanks!
2011/2/11 Adrian Georgescu
> The proxy is using DNS to lookup the destination server.
>
> Google for RFC 3263
>
> Adrian
>
>
> On Feb 11, 2011, at 10:19 AM, Toyima Dias wrote:
>
> Hello community,
>
> I have a doubt, how does a SIP Proxy (OpenSI
You're right my friend...the problem is with the softphones and the OS of
the machine...it's ok now!
Thanks!
2011/2/10 Anca Vamanu
> Hi Toyima,
>
> That when you un-Register and the phone sends expires=0 you get that reply
> with contact and expires is correct, because of what you already had
Hi All,
Need help with a nagging issue:
UA->Opensips 1->Opensips 2->PSTN
UA sends an invite on Opensips 1, and is routed via do_routing() to Opensips
2, Opensips 2 uses do_routing to get to the PSTN, call starts ringing.
UA cancels call before answer, but now t_check_trans fails and
Good to hear that!
Cheers,
Henk
On 11-02-11 02:15, Chris Stone wrote:
Well, looks like it WAS the ip_nat_sip and related kernel modules, but
not just on the Opensips server, also on the Asterisk server. I
unloaded all of the modules on the backend Asterisk server too and
tried a test call aga
The proxy is using DNS to lookup the destination server.
Google for RFC 3263
Adrian
On Feb 11, 2011, at 10:19 AM, Toyima Dias wrote:
> Hello community,
>
> I have a doubt, how does a SIP Proxy (OpenSIPS) would handle a call for a
> domain that he doesn't now? i mean...user A is registered i
Hello community,
I have a doubt, how does a SIP Proxy (OpenSIPS) would handle a call for a
domain that he doesn't now? i mean...user A is registered in proxy AA, if A
wants to call to another user in another domain (not registered in the Proxy
AA) how does this proxy should handle the call? how do
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