:-) Kamailio's route script is brutally unforgiving in this regard.
Welcome to the community!
-- Alex
> On Apr 10, 2017, at 5:38 PM, Nicolas Pace wrote:
>
>> On Mon, 2017-04-10 at 17:15 -0400, SamyGo wrote:
>> Maybe a restart on kamailio service wasn't done. Or maybe you're
>> editing a differ
___
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users@lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
Web: http:
(Filed by UC Celeb Talk - 1 Apr 2017.)
BERLIN (Tegel), GERMANY--In the now-infamous breakdown episode on the
VIP tarmac at Berlin's Tegel Airport, Alex Balashov shocked the IP
telecoms celebrity world last month.
Shielding his eyes from the punishingly bright daylight of an overcast
d
On Tue, Mar 28, 2017 at 12:52:32AM +0500, Jade SZ wrote:
> PS: Also thanks a lot for the great blog post on SIP server tuning.
You're very welcome. Glad it proved useful to someone!
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1-706-510-6800 / +1-800-250-5920 (toll-fr
You might be running into semi-invisible/not widely publicised (but definitely
very real) PPS limits on given instance sizes.
On March 27, 2017 3:09:56 PM EDT, Jade SZ wrote:
>Hi Guys,
>
>I am running a simple REGISTER load test on:
>
>1) Kamailio sever with 2 cores - mem 5G
>2) Kamailio server
27;d want, you'd use:
$(ct{nameaddr.uri}{uri.host})
-- Alex
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
___
SIP Express
database.
>
> Thanks,
> Vivek
> ___
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users@lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Alex Balashov | Princ
s of the received reply.
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-user
append_to_reply("Record-Route: \r\n");
But I'd like to advise against it once more. :)
-- Alex
--
Principal, Evariste Systems LLC (www.evaristesys.com)
Sent from my Google Nexus.
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailin
After that I would create option for record calls for client
>and
>this is why I look for solution :)
>
>2017-03-14 7:47 GMT+01:00 Alex Balashov :
>
>> Yes, though of course you would have to correlate the calls (most
>likely
>> by Call-ID) and integrate all this.
>&g
relay and recorder ?
>
>2017-03-14 7:44 GMT+01:00 Alex Balashov :
>
>> It can record, as can a number of other media relays.
>>
>> On March 14, 2017 2:43:15 AM EDT, przeqpiciel
>> wrote:
>> >>> WHy not installing rtpproxy and proxying all
>
per call.
>>>
>>> The reality I see is however often very different RTP ports, most
>likely
>>> because load isn't 100% identical.
>>>
>>>
>>> Med venlig hilsen / Best regards
>>> Kjeld Flarup (Christensen) M.Sc E.E, Teknisk
ssibilities to have 2 concurent calls each through another asterisk
>
>> instance with this same rtp port. Am i right?
>>
>> So mqybe this magic device could see source IP address and route rtp
>> to correct adterisk?
>>
>> 13.03.2017 7:15 AM "Alex Balas
Otherwise, I'm not sure how you're avoiding it, since Asterisk
isn't aware of which ports from within the range are in use.
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
Web: http://www.evaristesys.com/, h
available to every instance.
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-us
Can you not assign different Asterisk instances different ranges of RTP ports
to allocate from?
On March 12, 2017 3:47:22 PM EDT, przeqpiciel wrote:
>I would like to ask you how you deal with several asterisks and
>kamailio on
>that same IP address, I have installation where i route 5060 to
>i
ion?
The dispatcher has no such limits. Also, how did you determine that the
dispatcher module is the essence of the "loss" that you are seeing? It
is most likely happening in a more general context:
http://blog.csrpswitch.com/tuning-kamailio-for-high-throughput-and-performance/
-- Alex
ored, however. That's on the endpoints, of course,
but it can make things worse.
> I'm trying to find a way to get a pesky setup to start behaving, of course,
> without requiring them to make any changes.
That's always a tough gig.
--
Alex Balashov | Principal | Evariste Systems
their replies contain Route headers that
are constructed from that Record-Route set.
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
_
modify the remote target URI.
-- Alex
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
___
SIP Express Router (SER) and Kamailio (Ope
Daniel,
Would you accept a patch to modify the size of that buffer to something
larger than 128? Like 256?
