On 02/02/2014 08:17 PM, Alex Balashov wrote:
A wiser idea, however, would be to run Kamailio on a real, public
server, not behind
DNAT + dynamic IP. :-)
(crap, accidentally hit Ctrl+S, sorry).
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to reach the proxy, not that the proxy should be telling them how to
reach it. Putting Kamailio behind consumer-grade DNAT inverts the
envisaged client-server model.
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Tel: +1-678
On 02/03/2014 12:22 AM, Shankar wrote:
Also we found that in ‘dialog.c’ the variable dlg_hash_size should be
‘unsigned int’. Hope this will be taken care in next release of Kamailio.
How did you arrive at that conclusion?
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some
performance-related optimisations, such as only loading the language
interpreter into memory once, instead of at every invocation.
-- Alex
[1] http://kamailio.org/docs/modules/4.1.x/modules/sipt.html
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On 02/04/2014 01:44 AM, Alex Balashov wrote:
In fact, the only module that I am aware of that deals with
manipulation of encapsulated bodies at all (besides the far-end NAT
traversal and rtpproxy modules that modify RTP endpoints and other
SDP attributes) is the SIP-T module[1].
Well, I mean
/cgi-bin/mailman/listinfo/sr-users
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Tel: +1-678-954-0671
Web: http
the caller
and your Freeswitch servers. If Kamailio is able to maintain all the
state, this lends itself easily to use of the 'dispatcher' module:
http://kamailio.org/docs/modules/4.1.x/modules/dispatcher.html
... which is designed specifically for load-balancing applications.
-- Alex
--
Alex
):
modparam(tm, fr_timer, 3000)# 3 seconds
modparam(tm, fr_inv_timer, 9) # 1.5 minutes.
You can read more about these values here:
http://kamailio.org/docs/modules/4.1.x/modules/tm.html#fr_timer
http://kamailio.org/docs/modules/4.1.x/modules/tm.html#fr_inv_timer
-- Alex
--
Alex
to certain GSM
modems for the purpose of sending and receiving SMS and converting it to
SIP MESSAGE.
It doesn't aid in parsing encapsulated GSM messages in general.
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Tel: +1-678-954-0670
. :-)
-- Alex
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Tel: +1-678-954-0670
Web: http://www.evaristesys.com/, http://www.alexbalashov.com/
___
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the same for the second one.
This year, in addition to the main presentation track, there will be
some technical workshops, which I think might be of interest to new users.
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Alex Balashov - Principal
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this before... any suggetions appriciated.
greetz
Am 04.02.2014 15:56, schrieb Alex Balashov:
Indeed, Section 3.1 in the referenced RFC covers this pretty well.
On 4 February 2014 09:47:38 GMT-05:00, Olle E. Johansson
o...@edvina.net wrote:
On 04 Feb 2014, at 15:43, Moritz Graf
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of the initial INVITE
(and thus invocation of the registered dialog init TM callback)?
More generally, I was wondering if there has been any thought given to
the idea of separate timeout modparams for confirmed-nonACKed state, as
well as early state perhaps.
Thanks,
-- Alex
--
Alex Balashov
question is that yes, database operations
block the current SIP worker thread, just as almost all external
operations do. So, you definitely want your database queries to be fast
and to return quickly.
-- Alex
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;
flags |= PINGPONG;
if((flags POWPOW) != POWPOW)
printf(Problem?\n);
sasha@tambourine:~$ ./ns
Problem?
Are there other UL contact flags that could be in use here and breaking
this comparison in get_all_mem_ucontacts()?
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235 E
)
c-expires = tnow + 10;
continue;
}
}
}
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fledged keyboard.
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Tel: +1-678-954-0671
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through some rather
complicated spoofing.
You might consider instead reconfiguring your UAC to send different From
values, most likely by changing 'caller ID' settings in Asterisk, in
your case. For instance, there is the 'callerid' option for sip,conf peers.
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keyboard.
