Re: [SR-Users] Kamailio does not send reply to source port

[Mino Haluz]

(palmface). I am always forgetting about this, it should be called where
initial request is processed or also for sequential requests?




On Mon, Aug 19, 2013 at 11:39 AM, Daniel-Constantin Mierla <
mico...@gmail.com> wrote:

>  Hello,
>
> do you use force_rport()? Normally, the reply is routed using Via header
> address.
>
> Cheers,
> Daniel
>
>
> On 8/19/13 12:31 PM, Mino Haluz wrote:
>
> Hi,
>
>  I really do not know what is happening but when I'm sending SIP INVITE
> form port other than 5060, if it reaches sl_send_reply("403","Forbidden")
> response is sent to 5060 and not to the source port of the initial INVITE.
>
>  I am not changing anyhow $du or $rd ...
>
>  Mino
>
>
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[SR-Users] Kamailio does not send reply to source port

[Mino Haluz]

Hi,

I really do not know what is happening but when I'm sending SIP INVITE form
port other than 5060, if it reaches sl_send_reply("403","Forbidden")
response is sent to 5060 and not to the source port of the initial INVITE.

I am not changing anyhow $du or $rd ...

Mino
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Re: [SR-Users] kamailio and web services

[Mino Haluz]

Ok guys, seems that no one interested in this, but I decided for http_query
with xmlops module - it gives quite good performance and reliability
(timeout, reply code checking).


On Mon, Jul 29, 2013 at 12:20 PM, Mino Haluz  wrote:

> Oh, I see, there is app_lua as well, so all I found that could be used for
> calling webservices:
>
> app_python
> app_perl
> app_lua
> app_mono
> http_query form utils module
>
> So the question is, what would you use and why? :)
>
>
>
>
> On Mon, Jul 29, 2013 at 10:09 AM, Mino Haluz  wrote:
>
>> Hi,
>>
>> we would like to remove any SQL in our configuration and to fetch all
>> data from web services (REST/RPC/whatever). What do you think, what would
>> be the best way to integrate webservices in kamailio? app_python? app_perl?
>> When calling any method from app_python configured script, it executes
>> everytime the interpreter right? I am afraid this will be a bit slow.
>>
>> M
>>
>
>
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Re: [SR-Users] kamailio and web services

[Mino Haluz]

Oh, I see, there is app_lua as well, so all I found that could be used for
calling webservices:

app_python
app_perl
app_lua
app_mono
http_query form utils module

So the question is, what would you use and why? :)




On Mon, Jul 29, 2013 at 10:09 AM, Mino Haluz  wrote:

> Hi,
>
> we would like to remove any SQL in our configuration and to fetch all data
> from web services (REST/RPC/whatever). What do you think, what would be the
> best way to integrate webservices in kamailio? app_python? app_perl? When
> calling any method from app_python configured script, it executes everytime
> the interpreter right? I am afraid this will be a bit slow.
>
> M
>
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[SR-Users] kamailio and web services

[Mino Haluz]

Hi,

we would like to remove any SQL in our configuration and to fetch all data
from web services (REST/RPC/whatever). What do you think, what would be the
best way to integrate webservices in kamailio? app_python? app_perl? When
calling any method from app_python configured script, it executes everytime
the interpreter right? I am afraid this will be a bit slow.

M
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[SR-Users] Modify Allow header

[Mino Haluz]

Hi,

I need to modify Allow header that is sent from Freeswitch, to restrict it
to only some very important ones. Do you think it really violates RFC ? :))

M
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Re: [SR-Users] Script - /kamailio.log -> MongoDB ?

[Mino Haluz]

http://asylum.madhouse-project.org/projects/syslog-ng/mongodb/


On Thu, Jun 20, 2013 at 12:25 AM, Daniel-Constantin Mierla <
mico...@gmail.com> wrote:

>  Hello,
>
> Kamailio sends the logs to syslog, perhaps you can find something on the
> net that allows syslog application to write to mongodb. I know some
> versions have option to write to a database or send via network to another
> host.
>
> Cheers,
> Daniel
>
>
> On 6/19/13 9:27 PM, Mick Stevens wrote:
>
>  Hi All,
>
>  I'm not a programmer & am new to Kamailio (but have managed to master
> 'ish FreeSWITCH) & am looking for a script I can use/adapt/learn from to
> import kamailio.log into MongoDB.
>
>  Any offers?
>
>  Mick
>
>  Rgds, Mick
> Tel/SMS. +44(0)7967 594432
>  Email/IM/MSN/Skype. mickstev...@yahoo.com
> www.facebook.com/mickstevens
> @mickstevens
>
>
>
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>
> --
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> - http://www.linkedin.com/in/miconda
> Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013
>   * http://asipto.com/u/katu *
>
>
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Re: [SR-Users] Dispatcher behaviour

[Mino Haluz]

Thank you for explanation Daniel.

-- probing is intermediary state, toward active or inactive. So with first
threshold result, it switches to the appropriate state.

but I see no difference between this and this mode below (documentation)

1 (value 2) - temporary trying destination (in the way to become inactive
if it does not reply to keepalives - there is a module parameter to set the
threshold of failures);

So we have

AP - active probing (proxy keeps on sending OPTIONS, routing to this
destination possible)
IP - inactive probing, because it failed "threshold" times (proxy keeps on
sending OPTIONS, routing is not possible)
TP - last check failed, from this state it can go AP or IP (routing is
possible)
DX - disabled (no OPTIONS sent, routing not possible)
AX - active (routing possible, OPTIONS are not sent) ?
TX - ? temporary trying
IX - ? inactive trying

But in fact if you manage to find some time for that NOPROBE thing, the
cases we need could be implemented, it would be awesome.






On Wed, Jun 19, 2013 at 2:44 PM, Daniel-Constantin Mierla  wrote:

>  Hello,
>
>
> On 6/19/13 1:57 PM, Mino Haluz wrote:
>
> Hi,
>
>  I've been playing with dispatcher settings, and I cannot get it working
> as expected. I would like to have my destination IP1 being always probed
> and some destinations should be marked as active non-probing.
>
>  So, in my config I have
>
>  ds_probing_mode = 0  (documentation:  If set to 0, only the gateways
> with state PROBING are tested;)
>
>
> probing is intermediary state, toward active or inactive. So with first
> threshold result, it switches to the appropriate state.
>
>
>
>  Ok, so I set flag = 8 in dispatcher list for IP1 (documentation:  3
> (value 8) - probing destination (sending keep alives);)
>
>  Restart of kamailio, host IP1 is AP, after first interval (10sec), 200OK
> from gateway is received, and host becomes AX (active non-probing). Why ??
> What I expect is that host stays AP.  Why the probing mode is turned off
> when first check is OK?
>
>  I can somehow do it with ds_probing_mode = 1 but this is not I want - I
> need that some gateways are not probed at all (documentation:  if set to
> 1, all gateways are tested - useless for me)
>
>  I tried every flag range 0-10. I think it does not work as documentation
> states or I simply do not understand its meaning.
>
>  Please help me to have this clear, this module is quite critical for us.
>
> Probably it is needed a new flag to say NOPROBE when ds_probing_mode = 1 .
> I will try to look over it when I have some time.
>
> Cheers,
> Danie
>
> --
> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda 
> - http://www.linkedin.com/in/miconda
> Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013
>   * http://asipto.com/u/katu *
>
>
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[SR-Users] Dispatcher behaviour

[Mino Haluz]

Hi,

I've been playing with dispatcher settings, and I cannot get it working as
expected. I would like to have my destination IP1 being always probed and
some destinations should be marked as active non-probing.

So, in my config I have

ds_probing_mode = 0  (documentation:  If set to 0, only the gateways with
state PROBING are tested;)

Ok, so I set flag = 8 in dispatcher list for IP1 (documentation:  3 (value
8) - probing destination (sending keep alives);)

Restart of kamailio, host IP1 is AP, after first interval (10sec), 200OK
from gateway is received, and host becomes AX (active non-probing). Why ??
What I expect is that host stays AP.  Why the probing mode is turned off
when first check is OK?

I can somehow do it with ds_probing_mode = 1 but this is not I want - I
need that some gateways are not probed at all (documentation:  if set to 1,
all gateways are tested - useless for me)

I tried every flag range 0-10. I think it does not work as documentation
states or I simply do not understand its meaning.

Please help me to have this clear, this module is quite critical for us.

Thanks,
Mino
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[SR-Users] Dispatcher mode for Active non-probing mode

[Mino Haluz]

Hi,

sorry, but from documentation I cannot understand what mode should I choose
for dispatcher host, so that it will not be probed at all and marked always
as active. I have sipp on another side, which does not respond to any
OPTIONS, so I want to temporarily disable checking.

Thanks,
Mino
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Re: [SR-Users] Is single line Via with multiple entries allowed?

[Mino Haluz]

Thank you Daniel, even though it was not related, I set it to 1, and
error disappeared. I read RFC about 503 and could not find info, that 503
should be remapped to 500..


On Fri, Jun 14, 2013 at 5:46 PM, Daniel-Constantin Mierla  wrote:

>  Hello,
>
> they are not related, the warning is usually for requests, but is harmless
> anyhow.
>
> Changing 503 in 500 is from RFC specs, but you can disable it:
>
> http://kamailio.org/docs/modules/stable/modules/tm.html#remap_503_500
>
> Cheers,
> Daniel
>
>
> On 6/14/13 5:42 PM, Mino Haluz wrote:
>
> Hi,
>
>  I'm struggling to solve one problem with 503 reply code.
>
>  Kamailio receives 503 and changes it to 500, with error in log
>
>  WARNING: script writer didn't release transaction
>
>  I inspected incoming 503 (from Cisco gateway), everything seems to be ok
> except for Via:
>
>  SIP/2.0/UDP localip:5060;branch=z9hG4bK47e3.c400aa13.0,SIP/2.0/UDP
> ip2:5060;branch=z9hG4bK37170104;rport=5060
>
>  Does kamailio support this kind of Via?
>
>  Thanks,
> Mino
>
>
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>
> --
> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda 
> - http://www.linkedin.com/in/miconda
> Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013
>   * http://asipto.com/u/katu *
>
>
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[SR-Users] Is single line Via with multiple entries allowed?

[Mino Haluz]

Hi,

I'm struggling to solve one problem with 503 reply code.

Kamailio receives 503 and changes it to 500, with error in log

WARNING: script writer didn't release transaction

I inspected incoming 503 (from Cisco gateway), everything seems to be ok
except for Via:

SIP/2.0/UDP localip:5060;branch=z9hG4bK47e3.c400aa13.0,SIP/2.0/UDP
ip2:5060;branch=z9hG4bK37170104;rport=5060

Does kamailio support this kind of Via?

Thanks,
Mino
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Re: [SR-Users] kamailio and virtual IP

[Mino Haluz]

Ok, this works, but dispatcher module keeps on sending OPTIONS from
non-existent interface and it generates lots of errors, that I do not like
much :) These two hosts share one dispatcher table, so I cannot set it to
active non-probing mode.


On Fri, Jun 14, 2013 at 10:56 AM, Alex Balashov
wrote:

> Hello Mino,
>
>
> On 06/14/2013 04:53 AM, Mino Haluz wrote:
>
>  If I bind to non-existent IP address, than there is problem with
>> sending OPTIONS.
>>
>
> Perhaps this can help?
>
>echo 1 > /proc/sys/net/ipv4/ip_**nonlocal_bind
>
> This will allow you to bind to IPs that don't correspond to any existing
> interfaces right now.
>
> --
> 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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Re: [SR-Users] kamailio and virtual IP

[Mino Haluz]

I am always witnessing the fact, that administration of 3rd party HA
software cost you more problems, than coding your own system which you
understand deeply.

That's why I've done RPC services in python which do the job for me and
works in much more flexible way, than manual editing of XML files of
heartbeat (which I consider a bit pain in ...).

If I bind to non-existent IP address, than there is problem with sending
OPTIONS.

So i gave up this, it will run/kill kamailio while taking up/down virtual
interface.




On Fri, Jun 14, 2013 at 9:32 AM, phillman25  wrote:

> Hi Mino
>
> I have set up this scenario using heartbeat on both hosts and it works
> wonderfully.
>
> Regards
> Phillip
>
>
>
>
>
>
>
> Message: 1
> Date: Fri, 14 Jun 2013 08:15:28 +0200
> From: Mino Haluz 
> To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) -
> Users   Mailing List"  >
> Subject: [SR-Users] kamailio and virtual IP
> Message-ID:
>  zb0wfzrtszew6fy5tjhqza...@mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Hi,
>
> I want to do this:
>
> 2 hosts, with 2 running kamailios, every host has 1 IP address and hostA
> has virtualIP assigned. kamailio should run on both hosts. I made a script
> which can transfer virtual IP from hostA to hostB.
>
> The problem is, I cannot tell kamailio to use virtualIP on hostB because,
> when I set
>
> listen=virtualIP:5060 - it cannot start because virtualIP is not present
> listen=0.0.0.0:5060 and force_send_socket(virtualIP) - it does not work as
> virtualIP is not in the listen list
>
> Some hint? I could run kamailio automatically after chaning IP address, but
> it requires time to load, so it will not be as quick as I change simply the
> owner of virtual IP.
>
> Thanks,
> Mino
>
>
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Re: [SR-Users] config test

[Mino Haluz]

kamailio -c


On Fri, Jun 14, 2013 at 9:29 AM, Victor V. Kustov  wrote:

> Hi all!
>
> Is check config without restart possible?
>
> --
>  WBR, Victor
>   JID: coy...@bks.tv
>   JID: coy...@bryansktel.ru
>   I use FREE operation system: 3.9.4-calculate GNU/Linux
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[SR-Users] kamailio and virtual IP

[Mino Haluz]

Hi,

I want to do this:

2 hosts, with 2 running kamailios, every host has 1 IP address and hostA
has virtualIP assigned. kamailio should run on both hosts. I made a script
which can transfer virtual IP from hostA to hostB.

