u.
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eled.
>
> Sent from Outlook for Android
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r. Still rtpengine is required?
>
>
> --
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> B.Prathibha
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gt; Any hint, please?
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> On Mar 15, 2022, at 10:13 AM, Saint Michael wrote:
>
> Every call that goes through Opensips should generate a record.
Is this just your opinion, or…?
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This is precisely due to the proportion of short-duration calls. But I agree
that it is a very stupid way to approach the problem, particularly when you
consider where most of the hangups in short duration calls actually come from.
Some of them do you come from the originator, in the case of AMD
b.com/sipwise/rtpengine.git
But still i dont have a clue on how to solve it, any advise?
Thank you
Mario
On Fri, Jul 17, 2020 at 2:34 PM Alex Balashov <mailto:abalas...@evaristesys.com>> wrote:
This happens when an SDP body that has already been passed to
RTPEngine, and alrea
This happens when an SDP body that has already been passed to RTPEngine, and
already adulterated by RTPEngine, is passed to it yet again.
Newer versions of RTPEngine have a loop protection feature to deal with it. It
involves injecting an unregistered a=rtpengine attribute into the SDP, to say
se
life in kernel space is pretty efficient. But async execution in
user-space requires user-space contrivances that suffer from all the
problems of user-space in turn, so the economics can be really
different. Mileage of course greatly varies with the implementation detai
hstanding page sizes and blocks in the underlying
allocator), but nothing on the order of gigabytes upon gigabytes.
Assuming 4 KB per call and 200,000 concurrent calls, that's ~800 MB, and
that is a very generous assumption indeed.
-- Alex
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this amounts to a "tuning" or "optimisation" that yields a "performance
increase" is profoundly misleading.
-- Alex
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, but that
the before and after don't lie.
On Fri, Jun 12, 2020 at 4:02 PM Alex Balashov
mailto:abalas...@evaristesys.com>> wrote:
But increasing the depth of the queue by 78x (if I'm not mistaken,
212992 is the default-
ope with their workload don't cause
non-trivial packet or connection queueing on the OS side.
-- Alex
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e you can't
process them fast enough isn't a solution to the problem of not
processing them fast enough. You're infinitely better off just
processing them faster.
-- Alex
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all my CentOS 7.x and 8.x
systems, which I guess also have "absolutely terrible sysctl defaults")
is faker than a Ponzi scheme. In some other contexts, this would be
called morally bankrupt and intellectually fraudulent. I guess here we
call it "mad dialer CPS" or whate
. would be a
great starting point.
Regards,
*Calvin Ellison*
Senior Voice Operations Engineer
calvin.elli...@voxox.com <mailto:calvin.elli...@voxox.com>
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│ ├─pickup
│ ├─qmgr
│ └─trivial-rewrite
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It will trigger a failure_route.
—
Sent from mobile, with due apologies for brevity and errors.
> On Mar 15, 2020, at 4:20 AM, johan wrote:
>
> How can I catch in the script that fr-timer has expired ?
>
> I need to be able to see this expiry as I would like to failover on this.
>
>
> BR,
>
On Mon, Aug 12, 2019 at 10:52:09AM +0200, Giovanni Maruzzelli wrote:
> https://xkcd.com/378/
<3
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Unfortunately, binding to the same port will not preserve TCP connection state.
On May 1, 2018 10:05:39 PM EDT, "Podrigal, Aron"
wrote:
>Hi everyone I hope you are having a good time @opensipsSummit I wish I
>would be able to attend.
>
>Can anyone suggest how to handle graceful restart of opens
...@host.mysql.azure.com/opensips")
>
>
>
> which fails
>
>
>
> Any ideas on a workaround on this?
>
>
>
> BR / Olle
>
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The simplest and lowest-hanging fruit here is profit margin protection. Don't
return routes from the LCR whose cost exceeds the sell rate. This will ensure
that client 1 is never routed out Trunk 2, period - if Trunk 1 is exhausted,
you don't complete the calls.
Then, add some logic to ensure t
request] {
># How can we arrive here right upon the receipt of the 302, not in
>onreply_route?