On Mon, Feb 20, 2017 at 07:25:02PM -0500, Alex Balashov wrote:
> On Mon, Feb 20, 2017 at 10:11:32PM +0100, Daniel-Constantin Mierla wrote:
>
> > There is also a way to set sor
lso in 4.4, but it was missing in the wiki docs, I just added for
> devel.
I think that's exactly what I need, thank you!
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
Web: http://www.evaristesys.com/, ht
ver, so it should be okay as long as I
don't run into any arbitrary limits on the size of the username.
On Mon, Feb 20, 2017 at 03:39:06PM -0500, Alex Balashov wrote:
> On Mon, Feb 20, 2017 at 09:19:45PM +0100, Olle E. Johansson wrote:
>
> > If you configure yourself as an out
rictly endogenously generated request.
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
___
SIP Express Router (SER) and Kamailio (O
On Mon, Feb 20, 2017 at 09:19:45PM +0100, Olle E. Johansson wrote:
> If you configure yourself as an outbound proxy you can start a
> transaction as you would normally do.
So you're saying I can wrap this entire exchange in t_newtran()?
--
Alex Balashov | Principal | Evariste Syste
that I can pass to the redirect server that it is
going to send back in the 3xx response.
On Mon, Feb 20, 2017 at 02:40:50PM -0500, Alex Balashov wrote:
> Hello,
>
> I am using uac_req_send() to send INVITEs to a redirect server. However,
> upon receiving the redirect, I need access t
, what are my options, other than building my own
state using htable or what have you? I would really rather not
maintain—and be responsible for garbage-collecting—externally
constructed state like that if possible.
Thanks!
-- Alex
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1-706-510
ded it
> to both.
>
> Thanks.
>
>
> --
> Andy Chen
> ___
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users@lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Ale
Thanks.
>
> JR
> --
> JR Richardson
> Engineering for the Masses
> Chasing the Azeotrope
>
> ___
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users@lists.sip-router.org
> http://li
Hello,
Is pua.publish strictly an MI function, or is it possible nowadays to
call it via the RPC channel, and specifically, using jsonrpc_exec()?
Thanks!
-- Alex
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free)
Web: http
ot;200", "OK", "text/plain", "");
} else
xhttp_reply("404", "Not Found", "text/html", "");
}
-- Alex
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1-706-510-680
gt; Active Internet connections (servers and established)
> Proto Recv-Q Send-Q Local Address Foreign Address State
> tcp0 0 homer02.me.com:sip*:* LISTEN
>
>
> I don't think this is a homer issue
elect,$var(i),DELIMITER})) {
$var(token) = $(var(str){s.select,$var(i),DELIMITER});
$var(i) = $var(i) + 1;
}
-- Alex
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free)
Web: http://www.evaristesys.com/, ht
Oh, I see. Yes, that could be.
-- Alex
> On Dec 22, 2016, at 12:49 PM, Daniel Tryba wrote:
>
>> On Thu, Dec 22, 2016 at 12:23:52PM -0500, Alex Balashov wrote:
>> That just sounds like the rtpproxy is not being engaged, i.e. that the
>> rtpproxy_manage() call is failing
That just sounds like the rtpproxy is not being engaged, i.e. that the
rtpproxy_manage() call is failing. When that happens, the SDP from .2 will be
passed through unaltered.
The Kamailio log should give you some idea of why the rtpproxy invocation has
failed.
RTPProxy is certainly not limited
The dialog module (dialog, not dialog_ng) would be a cleaner and more
natural solution, since it handles most possible eventualities of dialog
state transition for you more cleanly than if you keep state yourself
via htable.
-- Alex
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1
- sr-users mailing list
> sr-users@lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
__
On Sat, Dec 10, 2016 at 09:07:09PM +0200, Nahum Nir wrote:
> I tried Googling but didn'tfind anything.
> Is it possible to support DTMF using INFO?
Sure, absolutely. :-)
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free)
d be done
> automatically.
That's a good point. I forgot that mhomed= is off by default. :-)
-- Alex
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free)
Web: http://www.evaristesys.com/, http://w
4.x/modules/rr.html#idp20509612
-- Alex
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
___
SIP Express Router (SER) and Kama
with Kamailio. As such, it is off-topic for the list.
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
___
SIP Express Router
Wouldn't it be more fruitful to delete and reach old bindings in such a
scenario? Or are these bindings not only for web clients?
On December 1, 2016 5:54:06 PM EST, Colin Morelli
wrote:
>Hey all,
>
>I know Kamailio's registrar module has a max_contacts parameter that
>will
>limit the number o
On Thu, Dec 01, 2016 at 04:27:21AM +, Olvera, Victor wrote:
> Is this a real error?