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Tel
, and thus lacking in the refinement one might expect from a
fully fledged keyboard.
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
United States
Tel: +1-678-954-0671
Web: http://www.evaristesys.com/, http
-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
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Tel: +1-678-954-0670
Web: http://www.evaristesys.com/, http://www.alexbalashov.com
be
another issue masquerading as having this effect. I'm just trying to
rule possibilities out.
Are there any situations that can cause a dialog that is otherwise being
normally tracked to be dumped from a profile of which it was previously
part?
Thanks!
-- Alex
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Tel: +1-678-954-0670
Web: http://www.evaristesys.com/, http://www.alexbalashov.com
Abdul,
Would you mind sending screenshots of the respective phases of this process?
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Is SIREMIS returning you to page 1 of the installation process, or is it
simply returning you to the username/password login screen?
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Tel: +1-678-954-0670
Web: http
Something I should probably add:
On 03/11/2014 10:28 AM, Alex Balashov wrote:
Hello,
We've got an issue we've been trying to track for days with 'dialog',
where the dialogs go through the following life cycle:
1. INVITE --
2. -- 407 challenge
3. ACK --
4. INVITE --
5. -- 100 Trying
6. -- 180
manipulation work at all
beforehand?
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Tel: +1-678-954-0670
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___
SIP
with
profile affinity prior to t_relay(), and now it is consequential because
the callbacks are run immediately?
On 03/11/2014 01:44 PM, Alex Balashov wrote:
We appear to have fixed this problem by calling dlg_manage() before
doing any set_dlg_profile() manipulations.
The documentation is not clear
Daniel,
On 03/12/2014 04:27 AM, Daniel-Constantin Mierla wrote:
On 11/03/14 18:44, Alex Balashov wrote:
We appear to have fixed this problem by calling dlg_manage() before
doing any set_dlg_profile() manipulations.
The documentation is not clear on whether dlg_manage() needs to be
called
?
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Hi,
Does evapi_relay() have to emit JSON blobs? Or can it be used to emit
generic strings?
Thanks!
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Tel: +1-678-954-0670
Web: http://www.evaristesys.com/, http
behaviour and use the source port instead.
2) I have a 7960 desk phone that I use with Kamailio. It's behind NAT in
relation to Kamailio, and giving it the usual NAT treatment (nathelper
fixups, rtpproxy) works just fine.
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compiler
optimisations, and with additional debug information, e.g. -g.
-- Alex
--
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Tel: +1-678-954-0670
Web: http://www.evaristesys.com/, http://www.alexbalashov.com
Kamailio on the network and transport layers.
-- Alex
--
Alex Balashov - Principal
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235 E Ponce de Leon Ave
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Tel: +1-678-954-0670
Web: http://www.evaristesys.com/, http://www.alexbalashov.com
that involves buffering and parsing strings
internally, which in turn draws on static (stack) and dynamic (heap)
memory allocations. All of that influences the memory state of the
program, and thus has an impact on whether it'll crash, and when it will
do so.
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-users
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Tel: +1-678-954-0670
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___
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to take a different schematic approach, you can
always just use 'sqlops' to do your own queries.
-- Alex
--
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Tel: +1-678-954-0670
Web: http://www.evaristesys.com/, http
be
wiser to simply deal in URL references that pull an avatar image from
somewhere else.
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http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
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Alex Balashov - Principal
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235 E Ponce de Leon Ave
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Decatur, GA 30030
United States
Tel: +1-678-954-0670
Web: http
How are you checking the SRV records, exactly? Can you paste the
invocation and output of the 'host' command, or whatever you're using?
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Tel: +1-678-954-0670
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place calls
between two extensions. Getting 404 all the time.
Could that be a domain issue? (Domain is a component of location
resolution.)
That said, you can certainly run Kamailio in a registrar role without
database backing, using solely the in-memory 'usrloc'.
