Brilliant! Bravo! Bravo! Brovo!
Gonna miss you Bro! Get in this neck of the woods, got some dope a$$ hooch and
BBQ waitin!
Peace out!
JR Richardson
Engineering for the Masses
Chasing the Azeotrope
--
Message: 1
Date: Sat, 1
actually change the SIP message contact
when passing through the proxy. I'm using kamailio 4.2.
Thanks.
JR
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*:*
LISTEN 16018/kamailio
Thanks.
JR
> If you use netstat, what is in the recv queue for tcp packets on sip ports?
>
> netstat -altp
>
> Cheers,
> Daniel
>
>
> On 18/01/2017 16:43, JR Richardson wrote:
>> Yes, this is a sipcapture node. I'm listening on a s
.pid
Or pipe to kamailio local unix socket?
I don't know, I'm just guessing.
Thanks.
JR
> Somehow is not clear for me how you have the configuration there ...
> before commenting further, this needs to be clarified.
>
> The node you presented the config is a sipcapture insta
> On Mon, Jan 16, 2017 at 10:29:39AM -0600, JR Richardson wrote:
>> Yes, I'm familiar with the methods sipcapture uses, I don't use HEP,
>> using raw socket capture, I think this may be a sipcapture issue,
>> debuging kamailio shows normal startup and processing of U
> module detecting it's a packet for it, can execute the request_route
> once it builds a local sip packet.
>
> To troubleshoot further, run with debug=3 in kamailio.cfg and see what
> you get in the logs.
>
> Cheers,
> Daniel
>
> On 13/01/2017 16:33, JR
Iptables is not blocking, but it was worth a check.
Thanks.
JR
I assume you have ruled out firewall? It's something that can nab even
experienced people:
# iptables -Ln
-- Alex
On Thu, Jan 12, 2017 at 03:25:27PM -0600, JR Richardson wrote:
> Hi All,
>
> Just enabled SIP
TE \"$fU\" to \"$rU\"
from \"$si\"\n");
Logging reports all SIP UDP traffic to logs fine, but no TCP traffic.
root@homer02:~# netstat -al
Active Internet connections (servers and established)
Proto Recv-Q Send-Q Local Address Foreign Address
e are any safe scripts that check the running process
for kamailio and restart if automatically if not running?
I dug around a bit but not seeing any hits for auto start kamailio scripts.
And another question would be if this occurrence of shutting down
should be of concern or not?
Thanks.
JR
-
On Mon, Dec 30, 2013 at 6:08 PM, Jr Richardson wrote:
>>
>> Hello,
>>
>> you should enable core dumping - that should be via running 'ulimit -c
>> unlimited' before you start rtpproxy.
>>
>> If you get the coredump, then grab the backtrace with
th UDP in the media port range or
should I open up media port range to all PBX's and not worry about
attacks. Are there any UDP Media exploits that I should be concerned
with, or UDP flood attacks that could DOS my hosted PBX's?
Thanks for any feedback.
JR
--
JR Richardson
Enginee
n spot a fix for it.
>
> Hopefully rtpproxy package is with debug symbols, if not, the look for a
> rtpproxy-dbg package and install it if found.
>
> Cheers,
> Daniel
>
> On 20/12/13 18:59, Jr Richardson wrote:
>> Hi All,
>> root@sip-router3-ve206:/etc/kamailio# kama
127.0.0.1:7722")
inside route[]
add_path_received();
rtpproxy_manage("cwei");
record_route();
Any guidance on further identifying the issue?
Thanks.
JR
--
JR Richardson
Engineering for the Masses
___
SIP
Hi Daniel,
What is the status and possible release date for your admin book. The last
update is it will be released as an e-Book only, that's ok with me.
Thanks.
JR
JR Richardson
- - - - - - - - - - -
Engineering for the M
coming DIDs, etc.