>}
>
>> On Sep 5, 2017, at 4:54 PM, Alex Balashov
>wrote:
>>
>> Yes, failure_route is the answer to all your objectives here. You can
>> intercept the 302, extract wh
Yes, failure_route is the answer to all your objectives here. You can
intercept the 302, extract what you want from it, create a new branch
and fork the call elsewhere.
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gt; WARNING:core:do_action: error in expression at
> /etc/opensips/opensips.cfg:602
>
> does anyone have any idea what is causing this error or if this flag is
> even being evaluated ?
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, presume that the IP and port
endpoints on both ends stay the same.
So, if you suddenly start sending media from another place and expecting
to receive it there likewise, that will not be considered to be part of
the same phone call.
-- Alex
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een it behave, yeah.
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My understanding is that this is a rather simple module without sophisticated
state componentry, and that it logs things immediately as received, in the same
iteration of message processing.
-- Alex
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Sent from my Google Nexus.
_
r on 200 OK of BYE?
Are you referring to the ACC module, or some other method of accounting?
:-)
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OpenSIPS didn't hang up anything.
On April 28, 2017 10:30:08 PM EDT, volga...@networklab.ca wrote:
>Hello Everyone,
>Why opensips hang up session with 408 on
>SIP;cause=480;text="NO_ANSWER". I expected b2bua will send to
>voicemail.
>What way possible fix it.
>
>Please see attached trace.
>
>vol
at 11:39:14PM -0400, Satish Patel wrote:
> after google found bunch of post where people suggesting use
> fix_nated_sdp() is that right approach ?
>
> On Mon, Apr 24, 2017 at 11:25 PM, Alex Balashov
> wrote:
> > Yes, but RTP can come from a different address than the signa
#x27;t public then
> media never work.
>
> c=IN IP4 192.168.1.8.
>
> It should be
>
> c=IN IP4
>
> On Mon, Apr 24, 2017 at 11:04 PM, Alex Balashov
> wrote:
> > Satish,
> >
> > When you say "origin public address", do you mean the externa
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How do you view Redis?
On April 4, 2017 4:03:31 AM EDT, Ziv Gabel wrote:
>Hi All,
>Is there any intention to add database support to the rtpengine module
>?
>The same as rtpproxy.
>
>*Ziv Gabel *l Professional services l CommuniTake Technologies Ltd.
>
>
>
>*M*: +972535265553 l *Skype*: ziv_gabe
Record-Route headers contain URIs, and like any SIP URI, they can contain a
port component. If that port component is omitted, 5060 is presumed.
On March 29, 2017 6:27:30 PM EDT, Satish Patel wrote:
>what is the use of port number in record-route?
>
>I am having major issue with that look like
If MediaProxy is a SIP endpoint, that would be news to me.
-- Alex
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UA layer where failure to adhere to them does not
adversely affect the proxy's ability to relay the message.
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On 10/21/2016 06:36 PM, Newlin, Ben wrote:
Not only that, but provisional responses (except 100 Trying) are
required to have a To tag [1]. So you would likely run into issues with
UAs if you start returning messages without them.
That is an astute point.
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result on subsequent messages.
Do you guys see any problem on removing the to-tag of all 1XX messages?
Thanks
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It can certainly be done in the onreply_route.
Conceptually, doing it there makes more sense. The failure_route is not
triggered by any particular SIP reply per se, but rather a branch failure event
on a transaction. That can be the result of a timeout (e.g. fr_timer) or
something else that do
#x27;t do any unnecessary work to parse the whole message if the first
line doesn't suggest it's SIP.
I wouldn't worry about this "premature optimisation". :-)
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an/listinfo/users
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S too.
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Users ma
uot;);
You can run msg_apply_changes() after calling fix_nated_contact(),
assuming it doesn't have any effects harmful to your cause:
http://kamailio.org/docs/modules/4.4.x/modules/textopsx.html#textopsx.f.msg_apply_changes
-- Alex
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The tunnel server would then
send this data to OpenSIPS over UDP.
This sounds substantially similar to a VPN, except without the benefit
of encryption.
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On 05/02/2016 05:08 PM, Dragomir Haralambiev wrote:
I have registration activity with Radius asin
So, why do you expect fragmentation from time to time as the OpenSIPS
memory manager allocates and frees SHM blocks?