The problem is, I cannot tell kamailio to use virtualIP on hostB because,
when I set

listen=virtualIP:5060 - it cannot start because virtualIP is not present
listen=0.0.0.0:5060 and force_send_socket(virtualIP) - it does not work as
virtualIP is not in the listen list

Some hint? I could run kamailio automatically after chaning IP address, but
it requires time to load, so it will not be as quick as I change simply the
owner of virtual IP.

Thanks,
Mino
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[SR-Users] ds_is_from_list does not work in my code

[Mino Haluz]

Hi,

what this function in fact does? It only compares host from dispatcher list
and $si ? or it compares with source port as well ?

I'm distributing traffic across multiple gateways with dispatcher module,
these gateways create another leg and send it back to proxy. I would like
to compare it with my dispatcher list if the source IP of the request is
one of the dispatcher list.

So I cannot use this function, right? Any other functions that can do
something similar?

Mino
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Re: [SR-Users] How to lock hashtable variable

[Mino Haluz]

Ok so this is my case:

I have rtimer which checks for shv(startload) variable. If it is set, it
will run LOAD_DATA route which deletes the hashtable a fills with fresh
data from database. These hashtables contains prefixes and some custom
data. So I'm afraid when doing an update, data can be inconsistent a while.

If rtimer runs a route, any other processing is stopped until this route
ends? If so, I do not need any lock.


On Thu, Jun 13, 2013 at 9:15 AM, Daniel-Constantin Mierla  wrote:

> Hello,
>
>
> On 6/12/13 9:04 AM, Mino Haluz wrote:
>
>> Hi,
>>
>> I need to lock the hashtable but this
>>
>> lock("$sht(a)");
>>
>> does not work. Any hints?
>>
> htable locks are inside the code and per slot, not per hash table, why you
> would need to lock it?
>
> For synchronous access in config file, you can use the locks from cfg
> utils.
>
> Cheers,
> Daniel
>
> --
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> http://twitter.com/#!/miconda - 
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>
>
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[SR-Users] How to define variable as string explicitly

[Mino Haluz]

Hi,

I need to define variables as string explicitly.

What I have in the code:

in prefix_list is 123,456,789
$var(matched_prefix) = $(var(prefix_list){s.select,$var(i),,});

then in MAIN route:

$var(matched_prefix) = route(INCOMING_AUTH);

and this check

if ($var(matched_prefix) != "nullprefix")

returns:

WARNING:  [rvalue.c:1012]: automatic string to int conversion for
"nullprefix" failed
WARNING:  [rvalue.c:1916]: rval expression conversion to int failed
(153,32-153,43)

I would like to have all values as string to avoid these errors.

Thanks,
Mino
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[SR-Users] How to lock hashtable variable

[Mino Haluz]

Hi,

I need to lock the hashtable but this

lock("$sht(a)");

does not work. Any hints?

Mino
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[SR-Users] To simply run a route from externally

[Mino Haluz]

Hi,

I need to update some hash tables inside kamailio, but this update should
be triggered externally.

I know there is xmlrpc or xhttp, but I am just curious if there is some
easier way how to run this route. I have to run it instantly, so setting
some shv and run this route if new invite comes is not possible, nor timer.

Thanks,
Mino
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[SR-Users] How to iterate comma separated string

[Mino Haluz]

Hi,

I have to iterate string like 1,2,3,4 in kamailio. How could you do this? I
know we have s.select,index,separator but I dont know how to go through all
elements. Thanks,

Mino
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[SR-Users] How does save(location) choose the subscriber

[Mino Haluz]

Hi,

if I do save(location) when receiving REGISTER, what is the header which
indicates the subscriber for which it will be registered ?

$fu?$au?$tu?

Thanks,
Mino
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[SR-Users] Problem with frequent use of FIFO commands

[Mino Haluz]

Hi,

I have some nagios scripts binded to kamctl fifo commands, as well as some
web scripts that are directly calling kamctl fifo statements. I do not know
why, but it sometimes returns

** ERROR: Error opening Kamailio's FIFO /tmp/kamailio_fifo [1] => ** ERROR:
Make sure you have the line 'modparam("mi_fifo", "fifo_name",
"/tmp/kamailio_fifo")' in your config [2] => ** ERROR: and also have loaded
the mi_fifo module.

when it is called with web script with www-data account. From shell it
works everytime, but with web 20% of results are errors.  Other 80% are
sucessful, so no problem with permissions.

Does mi_fifo support multiple access, are there any other limits in this
regard ? Within OS level?

kamailio-3.3.4

Thanks,
Mino
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Re: [SR-Users] How to remove custom header after adding one

[Mino Haluz]

msg_apply_changes made a little mess in every SIP message, so I am trying
to avoid it at all cost..

rpid manipulation in branch route worked like a charm, thanks.


On Sun, Feb 24, 2013 at 7:40 PM, Marius Zbihlei  wrote:

> Hello,
>
> I can see two variants:
> 1. Move the append_hf in the branch route so you can have a per branch
> decision
> 2. call mg_apply_change() from textops to immediately apply the changes
> after append_hf()
>
> http://kamailio.org/docs/modules/devel/modules/textopsx.html#textopsx.msg_apply_changes
>
> You can call remove_hf() on the changed message.
>
> Cheers,
> Marius
>
>
> On Sun, Feb 24, 2013 at 6:19 PM, Javi Gallart 
> wrote:
>
>> Hello
>>
>> you may try to do the remove/add operations in each branch_route
>>
>> Regards
>>
>> Javi
>> -Original Message-
>> From: Mino Haluz 
>> Sender: sr-users-boun...@lists.sip-router.org
>> Date: Sun, 24 Feb 2013 19:06:29
>> To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users
>>Mailing List
>> Reply-To: "SIP Router - Kamailio \(OpenSER\) and SIP Express Router
>> \(SER\) -
>> Users Mailing List" 
>> Subject: [SR-Users] How to remove custom header after adding one
>>
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[SR-Users] How to remove custom header after adding one

[Mino Haluz]

Hi,

if I use append_hf() for adding my custom header to initial request, how I
can remove this header later ?

What I'm doing:

1) initial request: remove RPID header with remove_hf
2) initial request: add custom RPID header append_hf
3) after rerouting: in failure_route remove RPID header remove_hf
4) after rerouting: in failure_route add custom RPID header with append_hf

The result of this, are 2 RPID headers, because remove_hf did not catch
RPID header added in step 2, but I think the original one that was removed
already in step 1.

Thanks,
Mino
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Re: [SR-Users] Difference between shared and script variable

[Mino Haluz]

Thank you very much for this.

One question regarding $sht. Does it need locking like $shv does?


On Thu, Feb 14, 2013 at 8:35 AM, Daniel-Constantin Mierla  wrote:

>  Hello,
>
> $shv(...) is referred as shared memory variable because it stores the
> value in shared memory. That means if you set $shv(x) in one process, you
> can read its value from another process. You have to be sure you don't have
> races in setting the variable, that could be achieved with locks from
> cfgutils.
>
> $var(...) is referred as private memory variable because it stores the
> value in private memory. That means its value is valid in the context of
> the same process (e.g., use it while processing the same sip message on a
> single routing block type, like running the main request route block, or
> reply route block, etc). It is not safe to use it for transactions, like
> setting it in request route block and reading it in failure route block
> (use avps for that case).
>
> $var(...) is faster to use and does not need locking at all. These are
> usually referred as script variable, but this term can be confused with all
> the config file variables.
>
> Cheers,
> Daniel
>
>
> On 2/12/13 3:13 PM, Mino Haluz wrote:
>
> Hi,
>
>  what is the difference between shared and script variable ? Thanks
>
>  Mino
>
>
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> --
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> - http://www.linkedin.com/in/miconda
> Kamailio World Conference, April 16-17, 2013, Berlin
>  - http://conference.kamailio.com -
>
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Re: [SR-Users] kamctl restart and mediaproxy

[Mino Haluz]

You have to figure it somehow with adjusting the python code of
media-dispatcher. We have developed special "/etc/init.d/kamailio grace"
command, which waits for the moment, when there are no calls, so it can be
restarted. It queries mediaproxy every second.

In future we will not use mediaproxy anymore - it is not supported in
latest version either. rtpproxy gives us better performance results.

On Tue, Feb 12, 2013 at 2:44 PM, Eduardo Lejarreta
wrote:

> Good evening.
>
> Launching "mediaproxy-dispatcher" and "mediaproxy-relay" on the foreground
> and with "DEBUG" we can see:
>
> mediaproxy-dispatcher log:
> --> Kamctl
> restart
> debug: Connection to OpenSIPS lost: Connection was closed cleanly.
> debug: Connection to OpenSIPS lost: Connection was closed cleanly.
> debug: Connection to OpenSIPS lost: Connection was closed cleanly.
> debug: Issuing "remove" command to relay at A.B.C.D
> debug: Got statistics: {xxx}
> debug: Connection to OpenSIPS lost: Connection was closed cleanly.
>
> mediaproxy-relay log:
> --> Kamctl
> restart
> debug: removing session xxx
> (Port 5 Closed)
> (Port 50001 Closed)
> (Port 50002 Closed)
> (Port 50003 Closed)
>
> So "mediaproxy-dispatcher" doesn't receive any command from Kamailio, he
> just detects " Connection was closed cleanly" and then issues a "remove"
> command to "mediaproxy-relay".
>
> Does anybody know any way to avoid this?
>
> Thanks and regards.
> --
> Eduardo Lejarreta.
>
>
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[SR-Users] Difference between shared and script variable

[Mino Haluz]

Hi,

what is the difference between shared and script variable ? Thanks

Mino
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[SR-Users] event_route for pre-dialog start

[Mino Haluz]

Hi,

I think that it is more than likely not implemented, but is there any
event-route that is triggered just before event_route[dialog:start] ? I
need to check some security flags before the dialog is created. But it is
too early too check them in relay route..

Mino
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Re: [SR-Users] Is kamailio creating multiple connections to db?

[Mino Haluz]

Yes, mysql cluster. It seems to be 2x slower than mysqld myisam, but who
cares, 3000q/s is ok for me.


On Mon, Jan 28, 2013 at 10:10 AM, Daniel-Constantin Mierla <
mico...@gmail.com> wrote:

>  Adding references to two solutions:
> - shared IP with cross replication between mysql servers
> - mysql cluster
>
> Cheers,
> Daniel
>
>
> On 1/25/13 8:18 PM, Rumen Mihailov wrote:
>
> Hello Mino,
>
> Sorry for the offtopic, what HA database solution do you use ?
>
> Regards,
> Rumen
> On 25 Jan 2013 16:22, "Mino Haluz"  wrote:
>
>> Hi,
>>
>>  I made some DB test on our HA cluster, and I got that for 1 process I
>> have 1000 queries/sec and for 11 processes 3000 queries/sec. That's why I
>> would to ask if every kamailio process that is spawned has its own database
>> connection or queries are put in the queue and executed by some main
>> process ? I would like to know this, so that I could assess maximum call
>> count/sec that db can handle.
>>
>>  Thank you,
>> Mino
>>
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>
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> - http://www.linkedin.com/in/miconda
> Kamailio World Conference, April 16-17, 2013, Berlin
>  - http://conference.kamailio.com -
>
>
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[SR-Users] Is kamailio creating multiple connections to db?

[Mino Haluz]

Hi,

I made some DB test on our HA cluster, and I got that for 1 process I have
1000 queries/sec and for 11 processes 3000 queries/sec. That's why I would
to ask if every kamailio process that is spawned has its own database
connection or queries are put in the queue and executed by some main
process ? I would like to know this, so that I could assess maximum call
count/sec that db can handle.

Thank you,
Mino
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Re: [SR-Users] How to correctly check if the variable is set?

[Mino Haluz]

and how to check null value taken from database?

$dbr(ra=>[0,0]) == ?


On Mon, Jan 14, 2013 at 1:44 PM, Alex Balashov wrote:

> if(defined $var(x))
>
> Or, if checking for empty value:
>
> if(strempty($var(x))
>
> Mino Haluz  wrote:
>
> >Hi,
> >
> >how should I check if the value is set?
> >
> >if ($avp(s:test) == "") {
> >
> >or is there any null keyword ? If so, does it work for $avp, $sht, $var
> >and
> >$shv ?
> >
> >Thanks,
> >
> >Mino
> >
> >
> >
> >
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> --
> 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: +1-678-954-0670
> Web: http://www.evaristesys.com/, http://www.alexbalashov.com/
>
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[SR-Users] How to correctly check if the variable is set?

[Mino Haluz]

Hi,

how should I check if the value is set?

if ($avp(s:test) == "") {

or is there any null keyword ? If so, does it work for $avp, $sht, $var and
$shv ?

Thanks,

Mino
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Re: [SR-Users] How to set variable externally

[Mino Haluz]

Ok I did it like this, every xlog I'm calling is in the form:

xlog("L_INFO","XLOG: $ci [ROUTENAME] debuginformation");

and also after every sql_query I have

xlog("L_INFO","XLOG: $ci [ROUTENAME] SQL: select * from ");

and when SIP message is received/sent

xlog("L_INFO","XLOG: $ci [MAIN] $mb");
xlog("L_INFO","XLOG: $ci [event:onsend_route] $snd(buf)");

So when I'm vizualizing the siptrace, I can view all SQL queries, raw
packets (it is done already by siptrace however) and additional debug logs
associated to one particular callid.