---Fred
> On Oct 25, 2013, at 5:43 PM, Jr Richardson
wrote:
>
> Hi All,
>
> Starting a new project, roll your own SBC, not a full SBC, just need some
minor functionality. I'm interested in deploying Kamailio as a edge device
on a VSP for single entry po
Hi All,
Starting a new project, roll your own SBC, not a full SBC, just need some
minor functionality. I'm interested in deploying Kamailio as a edge device
on a VSP for single entry point for hosted PBX's, Asterisk based. I had
some wonderful and informative conversations at Astricon 2013, seve
be implemented to perform well for voice, just got to do
your homework.
Good luck!.
JR
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Hi All,
I'm considering running Kamailio as a virtual machine, with such low
utilization, it doesn't seem to make sense to keep running it on a physical
host server.
I've been virtualizing Asterisk PBX's for years and run a host of other
virtualized servers with OpenVZ, VMware and MS V-Serv
On Wed, Aug 8, 2012 at 12:34 PM, JR Richardson wrote:
> On Wed, Aug 8, 2012 at 12:13 PM, JR Richardson
> wrote:
>> Hi All,
>>
>> I'm running redundant kamailio 3.0.4 servers in production, have been
>> for a long time with great success. They were insta
On Wed, Aug 8, 2012 at 12:13 PM, JR Richardson wrote:
> Hi All,
>
> I'm running redundant kamailio 3.0.4 servers in production, have been
> for a long time with great success. They were installed on debian
> Lenny. One of my servers crashed. I can't seem to do a debian
justed the init script to check for networking and mysql to
start first and performed an 'update-rc.d kamailio defaults' but still
no luck.
Any help will be appreciated.
Thanks.
JR
--
JR Richardson
Engineering for the Masses
___
SIP Express
@gmail.com]
> Sent: Friday, January 20, 2012 5:06 AM
> To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users
> Mailing List
> Cc: JR Richardson
> Subject: Re: [SR-Users] onreply _route syntax error for homer capture
> server
>
> Hello,
>
> can y
}
51drop;
52 }
This example was used from the homer wiki
http://code.google.com/p/homer/wiki/Kamailio
root@homer:/etc/kamailio# kamailio -V
version: kamailio 3.2.1 (x86_64/linux) bee094
Any guidance is appreciated.
Thanks.
JR
--
JR Richardson
Engine
> On 01/16/2012 09:11 PM, JR Richardson wrote:
>
> > So my question; what is the difference between a drop; and exit;
> > within the on_reply route
>
> Remember that an onreply_route is a callback that allows you to
> intercept a reply, but the reply is forwarded by de
he hell is the capture server
sending messages out the eth0 network interface when the server is
100% bound to eth1 capture interface?
Any insight or guidance is appreciated.
Thanks.
JR
--
JR Richardson
Engineering for the Masses
___
SIP Express Router
>
>>> Above is the part that is a little misleading, it says that
>>>
>>> "When an address is marked as inactive or probing, it will be ignored
>>> by 'ds_select_dst' and 'ds_select_domain'."
>>>
>>> This, to me,
On Wed, May 11, 2011 at 10:42 AM, JR Richardson
wrote:
> Hi All,
>
> So I'm still banging away at monitoring various SIP trunk capacity,
> the dialog module is working well for this purpose. I am running
> local server cron job scripts to pull data and put into local flat
&
nd sharing some scripts or from anyone that has an
alternative method.
Any guidance or affirmation that this is a good method that works
would be helpful.
On the MRTG server, I'm using perl scripts for a lot of my data
pulling. I am using Kamailio 3.0.
Thanks.
JR
--
JR Richardson
Engineering
On Tue, May 10, 2011 at 3:54 PM, JR Richardson wrote:
> Hi All,
>
> Been playing around with the dialog module to help me keep track of
> SIP trunk utilization, trying to identify how many calls are active on
> each SIP trunk. My dialog count for inbound calls is 10 (correct) bu
27;kamctl fifo dlg_list' will list only 10 active dialogs.
So my question is, why am I getting a 20 count for outbound and only
10 count for inbound?
Thanks.
JR
--
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Engineering for the Masses
___
SIP Express Router (SER) and Kam
> On 09.05.2011 22:44, JR Richardson wrote:
> > Would there be any usage examples of dialog module? I'm not sure if I
> > really need profiles and [values] and when to set and unset. A push
> > in the right direction would be helpful.