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1447 Peachtree Street
On 05/02/2016 04:44 PM, Alex Balashov wrote:
On 05/02/2016 04:22 PM, Dragomir Haralambiev wrote:
Opensips has not routed any calls.
Has it done anything else, including passive registration activity or
any periodic database-bound synchronisation tasks?
Also, what about passively deflecting
On 05/02/2016 04:22 PM, Dragomir Haralambiev wrote:
Opensips has not routed any calls.
Has it done anything else, including passive registration activity or
any periodic database-bound synchronisation tasks?
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1447 Peachtree Street NE
cumstances of the
timeout in a failure_route.
-- Alex
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be included in an application-level
referral that might leave the scope).
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On 03/31/2016 02:49 PM, Travis Manson-Drake wrote:
I would be more than happy to send it to you privately if that's ok?
Of course!
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Tel: +1-800-250-5920 (toll-free)
same hardware as your SIP proxy, correct?
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otline: 520.545.0333
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Tel:
Hello Rodrigo,
If you're not going to use an outboard media relay, how do you propose
to "fix" media ports in the SDP offers & answers? What will you rewrite
them to if you have no RTP endpoint to give you a means of seeing where
media is coming from? :-)
-- Ale
On 03/25/2016 03:48 PM, Dragomir Haralambiev wrote:
unknown command , missing loadmodule?
That sounds like a typo in the config.
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On 03/21/2016 04:36 PM, Rodrigo Pimenta Carvalho wrote:
According to the documentation "...It can be an IP address, hostname or
network interface id".
So, can I do the following configuration?
listen=tcp:wlan0:5060
Why can't you just do exactly that?
--
Alex Balas
Travis,
rewriteuri() is a legacy core function that does not support PVs.
Have you considered ...?
# Option 1.
$rU = $fU;
$rd = $avp(variable);
# Option 2.
$ru = "sip:" + $fU + "@" + $avp(variable);
-- Alex
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What is the exact text of the error?
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Sent from my
Have you attempted to increase the shared memory allocation to OpenSIPS (-m CLI
option)?
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Web: http
Ali,
Is there any danger that you are calling rtpproxy_offer() twice, or
using rtpproxy_offer() in combination with fix_nated_sdp()[1]?
-- Alex
[1] http://www.opensips.org/html/docs/modules/2.1.x/nathelper.html#id293899
--
Alex Balashov | Principal | Evariste Systems LLC
303 Perimeter
Hello Aqs,
Why not strip the Route header instead of denying the request? That is
to say:
if(is_present_hf("Route"))
remove_hf("Route");
-- Alex
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On 02/11/2016 08:50 PM, surya wrote:
I am facing some performance issues that's why I was thinking about those.
Performance issues of what nature?
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.html
Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
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I'm pretty sure that only the SIP receiver threads create DB connections.
-- Alex
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Web:
x.x.x:5060
Process:: ID=9 PID=743941 Type=SIP receiver udp:x.x.x.x:5060
Process:: ID=10 PID=743942 Type=SIP receiver udp:x.x.x.x:5060
Process:: ID=11 PID=743943 Type=SIP receiver udp:x.x.x.x:5060
-- Alex
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A
ughts on this. Apologies, while I can read code,
I'm not a proper programmer to make a real patch.
Thanks,
Ross.
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On 02/04/2016 11:08 AM, Alex Balashov wrote:
Bogdan,
That's interesting. I had no idea OpenSIPS had a module specifically
devoted to fixing bad SIP that really shouldn't be fixed. :-)
s/a module/dialog module functionality/
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303
Bogdan,
That's interesting. I had no idea OpenSIPS had a module specifically
devoted to fixing bad SIP that really shouldn't be fixed. :-)
-- Alex
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ere's no reasonable way to work around it: the UA in question needs to be
fixed.
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as a globally unique
identifier over space and time unless overridden by method-specific
behavior. All SIP UAs must have a means to guarantee that the Call-
ID header fields they produce will not be inadvertently generated by
any other UA.
2) Is there some way to override this
.voice-system.ro <http://www.voice-system.ro>
>>
>>
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>
>
OpenSIPS is a SIP proxy. It has no imaginable relation to codecs or
anything audio-bearer related. It does not deal with RTP at all.