The question is, has xlog maximum length of the second parameter ? Because
when SIP message length exceed some value (around 1500bytes, maybe some
relation to packet max size) it is stripped.
So when calling xlog("L_INFO","XLOG: $ci [event:onsend_route] START
$snd(buf) END"); and $snd(buf) is too big, it is stripped and END is not in
the logs.

Are you aware of maximum length of this xlog message parameter? I cannot
find it in documentation.

Mino.


On Fri, Dec 21, 2012 at 5:18 PM, Daniel-Constantin Mierla  wrote:

>
> On 12/21/12 5:15 PM, Olle E. Johansson wrote:
>
>> 21 dec 2012 kl. 17:10 skrev Ovidiu Sas :
>>
>>  If you just want to control the debug level externally, take a look at
>>> the debug parameter:
>>> http://www.kamailio.org/wiki/**cookbooks/3.3.x/core#debug
>>> It can be controlled via sercmd (kamcmd in future versions).
>>>
>>> If you want to play with global flags, take a look at cfgutils:
>>> http://kamailio.org/docs/**modules/3.3.x/modules_k/**cfgutils.html
>>> http://kamailio.org/docs/**modules/3.3.x/modules_k/**
>>> cfgutils.html#id2533439
>>> http://kamailio.org/docs/**modules/3.3.x/modules_k/**
>>> cfgutils.html#id2494518
>>> http://kamailio.org/docs/**modules/3.3.x/modules_k/**
>>> cfgutils.html#id2494559
>>>
>>
>> For your own logs:
>>
>> I usually declare a shared variable called "debug" in the pv module
>> modparams.
>>
>> When logging i do
>>
>> if ($shv(debug) > 0)
>> xlog()
>>
>> All shared variables, like Daniel said, can be changed with kamctl/sercmd.
>>
> More for the sake of archiving, one can even do:
>
> xlog("$shv(debug)", ...);
>
> and if $shv(debug) > than the value of debug parameter, then the message
> is not printed, otherwise will be printed to the level corresponding to its
> value.
>
> Cheers,
> Daniel
>
>
> --
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> http://twitter.com/#!/miconda - 
> http://www.linkedin.com/in/**miconda
>
>
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Re: [SR-Users] How to set variable externally

[Mino Haluz]

Ok I see that it is possible with pv module and shv_set MI command. How can
I get value of this shared variable inside code?


On Fri, Dec 21, 2012 at 1:17 PM, Mino Haluz  wrote:

> Hi,
>
> I would like to set my custom different debug levels (with flag?)
> externally with kamctl command. So I neednt restart kamailio if I want to
> enable/disable debug.
>
> Which module should I use in that case?
>
> Thanks,
> Mino
>
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[SR-Users] How to set variable externally

[Mino Haluz]

Hi,

I would like to set my custom different debug levels (with flag?)
externally with kamctl command. So I neednt restart kamailio if I want to
enable/disable debug.

Which module should I use in that case?

Thanks,
Mino
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[SR-Users] Send reply with Reason header

[Mino Haluz]

Hi,

I know how to use sl_send_reply, but I would like to add "Reason:" header
to the reply generated. Is there any command or variable for that?

Thank you,
Mino
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[SR-Users] Unable to set custom RPID

[Mino Haluz]

Hi,

what things should I set in order to set my custom RPID. What I am doing:

remove_hf("Remote-Party-ID");
append_rpid_hf("", ";party=calling;id-type=subscriber;screen=no");

but RPID is only removed, second command does not append any rpid. Thanks

Mino
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Re: [SR-Users] Eavesdropping SRTP sessions

[Mino Haluz]

Ok so what I understand from the document - there are in fact only these
possibilities, how to be sure there is not Mitm.

1) To use ZRTP for media encryption with SIP TLS (in case proxy is
compromised, attacker can not still decrypt ZRTP even though it goes
through the proxy)
2) To use IPSec for media between the clients (can be SIP or SIPS, does not
matter) if media goes directly between clients
3) To use SRTP with other key management (MIKEY, SDES) ?

When using these ways, audio could be decrypted

1) SRTP with SIP (keys are exchanged in SDP, so if they are not encrypted,
SRTP loses its sense)
2) SRTP with SIPs (if the proxy is hacked, SIPs packet are decrypted on
proxy and SDP payload can be seen, and SRTP packets can be decrypted)

Right?


On Tue, Nov 27, 2012 at 11:21 PM, Jesús Pérez Rubio
wrote:

> I forgot something, with Kamailio default configuration media goes always
> directly between clients. Moreover, if you want to be sure that any
> endpoint is who it says to be you should use client side autentication for
> SIP protocol. TLS module documentation clears how to do it.
>
> http://kamailio.org/docs/modules/devel/modules/tls.html
>
>
>
> 2012/11/27 Jesús Pérez Rubio 
>
>> Hi, If you are using SRTP your conversations will be encrypted, so nobody
>> could eavesdrop it. Only if  your Kamailio was compromised they could be
>> eavesdropped.
>>
>> I think you are confusing SRTP (media) with signaling (SIP). You should
>> implement SIP over TLS too, it makes no sense to use SRTP without encrypt
>> signaling. If not, it could be possible to sniff conversations with a MiTM
>> but, anyway, I don't know any tool which supports it.
>>
>> Here I speak a bit about VoIP encryption, I think it could help you:
>>
>> http://nicerosniunos.blogspot.com.es/2011/08/voip-eavesdropping-counter-measurements.html
>>
>> Best regards.
>>
>>
>>
>> 2012/11/27 Mino Haluz 
>>
>>>  Hi,
>>>
>>> maybe it is not that kamailio related question, but I dont know any
>>> other place with such good voip professionals ;) I have kamailio and
>>> mediaproxy. Clients are BudgetTone 200 (Grandstream) and CSipSimple. I am
>>> forcing clients to use SRTP but it does not support adding any certificate
>>> on both sides. SRTP call is working fine.
>>>
>>> The question is, in this case, is man-in-the-middle attack possible?
>>> Maybe I should study SRTP more, but basically, if there are no
>>> certificates, there is no method how to be 100% sure that the media goes
>>> directly between clients. Is it true?
>>>
>>> Thanks for response,
>>> Mino
>>>
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>>>
>>
>>
>> --
>> Jesús Pérez
>> VoIP Engineer at Quobis
>>
>> Fixed: +34 902 999 465
>> Site: http://www.quobis.com
>>
>>
>
>
> --
> Jesús Pérez
> VoIP Engineer at Quobis
>
> Fixed: +34 902 999 465
> Site: http://www.quobis.com
>
>
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[SR-Users] Eavesdropping SRTP sessions

[Mino Haluz]

Hi,

maybe it is not that kamailio related question, but I dont know any other
place with such good voip professionals ;) I have kamailio and mediaproxy.
Clients are BudgetTone 200 (Grandstream) and CSipSimple. I am forcing
clients to use SRTP but it does not support adding any certificate on both
sides. SRTP call is working fine.

The question is, in this case, is man-in-the-middle attack possible? Maybe
I should study SRTP more, but basically, if there are no certificates,
there is no method how to be 100% sure that the media goes directly between
clients. Is it true?

Thanks for response,
Mino
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[SR-Users] How to get dialog variable externally?

[Mino Haluz]

Hi,

I'm using dialog module and I set some dlg_var for every dialog. Is it
possible to get this variable value with MI command somehow ? I know to get
list of dialogs using dlg_list only. Thanks!

Mino
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Re: [SR-Users] Dispatcher question

[Mino Haluz]

Ok so it can be done according to these scenarios:

Initial invite -- uas -> dispatcher -> uas
200 ok to this invite -- uas -> dispatcher -> uac
re-invite (hold) -- uac -> uas
bye -- uac -> uas

Question, is there any command that forces kamailio to send packet to the
destination disregarding what is in the $rd ?




On Fri, Nov 9, 2012 at 5:37 PM, Alex Balashov wrote:

> On 11/09/2012 11:35 AM, Alex Balashov wrote:
>
>  If you don't want sequential (in-dialog) requests, such as BYEs,
>> reinvites, UPDATEs, etc. to go between the UAs directly and not through
>> the proxy, you can do that: just don't add a Record-Route header (take
>> out the record_route() call, if you have it).
>>
>
> Correction: if you DO want sequential requests [...]
>
> I had originally meant to formulate this sentence differently, and left
> that in there.
>
>
> --
> Alex Balashov - Principal
> Evariste Systems LLC
> 235 E Ponce de Leon Ave
> Suite 106
> Decatur, GA 30030
> Tel: +1-678-954-0670
> Fax: +1-404-961-1892
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>
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[SR-Users] Dispatcher question

[Mino Haluz]

Hi,

I have simple dispatcher that is dispatching initial invites to the core
router. The question is, how can I tell kamailio to not add the Via header
in the INVITE. I mean, I want that this dispatcher will be completely
transparent to the core router, so any responses will go directly to the
original sender, not to the dispatcher. This is my request route:

request_route {

if (is_method("INVITE")) {
if (!ds_select_domain(1, 0)) {
xlog("L_INFO", "[MAIN] ERROR: Proxy1 failed");
 if (!ds_select_domain(2, 0)) {
xlog("L_INFO", "[MAIN] FATAL: Proxy2 failed");
exit;
}
}
xlog("L_INFO","[MAIN] Dispatching INVITE to $rd");
forward();
}
}

1 and 2 are defined in dispatcher.list. If it can be done in kamailio,
please how. Thank you!

Mino
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[SR-Users] siptrace vs. sipcapture

[Mino Haluz]

Hi,

we are thinking about switching from siptrace to sipcapture + HEP
encapsulation. Do you know some reasons why we should switch to
sipcapture or why we should not ? ;) We have multiple SIP proxies and
Cisco gateways. In earlier versions of kamailio was siptrace blocking,
that's why we started to looking for some other capturing mechanism. I
think now siptrace is buffered, so we are hesitating ..

Mino

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Re: [SR-Users] Force to use mediaproxy/rtpproxy

[Mino Haluz]

My fault, typoo in dispatcher's IP address.

On Tue, Oct 9, 2012 at 3:25 PM, Mino Haluz  wrote:
> Hi,
>
> I am doing some performance tests and I cannot force SIP INVITEs to
> use mediaproxy within LAN, without any NAT. I'm using
> engage_media_proxy or rtpproxy_manage but phones are still exchanging
> voice data between them, not through the proxy. Is there some trick
> how to forcibly tell kamailio to use media/rtp proxy ? Maybe kamailio
> finds out that there is no nat and therefore will not use any nat
> traversal mechanism.
>
> Mino

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[SR-Users] Force to use mediaproxy/rtpproxy

[Mino Haluz]

Hi,

I am doing some performance tests and I cannot force SIP INVITEs to
use mediaproxy within LAN, without any NAT. I'm using
engage_media_proxy or rtpproxy_manage but phones are still exchanging
voice data between them, not through the proxy. Is there some trick
how to forcibly tell kamailio to use media/rtp proxy ? Maybe kamailio
finds out that there is no nat and therefore will not use any nat
traversal mechanism.

Mino

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Re: [SR-Users] How to do RTPProxy load balancing?

[Mino Haluz]

If one rtpproxy fails, kamailio will stop dispatching to this destination?

On Fri, Sep 14, 2012 at 9:59 AM, Jon Bonilla  wrote:
> El Fri, 14 Sep 2012 09:40:20 +0200
> Mino Haluz  escribió:
>
>> Hi,
>>
>> mediaproxy has dispatcher and relay, so it load balances the traffic
>> automatically. How it is done with rtpproxy ?
>>
>
> Via modparams I think
>
> http://kamailio.org/docs/modules/3.3.x/modules/rtpproxy.html#id2549966
>
>
>
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[SR-Users] How to do RTPProxy load balancing?

[Mino Haluz]

Hi,

mediaproxy has dispatcher and relay, so it load balances the traffic
automatically. How it is done with rtpproxy ?

Thanks,
Mino

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Re: [SR-Users] RTPProxy with kamailio : How to get calls count?

[Mino Haluz]

CPU E5504 Xeon 4MB cache 2GHz
800 concurrent calls with 92% CPU usage
So 400-450 calls for 1GHz (with 99% CPU usage, in fact I cannot get it
to 100% dunno why)

They had 325 calls per 1GHz, so yes, there must be some improvement ;)

On Thu, Sep 13, 2012 at 7:22 PM, Alex Balashov
 wrote:
> I'd be curious to know if your performance might actually be higher, owing
> to improved CPUs since the study was done, hyperthreading, more frames per
> NIC interrupt, etc.
>
>
>
>
> -- Alex
>
> --
> Sent from my Samsung mobile, and thus lacking in the refinement one might
> expect from a proper keyboard.
>
> Alex Balashov - Principal
> Evariste Systems LLC
> 235 E Ponce de Leon Ave
> Suite 106
> Decatur, GA 30030
> Tel: +1-678-954-0670
> Web: http://www.evaristesys.com/
>
> Mino Haluz  wrote:
> Thank you, setting ulimits worked! And the performance is the same as
> stated in the document I mentioned! :)
>
> One more thing, I found these errors in syslog:
>
> Sep 13 18:38:18 perftest kamailio[5268]: ERROR: 
> [parser/sdp/sdp.c:211]: Invalid payload location
> Sep 13 18:38:18 perftest kamailio[5268]: ERROR: 
> [parser/sdp/sdp.c:227]: Invalid payload location
> Sep 13 18:38:18 perftest kamailio[5270]: ERROR: 
> [parser/sdp/sdp.c:227]: Invalid payload location
>
> This is possibly something related to my sipp scenario, this is the
> INVITE sent from sipp
>
>   
>
> Do you see something that could cause this error ? Otherwise the call
> is initiated ok, but I really dont understand what is so strange to
> kamailio in this INVITE.
>
> On Thu, Sep 13, 2012 at 6:37 PM, Peter Lemenkov  wrote:
>> 2012/9/13 Mino Haluz :
>>> You mean on the proxy side? I'm running rtpproxy as root, limits are
>>> still applied ? ulimit -s unlimited should do the trick ?
>>
>> Yes, they usually applied even for superuser, and yes - this should
>> help (if that's the issue).
>>
>>
>> --
>> With best regards, Peter Lemenkov.
>>
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Re: [SR-Users] RTPProxy with kamailio : How to get calls count?