>
> Hi,
> here is an
> > The problem I am having specifically is for outbound calls flowing
> > through a dispatcher, I want to separate calls into a profile for each
> > SIP domain that I send calls to. I'm not sure how do accomplish this.
> > I see in kamctl fifo dlg_list, there is 'caller_bind_addr::
> > udp:10.1
:10.10.12.24:5060'. Is there a way to export that into a variable
> that I can use as the 'profile' for the dialog?
>
Actually, I can't use the 'caller_bind_addr::udp:10.10.12.24:5060' it
should be the 'callee_contact:: sip:5551212@10.10.14.101' without
o export that into a variable
that I can use as the 'profile' for the dialog?
Thanks.
JR
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http:
; need to do billing or capture cdr's on the SIP-Routers so the simpler
>> the better.
>
> Honestly, going the dialog module route is your easiest bet.
>
> --
> Alex Balashov - Principal
Thanks Alex,
Would there be any usage examples of dialog module? I'm no
g for ver 3.0? I'm not
opposed to using the dialog module, I'm just not familiar with it and
not sure the best way to integrate it for usage tracking. I don't
need to do billing or capture cdr's on the SIP-Routers so the simpler
the better.
Any guidance will be much appreci
On Mon, Nov 15, 2010 at 9:13 AM, Daniel-Constantin Mierla
wrote:
>
>
> On 11/13/10 5:42 PM, JR Richardson wrote:
>>>
>>> -Original Message-
>>> From: Iñaki Baz Castillo [mailto:i...@aliax.net]
>>> Sent: Saturday, November 13, 2010 8:10 A
> -Original Message-
> From: Iñaki Baz Castillo [mailto:i...@aliax.net]
> Sent: Saturday, November 13, 2010 8:10 AM
> To: Daniel-Constantin Mierla
> Cc: JR Richardson; SR-Users
> Subject: Re: [SR-Users] Handle '486 busy here' from upstream carrier
>
> 2
On Thu, Nov 11, 2010 at 6:29 PM, Iñaki Baz Castillo wrote:
> 2010/11/11 JR Richardson :
>> I was thinking about including this in my failure route:
>>
>> if (t_check_status("486")) {
>> append_branch();
>> t_relay();
>> }
>>
>>
Hi All,
Asterisk>http://pastebin.com/crfMe81D
Here is a pastebin of the call graph:
http://pastebin.com/rnQZDyFU
I was thinking about including this in my failure route:
if (t_check_status("486")) {
append_branch();
t_relay();
}
Would that do any good?
Thank
On Thu, Nov 11, 2010 at 1:05 PM, Daniel-Constantin Mierla
wrote:
>
>
> On 11/11/10 7:42 PM, JR Richardson wrote:
>>>
>>> Hi All,
>>>
>>> I'm still getting these errors and I'm struggling to resolve the
>>> problem. I think I&#x
>
> http://pastebin.com/VHhQ1sJY
>
> My config:
>
> http://pastebin.com/BYn4g5ur
>
> Do I need to have a mechanism to deal with a 486 form the carrier?
>
> Thanks.
>
> JR
> --
> JR Richardson
> Engineering for the Masses
>
>
>
> --
need to have a mechanism to deal with a 486 form the carrier?
Thanks.
JR
--
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Engineering for the Masses
___
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http://lists.sip-router.org/
n't filter out canceled
> transaction and try to re-route even the caller canceled the call.
>
> Add at the top of your failure route:
>
> if (t_is_canceled()) {
> exit;
> }
>
> Cheers,
> Daniel
>
> On 11/5/10 4:48 PM, JR Richardson wrote:
>
7;s without a
response. I believe this is why I am getting the log errors. I
played around with "modparam("tm", "cancel_b_method", 2)" but have the
same results.