>
> 9(1899) ERROR: cdp [peerstatemachine.c:635]: I_Snd_CER(): I_Snd_CER(): Error
> on finding local host address > Socket operation on non-socket
Is that, like, a philosophical question?
at these queries get "stuck". If they
become even a little slow, and you don't have sufficient child processes to
wait on them, at 200 CPS that can have consequences. Perhaps profile the
queries and/or log slow queries?
-- Alex
--
Alex Balashov | Principal | Evariste System
/tuning-kamailio-for-high-throughput-and-performance/
-- Alex
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
___
SIP Expr
not initiate a connect to the database
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
___
SIP Express Router (SER) and Kamai
Path is indeed the exact solution for this type of problem.
-- Alex
--
Principal, Evariste Systems LLC (www.evaristesys.com)
Sent from my Google Nexus.
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-ro
-router.org/cgi-bin/mailman/listinfo/sr-users
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
___
SIP Express Router (SER) and
do_things();
}
--
But without the onreply_route, it doesn't work.
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
_
Dialog Clean Timer
27700 tcp main process
So it is safe to remove the async_workers declaration entirely or set it
to 0, and I can still benefit from 8 async processes for async_route()?
-- Alex
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1-706-510-6800 (direct) / +1-800-250-5
ot;L_INFO", "xyz") happens to occur in a route[] block that is
resumed after async_route()—I have not tested other async
functions—there is no prefix:
Nov 22 00:29:52 allegro-4.evaristesys.com
/usr/local/sbin/kamailio[27678]: INFO: xyz
-- Alex
--
Alex Balashov | Principal |
order?
I would happily contribute, I just don't know how to update the core
cookbook.
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free)
Web: http://www.evaristesys.com/, http://www
rt it particularly, is it
still an encouraged practice? Or should one use #!defined constants
instead, as in the stock config?
Thanks,
-- Alex
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free)
Web: http://www.evariste
If I do, what happens if I have fewer core async_workers than modparam
workers? What if I have more? Should I aim to keep them in sync?
Thanks,
-- Alex
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free)
Web: http://www.e
Hi,
fr_timer and fr_inv_timer do different things:
- fr_timer goes off when there's no response _at all_ to the INVITE, not
even 100 Trying.
- If you received a 100 Trying or other provisional (1xx) repy,
fr_inv_timer will apply instead.
-- Alex
--
Alex Balashov | Principal | Eva
would prevent it is another; that's my preferred approach.
The ACC module is stateless. Think of it as a relatively unintelligent
logger. As with all loggers, it's your choice what to log, and to devise
means by which to prevent certain things from being logged.
--
Alex Balashov | Princ
s?
Don't do accounting for BYEs if they don't correspond to a tracked
dialog, or add a composote unique constraint on Call-ID + some other
column in your database to prevent the insertion of additional events.
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1-706-510-6800 (
You can always issue a second challenge that requires different credentials
than the first.
On October 28, 2016 7:41:16 AM EDT, Jayesh Nambiar wrote:
>Hello,
>I was wondering if it is possible to do a failover in digest
>authentication. For eg: I get two possible valid password values. Can I
>d
Hi,
Registration messages do not have SDP payloads. Typically, only INVITE
request messages and [some of] their corresponding replies do.
But you can access the encapsulated body via $rb.
-- Alex
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1-706-510-6800 (direct) / +1-800-250
>Cheers,
>Daniel
>
>
>On 21/10/16 11:45, Alex Balashov wrote:
>> Yeah, I'm trying to avoid something complex like keeping state in
>htable.
>>
>> I did try it - the docs are correct. drop() on a >= 2xx reply does
>nothing in a named (TM) onreply_route[
ive solution is using a hash table with expiration of the
>items matching the max timeout for transactions.
>
>Cheers,
>Daniel
>
>
>On 21/10/16 11:24, Alex Balashov wrote:
>> The core documentation says that in a named onreply_route[], only
>> provisional replie
is going to be called at the
3000 mark anyway? If so, does that mean the transaction is implicitly
woken up (t_continue()'d) at that moment?
2. Does the failure_route run in a normal SIP worker, or in an async
worker, if using async_task_route() here?
--
Alex Balashov | Principal | Eva
know whether to drop it, I need access to either AVPs
or, ideally, dialog variables. Since the global onreply_route is
executed by the core, I presume I won't have access to anything that
persists through TM there.