[Mino Haluz]

Thank you, setting ulimits worked! And the performance is the same as
stated in the document I mentioned! :)

One more thing, I found these errors in syslog:

Sep 13 18:38:18 perftest kamailio[5268]: ERROR: 
[parser/sdp/sdp.c:211]: Invalid payload location
Sep 13 18:38:18 perftest kamailio[5268]: ERROR: 
[parser/sdp/sdp.c:227]: Invalid payload location
Sep 13 18:38:18 perftest kamailio[5270]: ERROR: 
[parser/sdp/sdp.c:227]: Invalid payload location

This is possibly something related to my sipp scenario, this is the
INVITE sent from sipp

  

Do you see something that could cause this error ? Otherwise the call
is initiated ok, but I really dont understand what is so strange to
kamailio in this INVITE.

On Thu, Sep 13, 2012 at 6:37 PM, Peter Lemenkov  wrote:
> 2012/9/13 Mino Haluz :
>> You mean on the proxy side? I'm running rtpproxy as root, limits are
>> still applied ? ulimit -s unlimited should do the trick ?
>
> Yes, they usually applied even for superuser, and yes - this should
> help (if that's the issue).
>
>
> --
> With best regards, Peter Lemenkov.
>
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Re: [SR-Users] RTPProxy with kamailio : How to get calls count?

[Mino Haluz]

You mean on the proxy side? I'm running rtpproxy as root, limits are
still applied ? ulimit -s unlimited should do the trick ?

On Thu, Sep 13, 2012 at 6:25 PM, Peter Lemenkov  wrote:
> Hello!
>
> 2012/9/13 Mino Haluz :
>> The results:
>>
>> - rtpproxy calls count 280
>> - sipp calls count 2000
>> - iptraf on proxy 4.8MB/s
>> - G711a codec
>>
>> So if my calculations are right (16kB/s per stream * 280 = 4.5MB/s),
>> rtpproxy calls count is really the right value. CPU usage is ok on
>> every machine (rtpproxy 20-30% CPU). Does anybody know why rtpproxy
>> cannot serve more than 270-280 calls ?
>
> Let me be a dumb idiot here but did you set ulimits properly? 270-280
> is pretty close to 256 (1024 / 4 ports).
> --
> With best regards, Peter Lemenkov.
>
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Re: [SR-Users] RTPProxy with kamailio : How to get calls count?

[Mino Haluz]

According to this
(http://transnexus.com/index.php/performance-test-results-for-openser-and-rtpproxy)

"For a server hosting both OpenSER and RTPproxy, each 1 GHz of CPU processing
capacity can manage a maximum of 325 simultaneous calls."

I have 2.4GHz for rtpproxy, but CPU/Mem/network is ok, so the
bottleneck should be somewhere else probably..

On Thu, Sep 13, 2012 at 5:56 PM, Alex Balashov
 wrote:
> I'm not sure what a single instance of rtpproxy can handle, but most people
> squeezing thousand of concurrent calls per box are probably doing it on
> multicore boxes by binding multiple instances of rtpproxy with different
> core affinities, and round-robining among them.
>
>
>
>
> -- Alex
>
> --
> Sent from my Samsung mobile, and thus lacking in the refinement one might
> expect from a proper keyboard.
>
> Alex Balashov - Principal
> Evariste Systems LLC
> 235 E Ponce de Leon Ave
> Suite 106
> Decatur, GA 30030
> Tel: +1-678-954-0670
> Web: http://www.evaristesys.com/
>
> Mino Haluz  wrote:
> The results:
>
> - rtpproxy calls count 280
> - sipp calls count 2000
> - iptraf on proxy 4.8MB/s
> - G711a codec
>
> So if my calculations are right (16kB/s per stream * 280 = 4.5MB/s),
> rtpproxy calls count is really the right value. CPU usage is ok on
> every machine (rtpproxy 20-30% CPU). Does anybody know why rtpproxy
> cannot serve more than 270-280 calls ?
>
> On Thu, Sep 13, 2012 at 5:07 PM, Mino Haluz  wrote:
>> Ok, so I put there unforce_rtp_proxy even though I'm using
>> rtpproxy_manage. The tip with nc now really shows the calls count.
>>
>> But the dialog count is still higher and higher, so I have bug
>> somewhere in the configuration. I'll check it.
>>
>> On Thu, Sep 13, 2012 at 4:53 PM, Alex Balashov
>>  wrote:
>>> Correct, but you still need to call rtpproxy_manage() on receipt of a BYE
>>> or
>>> CANCEL. It'll just figure out what to do on its own.
>>>
>>> None of this has to do with dialog state, though. Just rtpproxy control.
>>>
>>>
>>>
>>>
>>> -- Alex
>>>
>>> --
>>> Sent from my Samsung mobile, and thus lacking in the refinement one might
>>> expect from a proper keyboard.
>>>
>>> Alex Balashov - Principal
>>> Evariste Systems LLC
>>> 235 E Ponce de Leon Ave
>>> Suite 106
>>> Decatur, GA 30030
>>> Tel: +1-678-954-0670
>>> Web: http://www.evaristesys.com/
>>>
>>> Mino Haluz  wrote:
>>> I'm using rtpproxy_manage, so I assume unforce_rtp is not needed.
>>>
>>> On Thu, Sep 13, 2012 at 4:10 PM, Peter Lemenkov 
>>> wrote:
>>>> 2012/9/13 Mino Haluz :
>>>>
>>>>> Peter: Thanks for the tip! Really interesting. But I do not
>>>>> understand, why also this list contains the calls that were ended by
>>>>> sipp... Should I search for some mistake in my kamaillio config ?
>>>>
>>>> Perhaps you don't close them with unforce_rtp_proxy:
>>>>
>>>> if(method=="BYE" || method=="CANCEL"){
>>>> unforce_rtp_proxy();
>>>> }
>>>>
>>>> --
>>>> With best regards, Peter Lemenkov.
>>>>
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Re: [SR-Users] RTPProxy with kamailio : How to get calls count?

[Mino Haluz]

The results:

- rtpproxy calls count 280
- sipp calls count 2000
- iptraf on proxy 4.8MB/s
- G711a codec

So if my calculations are right (16kB/s per stream * 280 = 4.5MB/s),
rtpproxy calls count is really the right value. CPU usage is ok on
every machine (rtpproxy 20-30% CPU). Does anybody know why rtpproxy
cannot serve more than 270-280 calls ?

On Thu, Sep 13, 2012 at 5:07 PM, Mino Haluz  wrote:
> Ok, so I put there unforce_rtp_proxy even though I'm using
> rtpproxy_manage. The tip with nc now really shows the calls count.
>
> But the dialog count is still higher and higher, so I have bug
> somewhere in the configuration. I'll check it.
>
> On Thu, Sep 13, 2012 at 4:53 PM, Alex Balashov
>  wrote:
>> Correct, but you still need to call rtpproxy_manage() on receipt of a BYE or
>> CANCEL. It'll just figure out what to do on its own.
>>
>> None of this has to do with dialog state, though. Just rtpproxy control.
>>
>>
>>
>>
>> -- Alex
>>
>> --
>> Sent from my Samsung mobile, and thus lacking in the refinement one might
>> expect from a proper keyboard.
>>
>> Alex Balashov - Principal
>> Evariste Systems LLC
>> 235 E Ponce de Leon Ave
>> Suite 106
>> Decatur, GA 30030
>> Tel: +1-678-954-0670
>> Web: http://www.evaristesys.com/
>>
>> Mino Haluz  wrote:
>> I'm using rtpproxy_manage, so I assume unforce_rtp is not needed.
>>
>> On Thu, Sep 13, 2012 at 4:10 PM, Peter Lemenkov  wrote:
>>> 2012/9/13 Mino Haluz :
>>>
>>>> Peter: Thanks for the tip! Really interesting. But I do not
>>>> understand, why also this list contains the calls that were ended by
>>>> sipp... Should I search for some mistake in my kamaillio config ?
>>>
>>> Perhaps you don't close them with unforce_rtp_proxy:
>>>
>>> if(method=="BYE" || method=="CANCEL"){
>>> unforce_rtp_proxy();
>>> }
>>>
>>> --
>>> With best regards, Peter Lemenkov.
>>>
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Re: [SR-Users] RTPProxy with kamailio : How to get calls count?

[Mino Haluz]

Ok, so I put there unforce_rtp_proxy even though I'm using
rtpproxy_manage. The tip with nc now really shows the calls count.

But the dialog count is still higher and higher, so I have bug
somewhere in the configuration. I'll check it.

On Thu, Sep 13, 2012 at 4:53 PM, Alex Balashov
 wrote:
> Correct, but you still need to call rtpproxy_manage() on receipt of a BYE or
> CANCEL. It'll just figure out what to do on its own.
>
> None of this has to do with dialog state, though. Just rtpproxy control.
>
>
>
>
> -- Alex
>
> --
> Sent from my Samsung mobile, and thus lacking in the refinement one might
> expect from a proper keyboard.
>
> Alex Balashov - Principal
> Evariste Systems LLC
> 235 E Ponce de Leon Ave
> Suite 106
> Decatur, GA 30030
> Tel: +1-678-954-0670
> Web: http://www.evaristesys.com/
>
> Mino Haluz  wrote:
> I'm using rtpproxy_manage, so I assume unforce_rtp is not needed.
>
> On Thu, Sep 13, 2012 at 4:10 PM, Peter Lemenkov  wrote:
>> 2012/9/13 Mino Haluz :
>>
>>> Peter: Thanks for the tip! Really interesting. But I do not
>>> understand, why also this list contains the calls that were ended by
>>> sipp... Should I search for some mistake in my kamaillio config ?
>>
>> Perhaps you don't close them with unforce_rtp_proxy:
>>
>> if(method=="BYE" || method=="CANCEL"){
>> unforce_rtp_proxy();
>> }
>>
>> --
>> With best regards, Peter Lemenkov.
>>
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Re: [SR-Users] RTPProxy with kamailio : How to get calls count?

[Mino Haluz]

I'm using rtpproxy_manage, so I assume unforce_rtp is not needed.

On Thu, Sep 13, 2012 at 4:10 PM, Peter Lemenkov  wrote:
> 2012/9/13 Mino Haluz :
>
>> Peter: Thanks for the tip! Really interesting. But I do not
>> understand, why also this list contains the calls that were ended by
>> sipp... Should I search for some mistake in my kamaillio config ?
>
> Perhaps you don't close them with unforce_rtp_proxy:
>
> if(method=="BYE" || method=="CANCEL"){
> unforce_rtp_proxy();
> }
>
> --
> With best regards, Peter Lemenkov.
>
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Re: [SR-Users] RTPProxy with kamailio : How to get calls count?

[Mino Haluz]

Ok, I'm tagging dialogs with dlg_manage(), but even if the call ends,
it still keeps info about this dialog in list "kamctl fifo dlg_list".
Should I somehow close the dialog when the BYE transaction is ended ?

Peter: Thanks for the tip! Really interesting. But I do not
understand, why also this list contains the calls that were ended by
sipp... Should I search for some mistake in my kamaillio config ?

On Thu, Sep 13, 2012 at 3:57 PM, Alex Balashov
 wrote:
> Really? Interesting, I had no idea. I thought the rtpproxy control protocol
> was binary and did not lend itself easily to interaction in this manner.
> Thanks for the tip.
>
>
>
>
> -- Alex
>
> --
> Sent from my Samsung mobile, and thus lacking in the refinement one might
> expect from a proper keyboard.
>
> Alex Balashov - Principal
> Evariste Systems LLC
> 235 E Ponce de Leon Ave
> Suite 106
> Decatur, GA 30030
> Tel: +1-678-954-0670
> Web: http://www.evaristesys.com/
>
> Peter Lemenkov  wrote:
> 2012/9/13 Alex Balashov :
>> You can't get it from rtpproxy. You'd really have to use something like
>> the
>> dialog or htable modules to keep call state and get that from Kamailio.
>
> On the contrary it's possible (using raw UDP reads/writes):
>
> work ~: echo "h1u203u03 I\n" | nc -w 1 -u 127.0.0.1 2
> sessions created: 0
> active sessions: 0
> active streams: 0
> work ~:
>
> Where
>
> * h1u203u03 is randomly chosen token,
> * 127.0.0.1 is the rtpproxy's control IP,
> * 2 is the rtpproxy's control port,
> * "-u" means that we're using UDP
> * -w 1 is the timeout in seconds to wait before closing nc.
>
> I can't imagine that someone will use nc in performance testing but I
> think it looks like a good start.
>
> --
> With best regards, Peter Lemenkov.
>
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[SR-Users] RTPProxy with kamailio : How to get calls count?