Here is a pastebin of my kamailio.cfg:
http://pastebin.com/Tn3humr0
Any suggestions on where to
Hi All,
I'm seeing random errors in my Kamailio 3.0.2 log:
ERROR: tm [tm.c:1300]: ERROR: w_t_relay_to: t_relay_to failed
ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding
After going blind staring at wireshark, I can consistently reproduce
this error by initiating a c
Dallas, no telecommuters please.
Thanks.
JR
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>
> JR, if you don't already know SIPSAK, go download it. It's what I use in
> Nagios and many other scripts.
> Nils has done a great job with it.
>
> /O=
Thanks Olle, I'll check it out.
JR
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-use
On Fri, Oct 15, 2010 at 10:50 AM, Fred Posner wrote:
> On Oct 15, 2010, at 11:42 AM, JR Richardson wrote:
>
>> On Fri, Oct 15, 2010 at 10:22 AM, Fred Posner wrote:
>>> Hey JR...
>>>
>>> I use this:
>>>
>>> #! /usr/bin/perl -w
>&
or die("FAIL\n");
>
> if ($msg) {
> print "UP\n";
> print "response is $msg\n";
> } else {
> print "FAIL no msg received\n";
> }
> close($socket);
>
>
> ---fred
> http://qxo
Hi All,
Can someone point me in the right direction of a command line SIP Ping
utility or how to invoke from Kamailio? I see there is a sip_ping.pl
script in voip-hacks, does anyone have copy-paste text version of
that, all I can find is the PDF?
Thanks.
JR
--
JR Richardson
Engineering for
snappy, even in realtime.
>
So the first thing I notice with using 'uac_replace_from()' is the
caller ID number is replaced with the replaced value, so I guess this
is not what I need for my particular application. I'll fool around
with the 'append_hf' and some patte
)},${EXTEN},1)
>
> ...
>
> [my_context]
>
> exten => 5551212,1,Answer
> exten => 5551212,n,Playback(hello-world)
> ...
> Alex Balashov - Principal
> Evariste Systems LLC
Hi Alex. I use a general context in asterisk
module
should I be looking into?
Thanks.
JR
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995 DOMAIN=2
PDT:: SDOMAIN=* PREFIX=96 DOMAIN=2
PDT:: SDOMAIN=* PREFIX=97 DOMAIN=2
PDT:: SDOMAIN=* PREFIX=98 DOMAIN=2
PDT:: SDOMAIN=* PREFIX=99 DOMAIN=2
Using the list prefix search option will pipe out about 10,000
records, but errors out with "no more pkg mem" past that
ue was occurring, I
restarted the mysql process and the table loaded into sr. I'm now
suspecting I have a buggy mysqld so I'm going to try to update that
and see if I can reproduce the problem. So for now I can't isolate
the issue, but I'm still working on it.
I do see a differ
On Tue, Jul 20, 2010 at 12:04 PM, Elena-Ramona Modroiu
wrote:
> On 07/20/2010 05:06 PM, JR Richardson wrote:
>>>>>
>>>>> [..]
>>>>> When I added 180K records in the database, I got the "no more pkg mem"
>>>>> error aga
c:283]: bad parameters
0(22456) INFO: pdt [pdt.c:490]: no prefix found in [2000171212]
Of course wihout a propper table load, pdt_list does not produce any
results (but i don;t get the pkg_mem error any more).
sip-router2:~# kamctl fifo get_statistics all
shmem:total_size = 536870912
shmem:used_si
IZE 64*1024*1024
/*used if SH_MEM is defined*/
#define SHM_MEM_SIZE 64
So now it looks like there is some hard limit on the route size you
can deploy for the pdt module? Is this the case or is there somewhere
I can increase the available shmem for pdt to use?
Thanks.
JR
--
JR Richardson
Engin
On Mon, Jul 19, 2010 at 2:41 AM, marius zbihlei wrote:
> JR Richardson wrote:
>>
>> On Fri, Jul 16, 2010 at 4:51 PM, JR Richardson
>> wrote:
>>
>>>
>>> Hi All,
>>>
>>> I loaded up the PDT database with about 35K records and when
On Fri, Jul 16, 2010 at 4:51 PM, JR Richardson wrote:
> Hi All,
>
> I loaded up the PDT database with about 35K records and when I issue
> the commad "kamctl fifo pdt_list" I get:
>
> 3(3018) ERROR: [tree.c:139]: no more pkg mem
> 3(3018) ERROR: mi_fifo [fif
5" to show this prefix
route. But without enough memory, doesn't work.