Thanks!
-- Alex
--
Alex Balashov | Principal | Evariste Systems LLC
Tel:
Hi,
If a transaction is currently t_suspended() and has not been
t_continued(), will t_check_trans() still find that transaction for
retransmission-dampening purposes, e.g. if a retransmission of the same
request as the original one is received?
Thanks,
--
Alex Balashov | Principal
man/listinfo/sr-users
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users
And yes, I was remiss in failing to mention that an effective solution
to scaling out rtpproxy is to bind multiple instances with different
core affinities.
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free)
Web: http
inst array of different SIP
>engines, including Kamailio. https://travis-ci.org/sippy/voiptests
>
>So with rtpproxy you are not locked in into single SIP engine, you can
>mix
>and match to fit your particular goal.
>
>And yes, last but not least, all our code is BSD licensed, so
Handling NAT, perhaps, but not correctly. The RURI doesn't change. But not does
it have to determine where the request is actually sent on the network and
transport layers.
-- Alex
--
Principal, Evariste Systems LLC (www.evaristesys.com)
Sent from my Google Nexus.
__
.
-- Alex
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users
also has lots of
other features which can be useful in, for example, running an RTP relay
in 1:1 NAT environments such as AWS, or in enabling WebRTC.
However, it is a bit more complicated to set up than vanilla rtpproxy.
Not much more, though.
-- Alex
--
Alex Balashov | Principal | Evariste
Hello,
Any reply code/status translation is going to invoke the absorption of the old
reply and the generation of a new one, within the proxy. Proprietary SIP
headers would not be conserved in such a scenario.
To add headers to a reply, you can use append_to_reply(), which works in much
the
in-dialog requests (e.g. end-to-end ACKs, BYEs,
reinvites).
Thus, it is entirely appropriate that your client is trying to send the
ACK to 192.168.0.200 on the network/transport layer.
-- Alex
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1-706-510-6800 (direct) / +1-800-250
Are you truly talking about 1000 call setups per second, or are you talking
about 1000 concurrent calls total?
If the latter, SIPp can handle that, with media.
If the former, you're going to need a lot of endpoints.
-- Alex
--
Principal, Evariste Systems LLC (www.evaristesys.com)
Sent from
support more than a few thousand concurrent
calls with RTP, you would be correct, but the same holds true of any
other endpoint out there. You would need a large, complicated,
horizontal farm of endpoints to establish, for example, tens of
thousands of calls with two-way RTP.
--
Alex Balashov
That should work, but be sure that this is executed in the context of an
onreply_route and not a request_route.
-- Alex
--
Principal, Evariste Systems LLC (www.evaristesys.com)
Sent from my Google Nexus.
___
SIP Express Router (SER) and Kamailio (Op
On 09/26/2016 08:34 PM, Carlyle Edwards wrote:
Hi, is there a manual for SIREMIS?
http://kb.asipto.com/siremis:index
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com
There's no simple answer to that. You have to learn to configure Kamailio in
general.
On September 26, 2016 3:50:02 AM EDT, Gaurav Bmotra
wrote:
>Hi
>i want to config gateway in kamailio
>i want to know how can i config IP Authenticate gateway and user/ pass
>gateway
-- Alex
--
Principal, E
Igor,
It might be simpler to do this:
remove_hf("Contact")
append_hf("Contact: <$fu>\r\n");
-- Alex
--
Alex Balashov | Principal | Evariste Systems LLC
1447 Peachtree Street NE, Suite 700
Atlanta, GA 30309
United States
Tel: +1-800-250-5920 (toll-free) / +1
the Via rather than the received source port (as would be
mandated by the presence of client-injected rport)?
As for trusting client headers, I strongly agree. Unfortunately,
sometimes business requirements collide with our philosophies.
--
Alex Balashov | Principal | Evariste Systems LLC
Tel
hen the proxy will return
replies to the source port of the request even if force_rport() is not
called, because that's the RFC 3581-compatible thing to do. Right?
2) Does reply_to_via=1 override the behaviour hypothesised in #1?
3) Does reply_to_via=1 override force_rport()?
Thanks!
