Re: [OpenSIPS-Users] Opensips caching

[Khalil Khamlichi]

Are you using dr_rules table ?

On Thu, Jan 17, 2019, 8:18 AM Vishal Pai  Hello everyone
>
> what is the best way to implement the caching in opensips. I am doing
> following
>
> we have a rate table with NPANXX with rate per minute. Every time when we
> do outbound calls we get NPANXX of dialed number and search it in database.
> Since in this way it will create a load on cpu for each select query.
>
> Can anyone help me in this also let me know if there is any caching
> functionality is available in opensips ?
>
>
> Vishal Pai
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Re: [OpenSIPS-Users] ACC module with JSON events

[Khalil Khamlichi]

Hi Dan,

Can you show us how to do that with some step by step tutorial, we had
a lot of trouble trying to figure out how to use CGRates, with this
functionality of json export, we will at least use some of CGRates
functionalities and hopefully slowly get familiar with it.

Thanks in advance.

On Mon, Feb 26, 2018 at 7:51 AM, DanB  wrote:
> Sammy,
>
> Another option on short term until the new feature will be implemented
> in OpenSIPS would be to use CGRateS as CDR format converter: receive
> CDRs from "cgrates" module in OpenSIPS and use online export of CGRateS
> to further export the CDR in the JSON over http (customizable fields).
> You don't need to configure much on CGRateS side in this case since no
> billing needs to be involved.
>
> DanB
>
>
>
>
>
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[OpenSIPS-Users] trigger on acc table

[Khalil Khamlichi]

Hi,

We are implementing a trigger on acc table, for each insert.
My question is : does opensips group multiple inserts into one single
transcation or it does an insert of one single row per call at a time.

this is very important for triggers as they can see only the last
inserted line in a transaction if opensips groups inserts into one
transaction.

Thanks in advance.

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Re: [OpenSIPS-Users] acc extra_fields

[Khalil Khamlichi]

modparam("acc", "db_extra", "from_uri; to_uri; src_uri")

then inside script when those variables are available you do :

for example :
$acc_extra(from_uri) = $rU;
$acc_extra(src_ip) = $si;




On Tue, Nov 28, 2017 at 7:41 PM, vudn5  wrote:
> i have tini trouble with opensips 2.3.2  .  In old version 2.2.2 i have a
> config
>
>  modparam("acc", "db_extra", "from_uri=$fU; to_uri=$tU; src_uri=$si")
>
> to save $fU ,$tU , $si  variable to columns   from_uri , to_uri , src_uri in
> table acc ( i create 3 fields)
>
>
>
> but i don't now how can i do that in opensips 2.3.2?i read a EXTRA_FIELD
> and it's my config:
>
> modparam("acc", "extra_fields", "db: $fU->from_uri; $tU-> to_uri;
> $si->src_uri ")
>
>  but it's not work, and i want to khow why it's not work?
> Thanks
>
>
>
> --
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Re: [OpenSIPS-Users] outbound call center services

[Khalil Khamlichi]

You can still use opensips callcenter module for your agents though. As
only dialer function will be handled by freeswitch.

Sent via mobile, please forgive typos and brevity

On Nov 8, 2017 12:36 PM, "Khalil Khamlichi" <khamlichi.kha...@gmail.com>
wrote:

> You will surely need answering machine detection at a certain point of
> your developement for this kind of job (dialer) which is not possible
> because opensips does not handle media. So your best bet is freeswitch or
> asterisk. Freeswitch is optimal but lacks a free answering machine
> detection module. Asterisk less programatic but has it for free.
>
> Sent via mobile, please forgive typos and brevity
>
> On Nov 8, 2017 12:31 PM, "Dragomir Haralambiev" <goup2...@gmail.com>
> wrote:
>
> Hello all,
>
> The Call Center module is develop for inbound call center service.
>
> Does anyone have any idea how to develop (auto dialer) for outbound call
> center services with call distribution to agents.
>
> Best regards,
> Dragomir
>
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Re: [OpenSIPS-Users] outbound call center services

[Khalil Khamlichi]

You will surely need answering machine detection at a certain point of your
developement for this kind of job (dialer) which is not possible because
opensips does not handle media. So your best bet is freeswitch or asterisk.
Freeswitch is optimal but lacks a free answering machine detection module.
Asterisk less programatic but has it for free.