[Mino Haluz]

Hi,

I'm doing the performance test with kamailio + RTPProxy, but I would
like to get the real calls count that the rtpproxy is serving. I don't
want to use value that I get from sipp.

So is there any management tool for rtpproxy, or should I get it
somewhere in kamailio config ?

Thanks,
Mino

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[SR-Users] 302 redirect with 2 numbers registered

[Mino Haluz]

Hi,

one number is registered on 2 phones. Phone1 has Always redirect set
to another number. When incoming call is initiated, Phone2 is ringing
and Phone1 sends 302 to the proxy. However the proxy does not send 302
to the caller (for ex. GW), but it waits for timeout of the Phone2.
Then the proxy sends 302 to the caller.

Can I do in kamailio, that it will ring on the Phone1 and also on the
number where it is redirected? I know kamailio is a proxy and cannot
initiate a call, but is there any solution? Thanks.

Mino

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[SR-Users] kamailio as SBC

[Mino Haluz]

Hi,

I know that kamailio is SIP proxy, but is there any way how to implement
kamailio as SBC like OpenSIPS with B2BUA module ?  I tried OpenSIPS with
this module, but it does not work with mediaproxy module.

Mino
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Re: [SR-Users] Kamailio ignores some ACK

[Mino Haluz]

We are troubleshooting this issue almost for 2 days and we did not find the
solution yet. The thing is, that it does not enter even the config file (we
have some debug messages at the start, that do not show). Otherwise we do
not have any problem with ACK, but only this particular ACK is not
forwarded... Was there some bug related to parsing in 3.1.0 which would not
print any error to syslog?

On Wed, May 9, 2012 at 4:41 PM, Jason Penton wrote:

> Seems like a loose routing issue. Are you loose routing in your config
> file?
>
>
> On Wed, May 9, 2012 at 4:34 PM, Stoyan Mihaylov <
> stoyan.v.mihay...@gmail.com> wrote:
>
>> You can use something like wireshark on Kamailio server to see if ACK
>> packets go in right direction.
>> I had problem with ACK and BYE, and I saw that in some cases ACK and BYE
>> packets looped back in kamailio.
>> May be I used wrong client.
>>
>>
>> On Wed, May 9, 2012 at 5:15 PM, Efelin Novak wrote:
>>
>>> Hi folks,
>>>
>>> I have a strange problem when Kamailio ignores ACKs in a specific
>>> scenario. The call flow is as follows:
>>>
>>> A -> INVITE -> kamailio -> INVITE -> B
>>> [omitting 100 and 180]
>>> A <- 200 OK <- kamailio <- 200 OK <- B
>>> A -> ACK -> kamailio
>>>
>>> There are INVITE Xlogs, Reply ROUTE xlogs and media-proxy logs in the
>>> syslog. However there is no information about these ACKs. No XLOGs are
>>> printed even if there is one on the top of the main route.
>>>
>>> "tcpdump -A -s0 -i any -n port 5060" receives this message correctly:
>>>
>>> 14:47:01.246153 IP 111.111.11.11.5060 > 80.80.80.80.60442: SIP, length:
>>> 915
>>> SIP/2.0 200 OK
>>> Via: SIP/2.0/UDP
>>> 111.111.11.11:5060
>>> ;rport=60442;x-route-tag="tgrp:A";branch=z9hG4bK1634E6A88
>>> Record-Route:
>>> 
>>> Contact: 
>>> To: "test_account";tag=cb7dd641
>>> From: ;tag=599248D4-260
>>> Call-ID: 9AFCFC51.11.50
>>> CSeq: 101 INVITE
>>> Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO,
>>> SUBSCRIBE, UPDATE
>>> Content-Type: application/sdp
>>> Content-Length:263
>>>
>>> v=0
>>> o=- 492575093 492575093 IN IP4 111.111.11.60
>>> s=test_device
>>> i=(o=IN IP4 192.168.1.10)
>>> c=IN IP4 111.111.11.71
>>> t=0 0
>>> m=audio 16416 RTP/AVP 18 101
>>> a=rtpmap:18 G729/8000
>>> a=fmtp:18 annexb=no
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-15
>>> a=ptime:20
>>>
>>> 14:47:01.254511 IP 111.111.11.50.60442 > 111.111.11.11.5060: SIP,
>>> length: 521
>>> ACK sip:80.80.80.80:65002;transport=udp SIP/2.0
>>> Via: SIP/2.0/UDP
>>> 111.111.11.50:5060;x-route-tag="tgrp:A";branch=z9hG4bK1634E7DE8
>>> From: ;tag=599248D4-260
>>> To: "test_account";tag=cb7dd641
>>> Call-ID: 9AFCFC51.11.50
>>> Route:
>>> 
>>> Max-Forwards: 70
>>> CSeq: 101 ACK
>>> Content-Length: 0
>>>
>>> My Kamailio version is kamailio 3.1.0 (i386/linux) 1e204f.
>>> Does anybody knows where can be a problem?
>>> How can I check whether Kamailio receives something?
>>>
>>> ...
>>>
>>> Jan
>>>
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Re: [SR-Users] siptrace and duplicate_uri - not forwarding the packets

[Mino Haluz]

Hm, the same settings works on testing 3.2.1. But this is not the solution
- I have to get it working on 3.2.0...

On Tue, Mar 6, 2012 at 5:16 PM, Mino Haluz  wrote:

> Hi,
>
> I have siptrace working but when I set duplicate_uri, no packet is sent to
> the destination. Was there some bug fix or anything I have to set further?
> There is nothing in syslog, siptrace locally is stored fine.
>
> modparam("siptrace", "db_url", "mysql://localhost/kamailio") #
> Database URL
> # modparam("siptrace", "traced_user_avp", "$avp(s:traced_user)")
> modparam("siptrace", "trace_on",1)
> modparam("siptrace", "trace_flag",  22) # Flag is used to
> mark messages to trace
> modparam("siptrace", "trace_sl_acks",0) # Do not trace
> ACKs separately. They are traced during normal sip_trace()
> modparam("siptrace", "duplicate_uri", "sip:IP1:5060")
>
> kamailio 3.2.0
>
> Thanks
>
> Mino
>
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[SR-Users] siptrace and duplicate_uri - not forwarding the packets

[Mino Haluz]

Hi,

I have siptrace working but when I set duplicate_uri, no packet is sent to
the destination. Was there some bug fix or anything I have to set further?
There is nothing in syslog, siptrace locally is stored fine.

modparam("siptrace", "db_url", "mysql://localhost/kamailio") #
Database URL
# modparam("siptrace", "traced_user_avp", "$avp(s:traced_user)")
modparam("siptrace", "trace_on",1)
modparam("siptrace", "trace_flag",  22) # Flag is used to
mark messages to trace
modparam("siptrace", "trace_sl_acks",0) # Do not trace ACKs
separately. They are traced during normal sip_trace()
modparam("siptrace", "duplicate_uri", "sip:IP1:5060")

kamailio 3.2.0

Thanks

Mino
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Re: [SR-Users] Problems with TCP

[Mino Haluz]

2) I tried to set mhomed=1, but it has become even worse - No out socket.
So I tried to disable it again and force kamailio to listen on physical
address as well (so simultaneously on virtual and physical). The error
disappeared, but the BYE message is not forwarded, it is processed and
onsend event-route is triggered as usually , but the packet is not sent
anywhere. I used tcpdump -i any. I will investigate if it is some network
issue, however, is there any way how to debug it further ?

On Mon, Feb 13, 2012 at 4:28 PM, Daniel-Constantin Mierla  wrote:

>  Hello,
>
> for 1) Is the device behind nat? Do you have tcp connection lifetime
> parameter value long enough?
>
> for 2) is the physical IP a public space IP? Do you have mhomed parameter
> set? It is a warning saying that the interface returned by OS for
> connecting to the destination is not in the listen list.
>
> for 3) yes it is expected behaviour, you just need to do record routing as
> usual, nothing special.
>
> Cheers,
> Daniel
>
>
> On 2/13/12 11:43 AM, Mino Haluz wrote:
>
> Hi,
>
> our customers are using mostly UDP but some of them want to use TCP. The
> problem is, I get various TCP errors in kamailio log and I do not
> understand what they mean
>
> 1) ERROR:  [tcp_main.c:4130]: connect  failed
> This means kamailio is trying to connect with TCP to the customer ??
> 2) WARNING: tcp_do_connect  : could not find
> corresponding listening socket for IPaddress , using default... ( where
> IPaddress is kamailio physical IP address)
> Kamailio has one virtual IP address (heartbeat IP address) and one
> physical. It listens on the virtual IP address but I do not understand why
> it is trying to use the physical IP address ?
>
> I was investigating this thing because the BYE message is not being
> relayed if the TCP is used (everytime the 2. error message is printed), so
> that's why I have to understand what it really means...
>
> And the last question,
>
> 3) If kamailio is relaying the traffic to the PSTN gateway, and the
> customer is using TCP, kamailio uses TCP as well for connecting to the PSTN
> gw. Is this behaviour normal ? I would expect that
>
> customer -- TCP --> kamailio --- UDP --> GW,
>
> and not
>
> customer -- TCP --> kamailio --- TCP --> GW,
>
> Thanks,
> Mino
>
>
>
>
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Re: [SR-Users] How to reload dispatcher destinations

[Mino Haluz]

-> It reloads the groups and included destinations. The command is disabled
for call load based dispatching (algorithm 10) since removal of
destinations may leave the list of active calls with broken references.

But nevermind, I realised that the dispatchers list can be updated with
set_state

On Thu, Feb 16, 2012 at 10:42 AM, Sammy Govind  wrote:

> which version are you using, there is no such condition in this page or is
> it?
> http://kamailio.org/docs/modules/3.1.x/modules_k/dispatcher.html#id2821010
>
>
>
> On Thu, Feb 16, 2012 at 2:31 PM, Mino Haluz  wrote:
>
>> Hi,
>>
>> is there any way how to reload dispatcher destinations (located in db)
>> without restarting the kamailio when algorithm 10 (call load balancing) is
>> used ? I read in documentation, that this command is disabled when this
>> algorithm is used.
>> Thanks,
>>
>> Mino
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[SR-Users] How to reload dispatcher destinations

[Mino Haluz]

Hi,

is there any way how to reload dispatcher destinations (located in db)
without restarting the kamailio when algorithm 10 (call load balancing) is
used ? I read in documentation, that this command is disabled when this
algorithm is used.
Thanks,

Mino
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[SR-Users] Problems with TCP

[Mino Haluz]

Hi,

our customers are using mostly UDP but some of them want to use TCP. The
problem is, I get various TCP errors in kamailio log and I do not
understand what they mean

1) ERROR:  [tcp_main.c:4130]: connect  failed
This means kamailio is trying to connect with TCP to the customer ??
2) WARNING: tcp_do_connect  : could not find corresponding
listening socket for IPaddress , using default... ( where IPaddress is
kamailio physical IP address)
Kamailio has one virtual IP address (heartbeat IP address) and one
physical. It listens on the virtual IP address but I do not understand why
it is trying to use the physical IP address ?

I was investigating this thing because the BYE message is not being relayed
if the TCP is used (everytime the 2. error message is printed), so that's
why I have to understand what it really means...

And the last question,

3) If kamailio is relaying the traffic to the PSTN gateway, and the
customer is using TCP, kamailio uses TCP as well for connecting to the PSTN
gw. Is this behaviour normal ? I would expect that

customer -- TCP --> kamailio --- UDP --> GW,

and not

customer -- TCP --> kamailio --- TCP --> GW,

Thanks,
Mino
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[SR-Users] cr_dump_routes buffer too small

[Mino Haluz]

Is there any way how to print it this way ?

sercmd -s unixs:/tmp/kamailio_ctl mi_fifo cr_dump_routes
error: 500 - Internal server error processing 's': buffer too small
(overflow) (-2)
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Re: [SR-Users] Dispatcher does not trigger the event_route

[Mino Haluz]

No, I do not see anything related in messages nor in syslog. Otherwise
dispatcher is working fine, but that event-route is still not triggered and
I really need the information about failure...

The output of kamctl fifo ds_list is ok, one gw is marked as IP and another
one as AP. When I flush the firewall the both are AP, so the mechanism is
working ..

Is there any debugging procedure of event-routes?

On Tue, Dec 20, 2011 at 11:13 AM, Peter Dunkley <
peter.dunk...@crocodile-rcs.com> wrote:

> **
> Once your destination has failed (and after you expect to see the output
> from the event_route), what is the output of kamctl fifo ds_list?
>
> Do you see anything else in /var/log/messages at this point?
>
> Peter
>
>
> On Tue, 2011-12-20 at 11:05 +0100, Mino Haluz wrote:
>
> nightly build 3.2.1.
>
> compiled on 05:08:17 Dec 19 2011 with gcc 4.4.5
>
>  On Tue, Dec 20, 2011 at 10:49 AM, Peter Dunkley <
> peter.dunk...@crocodile-rcs.com> wrote:
>
>  Hello,
>
> What version of Kamailio are you using?
>
> The dispatcher "event_route"s were added in Kamailio 3.2.0 so this will
> only work in Kamailio 3.2.0 and later.
>
> Regards,
>
> Peter
>
>
>
>
> On Tue, 2011-12-20 at 10:06 +0100, Mino Haluz wrote:
>
>
>Hi,
>
> I have two gateways pinged by kamailio. The both are AP (active/probing),
> when I cut the one gateway off, it becomes IP(inactive/probing) but the
> event_route is not fired up. Am I missing something ?
>
> event_route[dispatcher:dst-down] {
> xlog("L_ERR", "Destination down: $rm $ru ($du)\n");
> }
>
> Mino
>
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Re: [SR-Users] How to get Reason code from reply in failure_route

[Mino Haluz]

Ok, T_rpl(hdr(reason)) did the thing, however thanks!