-- Alex
--
Alex Balashov
;; MSG SIZE rcvd: 177
- Original Message -
From: Alex Balashov abalas...@evaristesys.com
To: sr-users@lists.sip-router.org
Sent: Monday, March 24, 2014 12:43:58 AM
Subject: Re: [SR-Users] kamailio db
On 03/24/2014 12:33 AM, Slava Bendersky wrote:
9(6003) DEBUG: core [dns_cache.c
:
Hello Alex,
Thank you for help, I added xlog, but I don't any records in log.
When I restart kamailio I see
CRITICAL: usrloc [udomain.c:447]: preload_udomain(): empty username
record in table location...skipping
Slava.
- Original Message -
From: Alex Balashov abalas...@evaristesys.com
/kamailio[9709]: INFO: script: This
120 and this null
When I specify proxy settings on sip client zoiper, rtp start working
and at least I can dial voicemail on asterisk behind.
Slava.
- Original Message -
From: Alex Balashov abalas...@evaristesys.com
To: Kamailio (SER) - Users Mailing
it doesn't matter what the response is.
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___
SIP Express
?
Yes, if you want a location binding to go into the 'location' table,
that is accomplished with save(). And that is something that should
happen after you've authenticated the request. :-)
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Alex Balashov - Principal
Evariste Systems LLC
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and pieces of software.
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Decatur, GA 30030
United States
Tel: +1-678-954-0670
Web: http://www.evaristesys.com/, http://www.alexbalashov.com/
___
SIP Express Router (SER
.
*From: *Alex Balashov abalas...@evaristesys.com
*To: *sr-users@lists.sip-router.org
*Sent: *Wednesday, March 26, 2014 11:55:18 AM
*Subject: *Re: [SR-Users] kamailio db
On 03/26/2014 11:53 AM, Slava Bendersky wrote:
Hello Alex,
I added
On 03/26/2014 03:23 PM, Slava Bendersky wrote:
Hello Alex,
Here config in pastebin
http://pastebin.com/hKDZC1iY
I'll take a look and get back to you as soon as I can. Right now, I need
to catch a flight to Kamailio World. :)
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Tel: +1-678
:
http://kamailio.org/docs/modules/4.1.x/modules/rr.html#append-fromtag-id
-- Alex
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, as well. However, I am not sure if you will have
access to them inside the local event_route. That's a good question.
[1] http://www.kamailio.org/wiki/cookbooks/4.1.x/pseudovariables#dlg_var_key
On 03/28/2014 03:56 AM, Alex Balashov wrote:
On 03/28/2014 03:53 AM, Kelvin Chua wrote:
I have
... only if they are visible inside the event_route. I haven't tried
it. Do let me know if it works, that would be interesting!
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Router (SER) and Kamailio (OpenSER) - sr-users mailing list
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Web: http
Sweet!
On 28 March 2014 15:24:10 GMT-04:00, Kelvin Chua kel...@gmail.com wrote:
It worked! perfect
Kelvin Chua
On Fri, Mar 28, 2014 at 1:34 AM, Alex Balashov
abalas...@evaristesys.comwrote:
On 03/28/2014 04:32 AM, Kelvin Chua wrote:
I tried, AVPs, they are not accessible
://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
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Tel: +1-678-954-0670
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While you're at it, port Kamailio to AIX, too. :-) Could be fun.
On 03/31/2014 09:25 AM, Rizwan Khan wrote:
OK, i will :)
Rizwan Khan
On Mon, Mar 31, 2014 at 6:16 PM, Alex Balashov
abalas...@evaristesys.com mailto:abalas...@evaristesys.com wrote:
I don't know that there are any HP
Social Configuration module for the Kamailio open-source SIP server
and SIP proxy.
Evariste principal Alex Balashov explained:
It's no secret that writing Kamailio configuration in its embedded
scripting language is complex. Social Configuration integrates with
Twitter and Facebook to allow tens
sorry if some questions
are
really naive. I am new to all this and trying to get hold of the
whole
technology stack.