Sent via mobile, please forgive typos and brevity

On Nov 8, 2017 12:31 PM, "Dragomir Haralambiev"  wrote:

Hello all,

The Call Center module is develop for inbound call center service.

Does anyone have any idea how to develop (auto dialer) for outbound call
center services with call distribution to agents.

Best regards,
Dragomir

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Re: [OpenSIPS-Users] opensips on Ubuntu >= 16.04

[Khalil Khamlichi]

http://apt.opensips.org/packages.php?v=2.3

Sent via mobile, please forgive typos and brevity

On Nov 6, 2017 8:53 AM, "Martin Horne"  wrote:

> Hi I have opensips built for arm64 (opensip-cp is working); has anybody
> got a cook book for installing opensips on Ubuntu 16.04?
>
> Would also be very interested if anybody has it building on Alpine.
>
> Thanks
>
>
>
>
>
> *Martin Horne*
>
> Developer Advocate  |
>
> System Architect| *Arm*
>
> *. . . . . . . . . . . . . . . . . . . . . . . . . . . *
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Re: [OpenSIPS-Users] OpenSIPS Businesses Requirements

[Khalil Khamlichi]

Hi,

I believe Opensips team does work worldwide.
anyway there is a section on the website that points to businesses
http://www.opensips.org/Community/Business

Regards,

On Tue, Oct 17, 2017 at 8:33 AM, Voice TAC  wrote:
> Gents,
>
> I’m looking for some companies that can work on OPENSIPS in Middle East.
>
> Where I can find a list of them.
>
>
> Sent from my iPhone
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Re: [OpenSIPS-Users] [NEW] SIP Recording module

[Khalil Khamlichi]

thanks a lot. always looking forward to reading your work.

On Fri, Sep 29, 2017 at 11:35 AM, Răzvan Crainea <raz...@opensips.org>
wrote:

> Hi, Khalil!
>
> I will write a more detailed tutorial in the upcoming weeks and get back
> to you.
>
> Best regards,
>
> Răzvan Crainea
> OpenSIPS Developerwww.opensips-solutions.com
>
> On 09/29/2017 12:24 PM, Khalil Khamlichi wrote:
>
> we need a tutorial for this.
>
> On Fri, Sep 29, 2017 at 11:16 AM, Răzvan Crainea <raz...@opensips.org>
> wrote:
>
>> Hi, Dragomir!
>>
>> Yes, using this module you can record sound, but you will also need a
>> recorder such as Oreka [4] to do the actual recording.
>>
>> Best regards,
>>
>> Răzvan Crainea
>> OpenSIPS Developerwww.opensips-solutions.com
>>
>> On 09/29/2017 11:56 AM, Dragomir Haralambiev wrote:
>>
>> Hello,
>>
>> I did not understand this module can record sound or only SIP signaling.
>>
>> Best regards,
>> Dragomir
>>
>> 2017-09-28 20:45 GMT+03:00 Răzvan Crainea <raz...@opensips.org>:
>>
>>> Hello, everyone!
>>>
>>> I am happy to announce you we have just released a new module: the
>>> SIPREC[1] module can be used to do call recording on a passive server using
>>> the SIPREC standardized protocol[2].
>>>
>>> The work for this module has been sponsored by OrecX[3] and the module
>>> is fully compliant with the Oreka [4] open-source call recoding tool.
>>>
>>> [1] http://www.opensips.org/html/docs/modules/2.4.x/siprec
>>> [2] https://tools.ietf.org/html/rfc7866
>>> [3] http://www.orecx.com/
>>> [4] http://oreka.sourceforge.net/
>>>
>>> Best regards,
>>>
>>> --
>>> Răzvan Crainea
>>> OpenSIPS Developer
>>> www.opensips-solutions.com
>>>
>>>
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>>
>>
>>
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>
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Re: [OpenSIPS-Users] [NEW] SIP Recording module

[Khalil Khamlichi]

we need a tutorial for this.