On Fri, Dec 23, 2011 at 2:01 PM, Mino Haluz  wrote:

> I meant the reason specified in the reply as : Reason: Q.850;cause=17 ,
> that is not the SIP response code in fact. T_reply_code relates to the SIP
> response code.
>
>
> On Fri, Dec 23, 2011 at 12:35 PM, Daniel-Constantin Mierla <
> mico...@gmail.com> wrote:
>
>> Hello,
>>
>>
>> On 12/23/11 11:44 AM, Mino Haluz wrote:
>>
>>> Hi,
>>>
>>> how can I get the reason code from reply in failure_route ? $hdr(reason)
>>> points to the INVITE request..
>>>
>> you can test it with a regular expression via t_check_status(...) from tm
>> module or get it in a config variable:
>>
>> http://www.kamailio.org/wiki/**cookbooks/3.2.x/**
>> pseudovariables#t_reply_code<http://www.kamailio.org/wiki/cookbooks/3.2.x/pseudovariables#t_reply_code>
>>
>> Cheers,
>> Daniel
>>
>> --
>> Daniel-Constantin Mierla -- http://www.asipto.com
>> http://linkedin.com/in/miconda -- http://twitter.com/miconda
>>
>>
>
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Re: [SR-Users] How to get Reason code from reply in failure_route

[Mino Haluz]

I meant the reason specified in the reply as : Reason: Q.850;cause=17 ,
that is not the SIP response code in fact. T_reply_code relates to the SIP
response code.

On Fri, Dec 23, 2011 at 12:35 PM, Daniel-Constantin Mierla <
mico...@gmail.com> wrote:

> Hello,
>
>
> On 12/23/11 11:44 AM, Mino Haluz wrote:
>
>> Hi,
>>
>> how can I get the reason code from reply in failure_route ? $hdr(reason)
>> points to the INVITE request..
>>
> you can test it with a regular expression via t_check_status(...) from tm
> module or get it in a config variable:
>
> http://www.kamailio.org/wiki/**cookbooks/3.2.x/**
> pseudovariables#t_reply_code<http://www.kamailio.org/wiki/cookbooks/3.2.x/pseudovariables#t_reply_code>
>
> Cheers,
> Daniel
>
> --
> Daniel-Constantin Mierla -- http://www.asipto.com
> http://linkedin.com/in/miconda -- http://twitter.com/miconda
>
>
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[SR-Users] How to get Reason code from reply in failure_route

[Mino Haluz]

Hi,

how can I get the reason code from reply in failure_route ? $hdr(reason)
points to the INVITE request..

Thanks,
Mino
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Re: [SR-Users] Dispatcher does not trigger the event_route

[Mino Haluz]

nightly build 3.2.1.

compiled on 05:08:17 Dec 19 2011 with gcc 4.4.5

On Tue, Dec 20, 2011 at 10:49 AM, Peter Dunkley <
peter.dunk...@crocodile-rcs.com> wrote:

> **
> Hello,
>
> What version of Kamailio are you using?
>
> The dispatcher "event_route"s were added in Kamailio 3.2.0 so this will
> only work in Kamailio 3.2.0 and later.
>
> Regards,
>
> Peter
>
>
> On Tue, 2011-12-20 at 10:06 +0100, Mino Haluz wrote:
>
> Hi,
>
> I have two gateways pinged by kamailio. The both are AP (active/probing),
> when I cut the one gateway off, it becomes IP(inactive/probing) but the
> event_route is not fired up. Am I missing something ?
>
> event_route[dispatcher:dst-down] {
> xlog("L_ERR", "Destination down: $rm $ru ($du)\n");
> }
>
> Mino
>
> ___
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing 
> listsr-us...@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
>   --
> Peter Dunkley
> Technical Director
> Crocodile RCS Ltd
>
>
> ___
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> sr-users@lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
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[SR-Users] Dispatcher does not trigger the event_route

[Mino Haluz]

Hi,

I have two gateways pinged by kamailio. The both are AP (active/probing),
when I cut the one gateway off, it becomes IP(inactive/probing) but the
event_route is not fired up. Am I missing something ?

event_route[dispatcher:dst-down] {
xlog("L_ERR", "Destination down: $rm $ru ($du)\n");
}

Mino
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Re: [SR-Users] Dispatcher does not send pings

[Mino Haluz]

Ok, my mistake, the modparams related to dispatcher module were in !endif
that was not processed. In future, I will do not edit the cfg file without
code highlighting ..arrg

On Mon, Dec 19, 2011 at 3:56 PM, Mino Haluz  wrote:

> Updated to 3.2.1 (debian package) and still the same behaviour.
>
> version: kamailio 3.2.1 (i386/linux)
> flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS,
> DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC,
> DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE,
> USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES
> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
> MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB
> poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
> id: unknown
> compiled on 05:08:17 Dec 19 2011 with gcc 4.4.5
>
>
>
> On Mon, Dec 19, 2011 at 2:29 PM, Daniel-Constantin Mierla <
> mico...@gmail.com> wrote:
>
>>  Hello,
>>
>>
>> On 12/19/11 1:35 PM, Mino Haluz wrote:
>>
>> This is in the log file:
>> ERROR: dispatcher [dispatcher.c:640]: failover functions used, but AVPs
>> paraamters required are NULL -- feature disabled
>>
>>
>> it is not related to ping options. You have to upgrade first to 3.2.1 to
>> get the ping functionality as written in the readme. 3.2.0 uses old states
>> where there were overlapping cases and you have to set probing mode to the
>> gateways to be pinged.
>>
>> Cheers,
>> Daniel
>>
>>
>>
>> and the output of kamctl fifo ds_list:
>> SET_NO:: 1
>> SET:: 1
>> URI:: sip:IP1:5060 flags=AX priority=0 attrs=
>> URI:: sip:IP2:5060 flags=AX priority=0 attrs=
>>
>> version: kamailio 3.2.0 (i386/linux)
>>
>> On Mon, Dec 19, 2011 at 1:29 PM, Daniel-Constantin Mierla <
>> mico...@gmail.com> wrote:
>>
>>>  Hello,
>>>
>>> is this at least version 3.2.1 (or the latest branch 3.2)?
>>>
>>> What do you get with: kamctl fifo ds_list?
>>>
>>> Cheers,
>>> Daniel
>>>
>>>
>>> On 12/19/11 12:49 PM, Mino Haluz wrote:
>>>
>>>  Hi,
>>>
>>> please, could someone specify the conditions when the dispatcher module
>>> sends the INFO pings to the gateways ?
>>>
>>> modparam("dispatcher", "flags", 2)
>>> modparam("dispatcher", "dstid_avp", "$avp(s:test)")
>>> modparam("dispatcher", "dst_avp", "$avp(AVP_DST)")
>>> modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)")
>>> modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)")
>>> modparam("dispatcher", "ds_ping_interval", 30)
>>> modparam("dispatcher", "ds_probing_mode", 1)
>>> modparam("dispatcher", "ds_ping_method", "INFO")
>>> modparam("dispatcher", "ds_ping_from", "sip:001122@test")
>>>
>>> And my dispatcher.list is:
>>>
>>> 1 sip:IP1:5060
>>> 1 sip:IP2:5060
>>>
>>> Am I missing something? I dumped all the communication and the
>>> kamailio(dispatcher) does not send anything..
>>>
>>> Thanks,
>>> Mino
>>>
>>>
>>>  ___
>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing 
>>> listsr-us...@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>
>>>
>>> --
>>> Daniel-Constantin Mierla -- 
>>> http://www.asipto.comhttp://linkedin.com/in/miconda -- 
>>> http://twitter.com/miconda
>>>
>>>
>>
>>
>> ___
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing 
>> listsr-us...@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>> --
>> Daniel-Constantin Mierla -- 
>> http://www.asipto.comhttp://linkedin.com/in/miconda -- 
>> http://twitter.com/miconda
>>
>>
>
___
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Re: [SR-Users] Dispatcher does not send pings

[Mino Haluz]

Updated to 3.2.1 (debian package) and still the same behaviour.

version: kamailio 3.2.1 (i386/linux)
flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS,
DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC,
DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE,
USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB
poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
id: unknown
compiled on 05:08:17 Dec 19 2011 with gcc 4.4.5


On Mon, Dec 19, 2011 at 2:29 PM, Daniel-Constantin Mierla  wrote:

>  Hello,
>
>
> On 12/19/11 1:35 PM, Mino Haluz wrote:
>
> This is in the log file:
> ERROR: dispatcher [dispatcher.c:640]: failover functions used, but AVPs
> paraamters required are NULL -- feature disabled
>
>
> it is not related to ping options. You have to upgrade first to 3.2.1 to
> get the ping functionality as written in the readme. 3.2.0 uses old states
> where there were overlapping cases and you have to set probing mode to the
> gateways to be pinged.
>
> Cheers,
> Daniel
>
>
>
> and the output of kamctl fifo ds_list:
> SET_NO:: 1
> SET:: 1
> URI:: sip:IP1:5060 flags=AX priority=0 attrs=
> URI:: sip:IP2:5060 flags=AX priority=0 attrs=
>
> version: kamailio 3.2.0 (i386/linux)
>
> On Mon, Dec 19, 2011 at 1:29 PM, Daniel-Constantin Mierla <
> mico...@gmail.com> wrote:
>
>>  Hello,
>>
>> is this at least version 3.2.1 (or the latest branch 3.2)?
>>
>> What do you get with: kamctl fifo ds_list?
>>
>> Cheers,
>> Daniel
>>
>>
>> On 12/19/11 12:49 PM, Mino Haluz wrote:
>>
>>  Hi,
>>
>> please, could someone specify the conditions when the dispatcher module
>> sends the INFO pings to the gateways ?
>>
>> modparam("dispatcher", "flags", 2)
>> modparam("dispatcher", "dstid_avp", "$avp(s:test)")
>> modparam("dispatcher", "dst_avp", "$avp(AVP_DST)")
>> modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)")
>> modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)")
>> modparam("dispatcher", "ds_ping_interval", 30)
>> modparam("dispatcher", "ds_probing_mode", 1)
>> modparam("dispatcher", "ds_ping_method", "INFO")
>> modparam("dispatcher", "ds_ping_from", "sip:001122@test")
>>
>> And my dispatcher.list is:
>>
>> 1 sip:IP1:5060
>> 1 sip:IP2:5060
>>
>> Am I missing something? I dumped all the communication and the
>> kamailio(dispatcher) does not send anything..
>>
>> Thanks,
>> Mino
>>
>>
>>  ___
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing 
>> listsr-us...@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>> --
>> Daniel-Constantin Mierla -- 
>> http://www.asipto.comhttp://linkedin.com/in/miconda -- 
>> http://twitter.com/miconda
>>
>>
>
>
> ___
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing 
> listsr-us...@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
> --
> Daniel-Constantin Mierla -- 
> http://www.asipto.comhttp://linkedin.com/in/miconda -- 
> http://twitter.com/miconda
>
>
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Re: [SR-Users] Dispatcher does not send pings

[Mino Haluz]

This is in the log file:
ERROR: dispatcher [dispatcher.c:640]: failover functions used, but AVPs
paraamters required are NULL -- feature disabled

and the output of kamctl fifo ds_list:
SET_NO:: 1
SET:: 1
URI:: sip:IP1:5060 flags=AX priority=0 attrs=
URI:: sip:IP2:5060 flags=AX priority=0 attrs=

version: kamailio 3.2.0 (i386/linux)

On Mon, Dec 19, 2011 at 1:29 PM, Daniel-Constantin Mierla  wrote:

>  Hello,
>
> is this at least version 3.2.1 (or the latest branch 3.2)?
>
> What do you get with: kamctl fifo ds_list?
>
> Cheers,
> Daniel
>
>
> On 12/19/11 12:49 PM, Mino Haluz wrote:
>
> Hi,
>
> please, could someone specify the conditions when the dispatcher module
> sends the INFO pings to the gateways ?
>
> modparam("dispatcher", "flags", 2)
> modparam("dispatcher", "dstid_avp", "$avp(s:test)")
> modparam("dispatcher", "dst_avp", "$avp(AVP_DST)")
> modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)")
> modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)")
> modparam("dispatcher", "ds_ping_interval", 30)
> modparam("dispatcher", "ds_probing_mode", 1)
> modparam("dispatcher", "ds_ping_method", "INFO")
> modparam("dispatcher", "ds_ping_from", "sip:001122@test")
>
> And my dispatcher.list is:
>
> 1 sip:IP1:5060
> 1 sip:IP2:5060
>
> Am I missing something? I dumped all the communication and the
> kamailio(dispatcher) does not send anything..
>
> Thanks,
> Mino
>
>
> ___
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing 
> listsr-us...@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
> --
> Daniel-Constantin Mierla -- 
> http://www.asipto.comhttp://linkedin.com/in/miconda -- 
> http://twitter.com/miconda
>
>
___
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[SR-Users] Dispatcher does not send pings

[Mino Haluz]

Hi,

please, could someone specify the conditions when the dispatcher module
sends the INFO pings to the gateways ?

modparam("dispatcher", "flags", 2)
modparam("dispatcher", "dstid_avp", "$avp(s:test)")
modparam("dispatcher", "dst_avp", "$avp(AVP_DST)")
modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)")
modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)")
modparam("dispatcher", "ds_ping_interval", 30)
modparam("dispatcher", "ds_probing_mode", 1)
modparam("dispatcher", "ds_ping_method", "INFO")
modparam("dispatcher", "ds_ping_from", "sip:001122@test")

And my dispatcher.list is:

1 sip:IP1:5060
1 sip:IP2:5060

Am I missing something? I dumped all the communication and the
kamailio(dispatcher) does not send anything..