-- Alex
You can also evaluate my take on whether Kamailio is interchangeable
with "SBC":
https://likewise.am/2013/03/10/kamailio-as-an-sbc-session-border-controller/
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free)
Hello,
Best advice I can give you is to listen to this podcast, where just this very
issue was discussed, among others:
https://www.vuc.me/2016/vuc-609-retrospective-highlights-of-15-years-of-ser-kamailio/
-- Alex
--
Principal, Evariste Systems LLC (www.evaristesys.com)
Sent from my Google
On 09/08/2016 01:00 AM, anfecora wrote:
THanks Alex,
do you think i can do something like where i check $FU reaching the rate
limit then blocket somehow for a few minutes.
Absolutely! I think you have the right idea.
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1-706-510-6800
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users
Not a problem. Glad it worked out!
-- Alex
--
Principal, Evariste Systems LLC (www.evaristesys.com)
Sent from my Google Nexus.
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip
issue?
Thanks
Jon
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1
ction to create dynamic pipes.
Excellent. That was indeed an important shift. :-)
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free)
Web: http://www.evaristesys.com/, http://www.csrpswitc
pc_dispatch();
else
xhttp_reply("404", "Not Found", "text/html", "");
}
Of course, in the real world, you'll want to consider authentication, IP
ACLs, and other things that go into some sort of security thought
process. But that's t
4.4.2.
Thanks,
Julia
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Alex Balashov | Principal | Evariste Systems LLC
1447 Peachtre
this nowadays for all cases that were handled
with fix_nated_contact() in the past for just this reason.
-- Alex
--
Alex Balashov | Principal | Evariste Systems LLC
1447 Peachtree Street NE, Suite 700
Atlanta, GA 30309
United States
Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (dir
uot;dropping scscf list on register
failure");
I_scscf_drop();
} else {
xlog("L_DBG", "This is a 401 - keep scscf list to do
optimisation");
}
break;
}
}
--
Alex Bala
d ACKs for 2xx responses is a problem
that occurs, both for initial and in-dialog invites (reinvites).
However, it indicates that something is misconfigured or improperly
implemented somewhere, and is a serious issue that needs to be addressed.
--
Alex Balashov | Principal | Evariste System
re-INVITE are
the same as for the initial INVITE (Section 13.2.1).
--
Alex Balashov | Principal | Evariste Systems LLC
1447 Peachtree Street NE, Suite 700
Atlanta, GA 30309
United States
Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct)
Web: htt
butes of the
negative reply on the winning branch.
-- Alex
--
Alex Balashov | Principal | Evariste Systems LLC
1447 Peachtree Street NE, Suite 700
Atlanta, GA 30309
United States
Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct)
Web: http://www.evaristesys.com/, http://www.csrpswitc
Huh. You're right, it's not! :-)
What value should I set it to for a 0.5 MB - 1 MB payload?
On August 10, 2016 3:03:32 AM EDT, Daniel-Constantin Mierla
wrote:
>
>
>On 10/08/16 09:01, Alex Balashov wrote:
>> I have indeed - it was in my original post:
>>
>
I have indeed - it was in my original post:
tcp_wq_max=20971520
On August 10, 2016 3:00:22 AM EDT, Daniel-Constantin Mierla
wrote:
>Have you set the parameter:
>
>https://www.kamailio.org/wiki/cookbooks/devel/core#tcp_conn_wq_max
>
>Cheers,
>Daniel
>
>
>On 09/08/16
vent fired when the server (Kamailio) aborts
the connection and closes the socket.
-- Alex
On 08/09/2016 12:27 PM, Alex Balashov wrote:
Hi,
I'm requesting 'dlg.list' from JSONRPC-S via XHTTP on a rather busy
server, i.e.
event_route[xhttp:request] {
xlog("L_INFO&qu
151674 bytes remaining to read
"profiles": ["total",[root@proxy kamailio]#
------
I can't seem to figure out what parameters regulate this. I'm using the
following TCP settings:
tcp_async=yes
tcp_connect_timeout=5
tcp_crlf_ping=no
tcp_keepa
On 08/08/2016 07:58 AM, Nahum Nir wrote:
When sending an invite and the calle not found or responding with an ACK
I want my script do some costume work.
When can I do it (using regular script)?
Sorry, what? :-)
--
Alex Balashov | Principal | Evariste Systems LLC
1447 Peachtree Street NE
, the app_python API is extremely limited.
--
Alex Balashov | Principal | Evariste Systems LLC
1447 Peachtree Street NE, Suite 700
Atlanta, GA 30309
United States
Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct)
Web: http://www.evaristesys.com/, http://www.csrpswitc
1 - 100 of 1526 matches
Mail list logo