Thanks.
JR
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http://lis
On Fri, Jul 16, 2010 at 2:50 PM, Alex Balashov
wrote:
> Custom queries are your way to go on this.
Thanks for the suggestion Alex. I'll play around with that.
JR
--
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Engineering for the Masses
___
SIP Express Router (SER) and
On Fri, Jul 16, 2010 at 2:37 PM, JR Richardson wrote:
> Hi All,
>
> I'm in the lab with sip router 3.0 mocking up some trunk-group prefix
> routing with PDT and load balancing with Dispatcher. This is working
> great and as expected. I would like to incorporate some sort
module parameter be added to not strip the prefix. But
I'm not sure this will work, can the PDT module hold in memory a few
hundred thousand records?
Thanks.
JR
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Engineering for the Masses
___
SIP Express Router (SER) and Kama
> -Original Message-
> From: Juha Heinanen [mailto:j...@tutpro.com]
> Sent: Thursday, July 01, 2010 1:43 AM
> To: JR Richardson; SR-Users
> Subject: Re: [SR-Users] permissions module address_reload not working
>
> Juha Heinanen writes:
>
> > revert
On Wed, Jun 30, 2010 at 4:51 PM, Daniel-Constantin Mierla
wrote:
> Hello,
>
> On 6/30/10 10:17 PM, JR Richardson wrote:
>>
>> Hi All,
>>
>> I'm running kamailio 3.0.2.
>>
>
> hmm, checking the logs I see changes done lately that introduced some bu
ERROR: permissions [address.c:82]: db_handle already exists
The address table loads fine during startup and if I reload kamailio
it will load any new addresses or subnets so I know there is no syntax
error in the database.
I can reload trusted table and pdt table without an error but address
tabl
On Tue, Jun 29, 2010 at 11:33 AM, Daniel-Constantin Mierla
wrote:
>
>
> On 6/29/10 6:17 PM, JR Richardson wrote:
>
> On Tue, Jun 29, 2010 at 10:25 AM, Daniel-Constantin Mierla
> wrote:
>
>
> Hi JR,
> On 6/28/10 11:39 PM, JR Richardson wrote:
> Hi All,
> I
On Tue, Jun 29, 2010 at 11:33 AM, Daniel-Constantin Mierla
wrote:
>
>
> On 6/29/10 6:17 PM, JR Richardson wrote:
>
> On Tue, Jun 29, 2010 at 10:25 AM, Daniel-Constantin Mierla
> wrote:
>
>
> Hi JR,
> On 6/28/10 11:39 PM, JR Richardson wrote:
> Hi All,
> I
On Tue, Jun 29, 2010 at 10:25 AM, Daniel-Constantin Mierla
wrote:
> Hi JR,
>
>
>
> On 6/28/10 11:39 PM, JR Richardson wrote:
>
> Hi All,
> I'm testing dispatcher functions with kamailio 3.0 and using database
> to load destination address. The problem I am
> 2010/6/28 JR Richardson :
> > Kamailio 3.0 permissions module, I see that the address database table
> > is cached by default, really the only option. ?But there is no MI
> > Function to reload the database into cache, you must restart kamailio
> > to update the cache
Hi All,
I'm testing dispatcher functions with kamailio 3.0 and using database
to load destination address. The problem I am seeing is the 1'st
destination address in a group is not being used, the dispatcher
always starts at the second database entry. For instance, if I have 2
destination addres
is be extended
to the trusted table?
Thanks.