Rizwan
On Mar 28, 2014 3:41 PM, Alex Balashov abalas...@evaristesys.com
wrote:
1. Kamailio does not handle RTP, so audio is not a scaling factor
at all;
2. The only scaling
-04:00, Alex Balashov abalas...@evaristesys.com
wrote:
All modules are listed here:
http://kamailio.org/docs/modules/4.1.x/
On 1 April 2014 03:12:10 GMT-04:00, Rizwan Khan rizk...@gmail.com
wrote:
Thanks a lot Pedro.
Just one question? Do we have module for Mongodb? how do i use it?
Rizwan
-router.org/cgi-bin/mailman/listinfo/sr-users
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not provide for that, no. However, there is also no
semantic significance to the order of appearance of the headers. Is
there some reason you want to do this beyond the merely aesthetic?
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, and thus lacking in the refinement one might expect from a
fully fledged keyboard.
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
United States
Tel: +1-678-954-0671
Web: http://www.evaristesys.com/, http
evaluates to true.
Much appreciated!
-- Alex
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Tel: +1-678-954-0670
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___
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Hello,
I withdraw this question on grounds of idiocy. My query was returning a
row when I thought it wasn't.
In actual fact,
if(!defined $xavp(gateways=id))
works just fine, because the xavp list is null in that case.
On 04/05/2014 11:32 AM, Alex Balashov wrote:
Hi,
When using
that.
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AM, Alex Balashov
abalas...@evaristesys.comwrote:
On 04/05/2014 12:14 PM, Fred Posner wrote:
I don't think so. As I understood the documentation, at least,
$dbr
doesn't get populated in this case; the rows just go straight to
an
xavp list. I suppose I should verify that.
Looks like
On 04/06/2014 12:37 AM, Kelvin Chua wrote:
$dbr for SELECTs
Unless it's not, because you're using sql_xquery(). :-)
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one might expect from a
fully fledged keyboard.
Alex Balashov - Principal
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235 E Ponce de Leon Ave
Suite 106
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United States
Tel: +1-678-954-0671
Web: http://www.evaristesys.com/, http://www.alexbalashov.com
);
if(!defined $xavp(gateways)) {
# dang nabbit
}
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?
In my opinion, the answer is no. The client should not be so
presumptuous. It should specify the transport in its Contact, to
indicate how it wants to be reached, and leave other decisions about
transport to the UAS. But is there something I am perhaps missing?
Thanks,
--
Alex Balashov
this preference. Since the next hop is opaque to
the client, why should it have a say in choosing the transport used to
reach it? Why should it care?
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.
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Alex Balashov
, Apr 8, 2014 at 12:51 AM, Alex Balashov
abalas...@evaristesys.comwrote:
What happens if you call a different number?
On 7 April 2014 21:14:45 CEST, Amogh meshram pacifir...@gmail.com
wrote:
Hello everyone,
From today i started getting SIP 480 error when making outbound
call.
I checked out
...@gmail.com
wrote:
same 480 error
On Tue, Apr 8, 2014 at 12:51 AM, Alex Balashov
abalas...@evaristesys.com wrote:
What happens if you call a different number?
On 7 April 2014 21:14:45 CEST, Amogh meshram
pacifir...@gmail.com
wrote:
Hello everyone,
From today i started getting SIP 480 error
/inter-jurisdictional rate differences.
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___
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at 1:21 AM, Alex Balashov abalas...@evaristesys.com
mailto:abalas...@evaristesys.com wrote:
Hello,
It's just a guess, but I think most SIP providers aren't going to
like the all-zeroes Caller ID/ANI that you're sending here:
Contact: sip:00@1.22.216.96
mailto:sip
defines its own api
3) other suggestions for what should be discussed?
Looking forward to opinions regarding these matters.