Thanks,
Mino
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Re: [SR-Users] How to get dispatcher list with sercmd command

[Mino Haluz]

Only this worked:

sercmd -s unixs:/tmp/kamailio_ctl system.listMethods

core.arg
core.echo
core.flags
core.info
core.kill
core.printi
core.prints
core.ps
core.psx
core.pwd
core.sctp_info
core.sctp_options
core.shmmem
core.tcp_info
core.tcp_options
core.udp4_raw_info
core.uptime
core.version
ctl.connections
ctl.listen
ctl.who
dispatcher.list
dispatcher.reload
dispatcher.set_state
dns.add_a
dns.add_
dns.add_srv
dns.debug
dns.debug_all
dns.delete_a
dns.delete_
dns.delete_all
dns.delete_all_force
dns.delete_cname
dns.delete_ebl
dns.delete_naptr
dns.delete_ptr
dns.delete_srv
dns.delete_txt
dns.lookup
dns.mem_info
dns.view
dst_blacklist.add
dst_blacklist.debug
dst_blacklist.delete_all
dst_blacklist.mem_info
dst_blacklist.view
mi
mi_dg
mi_fifo
mi_xmlrpc
pkg.stats
sl.stats
system.listMethods
system.methodHelp
system.methodSignature
tm.cancel
tm.hash_stats
tm.reply
tm.stats
tm.t_uac_start
tm.t_uac_wait



On Thu, Dec 15, 2011 at 4:14 PM, Mino Haluz  wrote:

> Do I have to load some specific module ?
>
> ERROR: connect_unix_sock: connect(/tmp/sercmd_ctl): No such file or
> directory [2]
>
>
>
> On Thu, Dec 15, 2011 at 3:55 PM, Daniel-Constantin Mierla <
> mico...@gmail.com> wrote:
>
>>
>>
>> On 12/15/11 3:45 PM, Mino Haluz wrote:
>>
>> kamctl dispatcher dump
>>
>> did the thing.
>>
>> that is using fifo file and MI command.
>>
>> sercmd is doing the RPC command. Can you give the output of:
>>
>> sercmd system.listMethods
>>
>> Cheers,
>> Daniel
>>
>>
>>
>>  kamailio 3.2.0.
>>
>> On Thu, Dec 15, 2011 at 3:38 PM, Daniel-Constantin Mierla <
>> mico...@gmail.com> wrote:
>>
>>>  Hello,
>>>
>>>
>>> On 12/15/11 2:50 PM, Timo Klecker wrote:
>>>
>>>  Hi Mino
>>>
>>>
>>>
>>> Try
>>>
>>> Sercmd dispatcher list
>>>
>>>
>>>
>>> Without the dot. This works with kamctl, haven’t used sercmd.
>>>
>>>  in sercmd should be dispatcher.list with dot, but the command is
>>> available starting with v3.2.0. What is your kamailio version?
>>>
>>> Cheers,
>>> Daniel
>>>
>>>
>>>
>>> Greetings
>>>
>>> Timo
>>>
>>>
>>>
>>> *Von:* sr-users-boun...@lists.sip-router.org [
>>> mailto:sr-users-boun...@lists.sip-router.org]
>>> *Im Auftrag von *Mino Haluz
>>> *Gesendet:* Donnerstag, 15. Dezember 2011 14:31
>>> *An:* SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) -
>>> Users Mailing List
>>> *Betreff:* [SR-Users] How to get dispatcher list with sercmd command
>>>
>>>
>>>
>>> Hi,
>>>
>>> # sercmd dispatcher.list
>>> error: 500 - command dispatcher.list not found
>>>
>>> The module is loaded, dispatcher.list file exists. Am I doing something
>>> wrong ? Thanks.
>>>
>>> Mino.
>>>
>>>
>>>  ___
>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing 
>>> listsr-us...@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>
>>>
>>> --
>>> Daniel-Constantin Mierla -- 
>>> http://www.asipto.comhttp://linkedin.com/in/miconda -- 
>>> http://twitter.com/miconda
>>>
>>>
>>
>> --
>> Daniel-Constantin Mierla -- 
>> http://www.asipto.comhttp://linkedin.com/in/miconda -- 
>> http://twitter.com/miconda
>>
>>
>
___
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Re: [SR-Users] How to get dispatcher list with sercmd command

[Mino Haluz]

Do I have to load some specific module ?

ERROR: connect_unix_sock: connect(/tmp/sercmd_ctl): No such file or
directory [2]


On Thu, Dec 15, 2011 at 3:55 PM, Daniel-Constantin Mierla  wrote:

>
>
> On 12/15/11 3:45 PM, Mino Haluz wrote:
>
> kamctl dispatcher dump
>
> did the thing.
>
> that is using fifo file and MI command.
>
> sercmd is doing the RPC command. Can you give the output of:
>
> sercmd system.listMethods
>
> Cheers,
> Daniel
>
>
>
>  kamailio 3.2.0.
>
> On Thu, Dec 15, 2011 at 3:38 PM, Daniel-Constantin Mierla <
> mico...@gmail.com> wrote:
>
>>  Hello,
>>
>>
>> On 12/15/11 2:50 PM, Timo Klecker wrote:
>>
>>  Hi Mino
>>
>>
>>
>> Try
>>
>> Sercmd dispatcher list
>>
>>
>>
>> Without the dot. This works with kamctl, haven’t used sercmd.
>>
>>  in sercmd should be dispatcher.list with dot, but the command is
>> available starting with v3.2.0. What is your kamailio version?
>>
>> Cheers,
>> Daniel
>>
>>
>>
>> Greetings
>>
>> Timo
>>
>>
>>
>> *Von:* sr-users-boun...@lists.sip-router.org [
>> mailto:sr-users-boun...@lists.sip-router.org]
>> *Im Auftrag von *Mino Haluz
>> *Gesendet:* Donnerstag, 15. Dezember 2011 14:31
>> *An:* SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) -
>> Users Mailing List
>> *Betreff:* [SR-Users] How to get dispatcher list with sercmd command
>>
>>
>>
>> Hi,
>>
>> # sercmd dispatcher.list
>> error: 500 - command dispatcher.list not found
>>
>> The module is loaded, dispatcher.list file exists. Am I doing something
>> wrong ? Thanks.
>>
>> Mino.
>>
>>
>>  ___
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing 
>> listsr-us...@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>> --
>> Daniel-Constantin Mierla -- 
>> http://www.asipto.comhttp://linkedin.com/in/miconda -- 
>> http://twitter.com/miconda
>>
>>
>
> --
> Daniel-Constantin Mierla -- 
> http://www.asipto.comhttp://linkedin.com/in/miconda -- 
> http://twitter.com/miconda
>
>
___
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Re: [SR-Users] How to get dispatcher list with sercmd command

[Mino Haluz]

kamctl dispatcher dump

did the thing. kamailio 3.2.0.

On Thu, Dec 15, 2011 at 3:38 PM, Daniel-Constantin Mierla  wrote:

>  Hello,
>
>
> On 12/15/11 2:50 PM, Timo Klecker wrote:
>
>  Hi Mino
>
> ** **
>
> Try 
>
> Sercmd dispatcher list 
>
> ** **
>
> Without the dot. This works with kamctl, haven’t used sercmd.
>
> in sercmd should be dispatcher.list with dot, but the command is available
> starting with v3.2.0. What is your kamailio version?
>
> Cheers,
> Daniel
>
>  
>
> ** **
>
> Greetings
>
> Timo
>
> ** **
>
> *Von:* sr-users-boun...@lists.sip-router.org [
> mailto:sr-users-boun...@lists.sip-router.org]
> *Im Auftrag von *Mino Haluz
> *Gesendet:* Donnerstag, 15. Dezember 2011 14:31
> *An:* SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) -
> Users Mailing List
> *Betreff:* [SR-Users] How to get dispatcher list with sercmd command
>
> ** **
>
> Hi,
>
> # sercmd dispatcher.list
> error: 500 - command dispatcher.list not found
>
> The module is loaded, dispatcher.list file exists. Am I doing something
> wrong ? Thanks.
>
> Mino.
>
>
> ___
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing 
> listsr-us...@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
> --
> Daniel-Constantin Mierla -- 
> http://www.asipto.comhttp://linkedin.com/in/miconda -- 
> http://twitter.com/miconda
>
>
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[SR-Users] How to get dispatcher list with sercmd command

[Mino Haluz]

Hi,

# sercmd dispatcher.list
error: 500 - command dispatcher.list not found

The module is loaded, dispatcher.list file exists. Am I doing something
wrong ? Thanks.

Mino.
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[SR-Users] Dispatcher module with call load balancing

[Mino Haluz]

Hi,

I would like to use the dispatcher module with algorithm 10 - call load
balancing. But there are multiple things that are a bit unclear to me. I
have multiple gateways that can serve different maximum number of calls.

1) Can I somehow set the maximum calls count for each dispatcher gateway,
so that it can fairly distribute the calls? I know it is stateless module,
but how could I solve this?
2) Is there any method how can I get the current number of calls for each
gateway from the dispatcher module?
3) What is the expiration timer in the dispatcher module for dialogs if the
dialog is canceled uncleanly ?
4) Imagine I will restart kamailio, it will lose the call count and the
distribution will be incorrect. Maybe it could nice, that kamailio would
offer MI command to set these values.

Thanks for the responses,
Mino
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Re: [SR-Users] Limiting simultaneous calls

[Mino Haluz]

1) Is this method per user?
2) Another thing is, what happens if the BYE is not received (network issue
or whatever), will the dialog expire on the kamailio side so the dialog
count could me decremented? I mean, if the user has just 1 voice channel,
he could not make a call anymore, that's why it should be reliable..

On Mon, Nov 21, 2011 at 11:43 AM, Alex Balashov
wrote:

> On 11/21/2011 05:42 AM, Mino Haluz wrote:
>
>  I was using cdrtool (prepaid table) and callcontrol to limit
>> concurrent calls. In fact this is only limiting the outbound calls,
>> but I would like to use another mechanism which should limit the
>> inbound calls too. So basically to limit voice channels.
>>
>> So is there some reliable method/module how to achieve this?
>>
>
> The 'dialog' module is a common approach to this problem.  Organise both
> inbound and outbound dialogs into profiles, and then get_profile_size() in
> both your inbound and outbound call processing.
>
> --
> Alex Balashov - Principal
> Evariste Systems LLC
> 260 Peachtree Street NW
> Suite 2200
> Atlanta, GA 30303
> Tel: +1-678-954-0670
> Fax: +1-404-961-1892
> Web: http://www.evaristesys.com/
>
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[SR-Users] Limiting simultaneous calls

[Mino Haluz]

Hi,

I was using cdrtool (prepaid table) and callcontrol to limit concurrent
calls. In fact this is only limiting the outbound calls, but I would like
to use another mechanism which should limit the inbound calls too. So
basically to limit voice channels.

So is there some reliable method/module how to achieve this?

Thanks,
Mino
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[SR-Users] CANCEL is not forwarded, if callee does not respond

[Mino Haluz]

Hi,

the scenario is like this:

- user1 makes a call to user2
- proxy sends receives INVITE and sends it to user2
- user2 does not respond at all (firewall for example)
- user1 after 5 seconds hangs up the call with CANCEL
- proxy sends to user1 200 canceling

CANCEL is not forwarded to user2. Ok this behaviour is maybe ok, but after
30 seconds Request timed out is sent to user1 (fr_inv_timer possibly hit).
So if I am thinking well, first CANCEL did not match the transaction ? Why
? It this behaviour ok or do I have to search for something wrong in the
config?

Thanks,
Mino
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[SR-Users] kamailio and high-available db backend

[Mino Haluz]

Hi,

I was using ndb cluster as the dabatase backend for my kamailio setup. The
thing is, mysql ndb cluster is difficult to maintain for me and it crashed
multiple times (desynchronization, datamemory limit reached, and other
strange reasons) and it is not that well documented (the user-base is quite
narrow). My question is, what mysql architecture would you recommend me, if
it has to be high available, well documented with various examples wihitn
the community, thus easy to maintain. DB has 5% writes, 100queries/sec. and
its size is 100MB.

Mino
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Re: [SR-Users] Carrierroute: Could not allocate shared memory

[Mino Haluz]

I increased the shared memory size to 1GB with

sysctl -w kernel.shmmax=10

The problem still persists. The error happens when "/etc/init.d/kamailio
restart" is executed. Is there any other way how to increase it ?