On Fri, Sep 29, 2017 at 11:16 AM, Răzvan Crainea 
wrote:

> Hi, Dragomir!
>
> Yes, using this module you can record sound, but you will also need a
> recorder such as Oreka [4] to do the actual recording.
>
> Best regards,
>
> Răzvan Crainea
> OpenSIPS Developerwww.opensips-solutions.com
>
> On 09/29/2017 11:56 AM, Dragomir Haralambiev wrote:
>
> Hello,
>
> I did not understand this module can record sound or only SIP signaling.
>
> Best regards,
> Dragomir
>
> 2017-09-28 20:45 GMT+03:00 Răzvan Crainea :
>
>> Hello, everyone!
>>
>> I am happy to announce you we have just released a new module: the
>> SIPREC[1] module can be used to do call recording on a passive server using
>> the SIPREC standardized protocol[2].
>>
>> The work for this module has been sponsored by OrecX[3] and the module is
>> fully compliant with the Oreka [4] open-source call recoding tool.
>>
>> [1] http://www.opensips.org/html/docs/modules/2.4.x/siprec
>> [2] https://tools.ietf.org/html/rfc7866
>> [3] http://www.orecx.com/
>> [4] http://oreka.sourceforge.net/
>>
>> Best regards,
>>
>> --
>> Răzvan Crainea
>> OpenSIPS Developer
>> www.opensips-solutions.com
>>
>>
>> ___
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>>
>
>
>
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Re: [OpenSIPS-Users] No data in CDR viewer

[Khalil Khamlichi]

This document contains instructions tested  up to version 2.3.2 so just
read carefully.


Sent from my Samsung Note

On Sep 26, 2017 11:51 AM, "Ketan Kothari" <kkothari...@gmail.com> wrote:

> Hello Khalil,
>
> Thanks for your response.
>
> I have checked in database there no records of calls.
> I'm using opensips 2.3.1 is there any document or sample of opensips.cfg
> for accounting.
>
>  http://www.opensips.org/Documentation/Tutorials-Advanced-Accounting i
> have checked this link is last modified at November 03, 2016 before
> opensips 2.3.1 release.
>
> So please suggest me if any doc available for newly version.
>
>
>
> On Tue, Sep 26, 2017 at 3:11 PM, Khalil Khamlichi <
> khamlichi.kha...@gmail.com> wrote:
>
>> Hi,
>>
>> honestly, I never used the opensips GUI, but I think you need first check
>> the database tables ; acc and missed_calls and and see if there are any
>> records in there. then you could activate logging on mysql and see what
>> queries are being run when you click that button, then third step you need
>> to check apache/php logs there must be an error or an indication in there.
>> if no luck you surely just need to re-read this http://www.opensips.org/D
>> ocumentation/Tutorials-Advanced-Accounting , it explains everything, you
>> have noway to get lost if you just read carefully.
>>
>> hope this helps.
>>
>> kkh
>>
>>
>> On Tue, Sep 26, 2017 at 11:26 AM, Ketan Kothari <kkothari...@gmail.com>
>> wrote:
>>
>>> Hello Guys,
>>>
>>> Is there any suggestion on this issue?
>>>
>>> On Fri, Sep 22, 2017 at 3:21 PM, Ketan Kothari <kkothari...@gmail.com>
>>> wrote:
>>>
>>>> Hello Guy,
>>>>
>>>> I'm using opensips 2.3.1 and when i click on system --> *CDRviewer*
>>>> its showing *"no data found*".
>>>> Is there any suggestion to resolve this issue?
>>>>
>>>> [image: Inline image 1]
>>>>
>>>> my *opensips.cfg* file :https://pastebin.com/jFP8xxRF
>>>>
>>>
>>>
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Re: [OpenSIPS-Users] No data in CDR viewer

[Khalil Khamlichi]

Hi,

honestly, I never used the opensips GUI, but I think you need first check
the database tables ; acc and missed_calls and and see if there are any
records in there. then you could activate logging on mysql and see what
queries are being run when you click that button, then third step you need
to check apache/php logs there must be an error or an indication in there.
if no luck you surely just need to re-read this
http://www.opensips.org/Documentation/Tutorials-Advanced-Accounting , it
explains everything, you have noway to get lost if you just read carefully.