Cheers,
Daniel
--
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
United States
Tel: +1-678-954-0670
Web: http
(OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
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Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
United States
Tel: +1-678-954-0670
Web: http://www.evaristesys.com/, http
should look at the value of $ru (which can be split into its component
parts: $rU - the user part, $rd - the domain part, $rp - the port).
-- Alex
--
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
United States
Tel: +1-678-954-0670
Web: http
keyboard.
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
United States
Tel: +1-678-954-0671
Web: http://www.evaristesys.com/, http://www.alexbalashov.com___
SIP Express Router (SER) and Kamailio
/listinfo/sr-users
--
Sent from my mobile, and thus lacking in the refinement one might expect from a
fully fledged keyboard.
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
United States
Tel: +1-678-954-0671
Web: http://www.evaristesys.com
.
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Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur
the 183/200 arrive, this
creates havock in some endpoints but is undesirable in all cases IMHO
if the delay of the 182/200 is to high (0.1s).
Just of curiosity, how exactly does the havoc manifest itself? What
are the symptoms of this being a problem?
--
Alex Balashov - Principal
Evariste Systems
, despite what the
documentation says?
--
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
United States
Tel: +1-678-954-0670
Web: http://www.evaristesys.com/, http://www.alexbalashov.com/
___
SIP Express
If it helps, here are my rtpproxy-ng modparams:
modparam(rtpproxy-ng, rtpproxy_sock, 1 == udp:xxx.xxx.xxx.xxx:5050)
modparam(rtpproxy-ng, rtpproxy_disable_tout, 120)
modparam(rtpproxy-ng, rtpproxy_tout, 1)
modparam(rtpproxy-ng, rtpproxy_retr, 2)
--
Alex Balashov - Principal
Evariste Systems LLC
of the control protocol is correct, this is actually
an answer, not a mislabeled offer, since the offer would contain some
other attributes.
But maybe not?
I expect an answer because I have an rtpproxy_manage() in an
onreply_route for 18x/2xx+SDP responses to this INVITE, of course.
--
Alex
is that the evaluative context tells
rtpproxy_manage() that it's dealing with a _reply_ (the 302 redirect),
not a _request_, so it should be sending an 'answer' on that basis.
Does that sound reasonable? I don't have an easy way of testing this
thesis right now since it's production.
--
Alex Balashov
a delete command if it's a = 300 reply.
On 04/23/2014 12:51 PM, Alex Balashov wrote:
Maybe it's not a bug. I think I've got a theory about what may be
happening. Admittedly, it's not trivial to follow, but bear with me.
The invocation of rtpproxy_manage() is happening in a REQUEST_ROUTE
as expected.
The legacy rtpproxy module may well behave the same way. I hadn't tried
to use rtpproxy_manage() in this scope before, which is why I was
imagining it to have always worked.
--
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
United
it's been discussed in various
dialog_ng discussions in the past that I have not carefully monitored.
Thanks!
-- Alex
(With apologies for cross-posting.)
[1] Or correctly associated based on tight matching.
--
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
: sip:ip.of.proxy:5060\r\n);
--
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
United States
Tel: +1-678-954-0670
Web: http://www.evaristesys.com/, http://www.alexbalashov.com/
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Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
United States
Tel: +1-678-954-0670
Web: http://www.evaristesys.com/, http://www.alexbalashov.com
, the question is, am I doing something wrong? What's the best way to
accommodate this scenario? I don't know if I want to track the dialog or
add it to a profile until after I get the 302.
Thanks,
--
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
On 04/28/2014 08:04 PM, Fred Posner wrote:
On 04/28/2014 06:54 PM, Alex Balashov wrote:
I don't think that will work, because no dialog is created by the 302
redirect.
11.3. event_route[dialog:failed]
Executed when dialog is not completed (+300 reply to INVITE).
Oh. I thought it required
could achieve my objective by tracking every
dialog--that is, by calling dlg_manage() in the initial request route
for every call, and then calling set_dlg_profile() out of a failure
route conditionally?
--
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur
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