JR
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> Date: Tue, 15 Jun 2010 16:57:37 -0400
> From: Alex Balashov
> Subject: Re: [SR-Users] sipp to kamailio doesn't match ACK
> To: sr-users@lists.sip-router.org
> Message-ID: <4c17e941.4020...@evaristesys.com>
> Content-Type: text/plain; charset=UTF-8; format=flowed
>
> 1) AFAIK, the Record-Route/R
On Tue, Jun 15, 2010 at 3:44 PM, JR Richardson wrote:
> Hi All,
>
> I have been in the lab testing various load balancing scenarios with
> kamailio 1.5 and 3.0 and distributing calls to asterisk servers. I am
> running into an issue that I'm sure is related to sipp not ka
andle RR headers?
Thanks.
JR
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o
state, then the whole session drops after a few more seconds. I think
there is something going on with the append_branch function that is
throwing me off. It is needed so the failure route completes but it's
not handling the complete dialog or I could be totally off.
Thanks.
JR
--
JR R
Hi All,
I'm using kamailio with carrierroute to load balance calls to other servers.
I have 2 testing scenarios set up:
sipp>http://pastebin.com/Uk9qVhX2
Any ideas on why the call is not maintaining SIP session through the proxy?
Thanks.
JR
--
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Engineering for th
On Thu, May 27, 2010 at 1:24 PM, Uriel Rozenbaum
wrote:
> JR,
>
> Maybe for subsequent routes you'd like to use $oU (original URI Username)
> that is the original number you intend to route.
>
> Rgds,
> Uriel
>
> On Thu, May 27, 2010 at 1:23 PM, JR Richardson
>
On Thu, May 27, 2010 at 9:57 AM, marius zbihlei wrote:
> JR Richardson wrote:
>>
>> On Thu, May 27, 2010 at 2:39 AM, marius zbihlei
>> wrote:
>>
>>>>
>>>> I think the database entries are setup ok, I used the example in the
>>>> mo
On Thu, May 27, 2010 at 10:09 AM, Henning Westerholt
wrote:
> On Thursday 27 May 2010, JR Richardson wrote:
>> > i assume according your description that the GW in question don't send a
>> > provisional response. Then the tm module should generate a internal 408
>>
On Thu, May 27, 2010 at 4:38 AM, Henning Westerholt
wrote:
> On Thursday 27 May 2010, JR Richardson wrote:
>> I am lab testing carrierroute modue on kamailio 1.5.4-notls
>> (i386/linux) and have a question on how to continue processing a call
>> if kamailio sends a call wi
e call will help. ngrep is very usefull (ngrep -W byline
> -d any sip port 5060 does the trick in most cases)
>
> Hope I was not that redundant and the info helps.
>
> Cheers
> Marius
>
Thanks for having a look.
JR
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___
mited several times.
Any ideas?
Thanks.
JR
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t;);
route(1);
}
}
failure_route[2] {
if (t_check_status("408|5[0-9][0-9]")) {
if(!cr_route("1", "2", "$rU", "$rU", "call_id")){
t_reply("403", "Not all
On Wed, May 26, 2010 at 2:23 AM, marius zbihlei wrote:
> JR Richardson wrote:
>>
>> Hi All,
>>
>> I'm testing carrierroute module on version: kamailio 1.5.4-notls
>> (i386/linux).
>>
>> I'm distributing calls to a group of Asterisk servers
cess fine, so does the
carrierroute module handle those requirements for stateful processing?
Thanks.
JR
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On Tue, May 25, 2010 at 11:10 AM, Daniel-Constantin Mierla
wrote:
> Hello,
>
> On 5/22/10 2:22 AM, JR Richardson wrote:
>>
>> On Fri, May 21, 2010 at 4:46 PM, Daniel-Constantin Mierla
>> wrote:
>>
>>>
>>> Hello,
>>>
>&g
On Mon, May 24, 2010 at 2:33 PM, Klaus Darilion
wrote:
>
>
> On 21.05.2010 23:46, Daniel-Constantin Mierla wrote:
>>
>> Hello,
>>
>> On 5/21/10 10:47 PM, JR Richardson wrote:
>>>
>>> Hi All,
>>>
>>> I'm doing some testing w
riciated.
Thanks.
JR
--
JR Richardson
Engineering for the Masses
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
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