On Mon, Oct 10, 2011 at 3:49 PM, marius zbihlei wrote:

> On 10/10/2011 03:51 PM, Mino Haluz wrote:
>
>> Hi,
>>
>> is there any setting which could allow me to set maximum memory per module
>> ? As I am testing the carrier route module, I've added for testing purposes
>> 100 000 rules sofar. When I start the kamailio, it gives me :
>>
>> Oct 10 14:46:46 kamrouter /usr/sbin/kamailio[2182]: ERROR: carrierroute
>> [../../ut.h:702]: could not allocate shared memory from available pool
>> Oct 10 14:46:46 kamrouter /usr/sbin/kamailio[2182]: ERROR: carrierroute
>> [cr_rule.c:155]: could not allocate shared memory from available pool
>>
>> Thanks.
>>
>>  Hello
>
> Can you increase the shared memory size ? But 100k rules should take only
> about 40-50MB of shared memory... I advise for at least 1GB(gigabyte) of
> SHM.
>
> The error is happening only when the rules are loaded (or reloaded via
> cr_reload_routes) or during a message processing (I understand the former
> but just want to make sure)
>
> Cheers,
> Marius
>
> --
> Zbihlei Marius
>
> Head of
> Linux Development Services Romania
>
> 1&1 Internet Development srlTel KA: 754-9512
> Str Mircea Eliade 18Tel RO: +40-31-223-9512
> Sect 1, Bucuresti   mailto: marius.zbih...@1and1.ro
> 71295, Romania
>
>
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[SR-Users] Carrierroute: Could not allocate shared memory

[Mino Haluz]

Hi,

is there any setting which could allow me to set maximum memory per module ?
As I am testing the carrier route module, I've added for testing purposes
100 000 rules sofar. When I start the kamailio, it gives me :

Oct 10 14:46:46 kamrouter /usr/sbin/kamailio[2182]: ERROR: carrierroute
[../../ut.h:702]: could not allocate shared memory from available pool
Oct 10 14:46:46 kamrouter /usr/sbin/kamailio[2182]: ERROR: carrierroute
[cr_rule.c:155]: could not allocate shared memory from available pool

Thanks.
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Re: [SR-Users] Multiple INVITEs and discarded call_control

[Mino Haluz]

Hi,

I finally resolved this issue. The problem is in the callcontrol module.

The original code:

static int
postprocess_request(struct sip_msg *msg, unsigned int flags, void *_param)
{
CallInfo *call;

if ((msg->msg_flags & FL_USE_CALL_CONTROL) == 0)
return 1;

// the FL_USE_CALL_CONTROL flag is still set => the dialog was not
created

LOG(L_WARN, "dialog to trace controlled call was not created. discarding
callcontrol.");

call = get_call_info(msg, CAStop);
if (!call) {
LOG(L_ERR, "can't retrieve call info\n");
return -1;
}
call_control_stop(msg, call->callid);

return 1;
}

When I was debugging the whole thing, I found out, that this postprocess
routine of the callcontrol module is called EVERYTIME the processing of the
main route block of kamailio is being ended.
Why? I do not understand, this must be considered as a bug.

When those 3 INVITEs come, the first is processed, but it still has
FL_USE_CALL_CONTROL flag. Another 2 INVITEs are absorbed by kamailio. As I
said, the postprocess_request block is executed each time the main route
block ends. So, it will enter this function, check the FL_USE_CALL_CONTROL
flag. And here comes a strange thing, why the flag FL_USE_CALL_CONTROL is
shared accross the same INVITEs? I thought every instance of the module
which processes every message should have independent variables. But it
seems that this flag is shared, that's why it executes call_control_stop and
sends to the callcontrol socket the command of discarding the callcontrol
feature (the call is not charged)

I patched the module by adding extra parameter  - actually name of the AVP
variable. This variable is set in the main route block when the first INVITE
is processed and the call is initiated. Then it is checked in the routing -
if the call has initiated bit set, do not discard the callcontrol. If
someone is experiencing the similar problem, I can send the patch.

Second simpler approach is to removing  this line

call_control_stop(msg, call->callid);

The question is, will the dialog expire in the callcontrol daemon? Is there
any timeout? If so, it can be clearly fixed by this way ;)

Hope this helps to somebody,
Cheers






On Mon, Aug 15, 2011 at 9:59 AM, Mino Haluz  wrote:

> Any updates on this? I updated callcontrol which has some bug fixed:
>
> callcontrol (2.0.14) unstable; urgency=low
>
>  * Avoid handling requests with a duplicated CallID
>
> But it still does not work. The callcontrol is executed 3 times and in
> the config, I have only once the mark "xxx" printed in syslog (see the
> config below).
> t_newtran before calling callcontrol does not work as well - it is
> executed 3 times.
>
> On Mon, Jun 13, 2011 at 4:29 PM, Mino Haluz  wrote:
> > It does not work, t_newtran always returns success, so it will never
> > absorb the retransmission.
> > So what I did was:
> >
> >   if ($sht(a=>$ci::retrans) == 0) {
> >   $sht(a=>$ci::retrans) = 1;
> >   } else {
> >   exit();
> >   }
> >
> >  xlog("L_INFO","XLOG: xxx");
> >  call_control();
> >
> > I prepared sipp scenario which generates 3 INVITEs separated by 200ms.
> > It will gives me this:
> >
> > Jun 13 16:07:04 no-testing /sbin/kamailio[5274]: INFO: 

Re: [SR-Users] Account 403 to RADIUS

[Mino Haluz]

Ok, the problem is the original feeradius package (squeeze) does not support
Update radius messages (it gives Unsupported Acct-Status-Type = 15).
It's not related to kamailio.

.. however anybody knows how to patch it?

On Mon, Sep 26, 2011 at 1:19 PM, Mino Haluz  wrote:

> Sorry for misunderstanding, yes, the failed status is firing the
> insert_radacct_record but with different parameters. I must see why it is
> not written to db.
>
> Thanks.
>
>
> On Mon, Sep 26, 2011 at 1:05 PM, Mino Haluz  wrote:
>
>> Ok, but when I use acc_rad_request only Failed status type is sent to
>> Radius, which cannot be written to DB. There is no START , so the
>> insert_radacct_record stored procedure is not executed before. The Failed
>> status type fires up the update stored procedure on the existing record. But
>> when there is no record in db, it cannot be updated.
>>
>>
>> On Mon, Sep 26, 2011 at 12:15 PM, Daniel-Constantin Mierla <
>> mico...@gmail.com> wrote:
>>
>>> Hello,
>>>
>>>
>>> On 9/26/11 11:26 AM, Mino Haluz wrote:
>>>
>>>> Hi,
>>>>
>>>> I have this code:
>>>>
>>>>if ( is_user_in("From", "blocked") && is_method("INVITE")) {
>>>>xlog("L_INFO", "XLOG: [number_and_ruri_checks] NOTICE:
>>>> Account ($fu) to ($ru) is blocked");
>>>>sl_send_reply("403", " Account blocked ");
>>>>exit;
>>>>}
>>>>
>>>> I would like to account also this answer to Radius, now I have it only
>>>> in application xlog.
>>>>
>>> an easy way is to use acc_rad_request("403 Account blocked") right in the
>>> config file after you send the reply.
>>>
>>> Cheers,
>>> Daniel
>>>
>>> --
>>> Daniel-Constantin Mierla -- http://www.asipto.com
>>> Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat
>>> http://linkedin.com/in/miconda -- http://twitter.com/miconda
>>>
>>>
>>
>
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Re: [SR-Users] Account 403 to RADIUS

[Mino Haluz]

Sorry for misunderstanding, yes, the failed status is firing the
insert_radacct_record but with different parameters. I must see why it is
not written to db.

Thanks.

On Mon, Sep 26, 2011 at 1:05 PM, Mino Haluz  wrote:

> Ok, but when I use acc_rad_request only Failed status type is sent to
> Radius, which cannot be written to DB. There is no START , so the
> insert_radacct_record stored procedure is not executed before. The Failed
> status type fires up the update stored procedure on the existing record. But
> when there is no record in db, it cannot be updated.
>
>
> On Mon, Sep 26, 2011 at 12:15 PM, Daniel-Constantin Mierla <
> mico...@gmail.com> wrote:
>
>> Hello,
>>
>>
>> On 9/26/11 11:26 AM, Mino Haluz wrote:
>>
>>> Hi,
>>>
>>> I have this code:
>>>
>>>if ( is_user_in("From", "blocked") && is_method("INVITE")) {
>>>xlog("L_INFO", "XLOG: [number_and_ruri_checks] NOTICE:
>>> Account ($fu) to ($ru) is blocked");
>>>sl_send_reply("403", " Account blocked ");
>>>exit;
>>>}
>>>
>>> I would like to account also this answer to Radius, now I have it only in
>>> application xlog.
>>>
>> an easy way is to use acc_rad_request("403 Account blocked") right in the
>> config file after you send the reply.
>>
>> Cheers,
>> Daniel
>>
>> --
>> Daniel-Constantin Mierla -- http://www.asipto.com
>> Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat
>> http://linkedin.com/in/miconda -- http://twitter.com/miconda
>>
>>
>
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Re: [SR-Users] Account 403 to RADIUS

[Mino Haluz]

Ok, but when I use acc_rad_request only Failed status type is sent to
Radius, which cannot be written to DB. There is no START , so the
insert_radacct_record stored procedure is not executed before. The Failed
status type fires up the update stored procedure on the existing record. But
when there is no record in db, it cannot be updated.

On Mon, Sep 26, 2011 at 12:15 PM, Daniel-Constantin Mierla <
mico...@gmail.com> wrote:

> Hello,
>
>
> On 9/26/11 11:26 AM, Mino Haluz wrote:
>
>> Hi,
>>
>> I have this code:
>>
>>if ( is_user_in("From", "blocked") && is_method("INVITE")) {
>>xlog("L_INFO", "XLOG: [number_and_ruri_checks] NOTICE:
>> Account ($fu) to ($ru) is blocked");
>>sl_send_reply("403", " Account blocked ");
>>exit;
>>}
>>
>> I would like to account also this answer to Radius, now I have it only in
>> application xlog.
>>
> an easy way is to use acc_rad_request("403 Account blocked") right in the
> config file after you send the reply.
>
> Cheers,
> Daniel
>
> --
> Daniel-Constantin Mierla -- http://www.asipto.com
> Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat
> http://linkedin.com/in/miconda -- http://twitter.com/miconda
>
>
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[SR-Users] Account 403 to RADIUS

[Mino Haluz]

Hi,

I have this code:

if ( is_user_in("From", "blocked") && is_method("INVITE")) {
xlog("L_INFO", "XLOG: [number_and_ruri_checks] NOTICE:
Account ($fu) to ($ru) is blocked");
sl_send_reply("403", " Account blocked ");
exit;
}

I would like to account also this answer to Radius, now I have it only in
application xlog.

Thanks,
Mino
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[SR-Users] carrierroute module

[Mino Haluz]

Hi,

I have a problem with configuring the carrierroute module, I do not clearly
understand how it should work from the documentation. What I would like to
do is that I have 2 gateways (A.A.A.A and B.B.B.B), lots of scan_prefixes
with different rewrite prefixes. The traffic should be sent to the 1
gateway, and if it fails (5xx,4xx codes) it should go to the another. I have
only one SIP domain (domain.com). The question is, how the tables
(domain_name, carrierroute, carrierfailroute) should be filled? Can I
achieve this scenario with the carrierroute module ?

Thanks.
Mino
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[SR-Users] Carrier module for kamailio

[Mino Haluz]

Hi,

I need a module which could allow me to send traffic to various carriers and
it has to support some important features. So some basic ones:

- possibility to re-route the call in case the original route fails
- peak/offpeak conditions (time-based)
- route traffic according to prefix

I found LCR and carrierroute module, but it does not have peak/offpeak
feature. Correct me if I am wrong..
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[SR-Users] Resend INVITE in case of failure

[Mino Haluz]

Hi,

I would like to resend the INVITE to another gateway if the original
fails (503, 500, etc.). Is there any module or mechanism which could
do this ?

Mino

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[SR-Users] Wrong formatting of Request URI

[Mino Haluz]

Hi,

my kamailio is sometimes receiving ruri from remote clients which I
assume are not right formatted.

For ex.: sip:user@kamailio=3Buser=3Dphone

Is there any mechanism which could transform this into something
readable for kamailio or do I have to simply cut off everything after
the host name of the RURI ?

Mino

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Re: [SR-Users] Multiple INVITEs and discarded call_control

[Mino Haluz]

Any updates on this? I updated callcontrol which has some bug fixed:

callcontrol (2.0.14) unstable; urgency=low

  * Avoid handling requests with a duplicated CallID

But it still does not work. The callcontrol is executed 3 times and in
the config, I have only once the mark "xxx" printed in syslog (see the
config below).
t_newtran before calling callcontrol does not work as well - it is
executed 3 times.

On Mon, Jun 13, 2011 at 4:29 PM, Mino Haluz  wrote:
> It does not work, t_newtran always returns success, so it will never
> absorb the retransmission.
> So what I did was:
>
>           if ($sht(a=>$ci::retrans) == 0) {
>               $sht(a=>$ci::retrans) = 1;
>           } else {
>               exit();
>           }
>
>          xlog("L_INFO","XLOG: xxx");
>          call_control();
>
> I prepared sipp scenario which generates 3 INVITEs separated by 200ms.
> It will gives me this:
>
> Jun 13 16:07:04 no-testing /sbin/kamailio[5274]: INFO: 

[SR-Users] kamailio conference calls support

[Mino Haluz]

Hi,

does kamailio support conference calls or it has to be implemented in
some B2BUA ?
If it is supported, what are the requirements (modules etc.).

Thanks,
Mino

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