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On Feb 8, 2010, at 2:11 AM, "Live School" wrote:
Hi,
is it possible to use openSIPS as transcoder only ?
if yes, any link of con
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ly been an
> asshole and a bad name on your/development group beautiful solution. Thanks
> to Inaki I will be looking into something else to enable me to do what I
> want. FUCK YOU INAKI. great waste of my time. Now go fuck yourself. So long.
>
>
>
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pretty much what I am asking you experts.
>
>
>
>
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avp(s:rpid))
> returns TRUE and is 1234567890
>
> Is this a bug or a feature? In either case, is there a way to get
> OpenSIPs to only use the current DB value?
>
> Thanks!
>
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Rewrite Request URI ($ru) and t_relay().
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On Jan 1, 2010, at 1:40 AM, Ahmed Munir wrote:
> Hi,
>
> I want to forward an Access Number from OpenSIPS to Asterisk
> machine. Kindly advise how can i do that? Which modules/functions
> are use to forward by INVITE sectio
RFC, or is the default
> config mistreating this somehow? Should an exception be added for
> REGISTERs?
>
>
> Thanks,
> Jeff
>
>
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intain one contact per AoR. If there are other
> contact addresses for AoR not matching current registration, remove
> them. This mode ensures one contact per AoR (user).
Alas, we all have our crosses to bear. :-)
--
Alex Balashov - Principal
Evariste Systems
Web : http://www.evaris
ax=1
>
> is there any way to save the latest AOR and then delete the old AOR
>
>
>
>
> Thank you
> Ha`
>
>
>
>
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/
ls. You are
not entitled to request "status reports" as if you are a micro-manager
coming around to check on your cubicle flock.
--
Alex Balashov - Principal
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
___
>>
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>
>
--
Alex Balashov - Principal
Evariste Systems
Web : http://
rehension to entertain an idea more bad than this one! :-)
--
Alex Balashov - Principal
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
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Users mailing list
Users@lists.opens
EM TRANSACAO");
> exit;
> }
> }
> sl_send_reply("404","Not here");
> }
>
>
> -
SEMS is another option for a B2BUA, though all its prebuilt B2BUA apps
are not pure B2BUA but incorporate some sort of initial media-based
announcement premise. But you can use its Python or C++ API to create
one.
Another option is YATE?
--
Sent from mobile device
On Oct 23, 2009, at 8:07
t;
>> So, it's your trunk provider the one who refuses to do the REFER
>>
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>
--
Alex Balash
No.
--
Sent from mobile device
On Oct 21, 2009, at 9:34 AM, Uwe Kastens wrote:
> Hi,
>
> I have the following requirement:
>
> If a from tm generated cancel is answered with a 200 OK I want to
> send a
> BYE to the UAC.
>
> Is this possible?
>
> BR
>
> Uwe
>
> --
>
> kiste lat: 54.322684, lo
understand what I’m missing?
>
>
>
> Thanks in advanced
>
>
>
> Ariadne
>
>
>
> P BE CARBON CONSCIOUS. PLEASE CONSIDER OUR ENVIRONMENT BEFORE PRINTING
> THIS E-MAIL
>
>
>
>
>
>
> -
The payload may be something other than an SDP body.
I suggest:
if(search("Content-Type: application/sdp"))
--
Alex Balashov - Principal
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
__
oxy in this fashion? I'm looking for the
> easiest fix.
>
> Thanks,
> Julian
>
> ___
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> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
--
Alex Balashov - Principal
Evar
Alex Balashov wrote:
> A new REGISTER request will certainly have a different Call-ID.
>
> However, I think the correct way to handle this is to just live with
> it. If the phone did not "unregister," then it will be subject to the
> consequences of that, and it'
info in the location
> table that is already there? maybe the callid or cseq or something?
>
> On Sat, Sep 26, 2009 at 1:43 PM, Alex Balashov
> mailto:abalas...@evaristesys.com>> wrote:
>
> As Inaki said, what you're trying to achieve is a little ridiculous.
>
__
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
--
Alex Balashov - Principal
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct : (+1) (678
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