hope this helps.

kkh


On Tue, Sep 26, 2017 at 11:26 AM, Ketan Kothari 
wrote:

> Hello Guys,
>
> Is there any suggestion on this issue?
>
> On Fri, Sep 22, 2017 at 3:21 PM, Ketan Kothari 
> wrote:
>
>> Hello Guy,
>>
>> I'm using opensips 2.3.1 and when i click on system --> *CDRviewer* its
>> showing *"no data found*".
>> Is there any suggestion to resolve this issue?
>>
>> [image: Inline image 1]
>>
>> my *opensips.cfg* file :https://pastebin.com/jFP8xxRF
>>
>
>
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Re: [OpenSIPS-Users] timeout settings

[Khalil Khamlichi]

Did you look at the documention of the dialog module ?

Sent from my Samsung Note

On Sep 22, 2017 8:52 PM, "Mundkowsky, Robert"  wrote:

> I assume openSIPS is following the SIP RFC for timeout settings, but is it
> possible to reduce the timeouts for things like INVITEs?
>
>
>
> Robert
>
>
>
>
>
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Re: [OpenSIPS-Users] trying to understand the E_DLG_STATE_CHANGED

[Khalil Khamlichi]

Hi Robert,

Thanks for your idea, the only issue is that it involves polling the server
and I don't want to go that path. actually with my solution the results
were quite good and precise, the reason is that a 200 OK is always an
established call and an E_ACC_CDR is always an end to a successful call and
never to another status of a call, and on both places we have access to all
dialog variables and/or acc_extra variables.

Again, Thanks for your idea.

On Tue, Sep 19, 2017 at 3:26 PM, Mundkowsky, Robert <rmundkow...@ets.org>
wrote:

> Using E_DLG_STATE_CHANGED to keep a count of active dialogs seems
> reasonable.  If you want the callerID, you would use the “hash_id of
> dialog” and the “b= hash_entry of dialog” to look up the dialog in the
> table and get more info like the callerid.  You use the dlg_list command to
> look up data in the table. I don’t know if you could do this in the
> openSIPS config file alone, you likely have to have a script that is called
> to query the table, parse the data you want, ….
>
>
>
> Looks like the key is of the form:
>
>
>
> :< hash_entry> like “1527:459551172” below
>
>
>
> Example of details from dlg_list:
>
>
>
> rmundkowsky: ~$ /export/Apps/opensips/sbin/opensipsctl fifo dlg_list
>
> database engine 'MYSQL' loaded
>
> Control engine 'FIFO' loaded
>
> entering fifo_cmd dlg_list
>
> dialog::  hash=1527:459551172 dialog_id=6558874612164
>
> state:: 3
>
> user_flags:: 0
>
> timestart:: 1505826879
>
> datestart:: 2017-09-19 09:14:39
>
> timeout:: 1505848479
>
> dateout:: 2017-09-19 15:14:39
>
> callid:: 28456fb41ba510f57257de424e73e...@xx.xx.xx.xx:5060
>
> from_uri:: sip:opens...@xx.xxx.xx.xx
>
> to_uri:: sip:7...@xx.xx.xx.xx:5060
>
> caller_tag:: as7e7bed90
>
> caller_contact:: sip:ZZZ@ XX.XX.XX.XX:5060
>
> callee_cseq:: 0
>
> caller_route_set::
>
> caller_bind_addr:: udp:XX.XX.XX.XX:5060
>
> caller_sdp::
>
> CALLEES::
>
> callee::
>
> callee_tag:: 255851520
>
> callee_contact:: sip:YYY@
> XX.XX.XX.XX:5090;transport=udp
>
> caller_cseq:: 102
>
> callee_route_set::
>
>     callee_bind_addr:: udp: XX.XX.XX.XX:5060
>
> callee_sdp::
>
> FIFO command was:
>
> :dlg_list:osips_rply_e7e14d04
>
>
>
> Robert
>
>
>
> *From:* Users [mailto:users-boun...@lists.opensips.org] *On Behalf Of *Khalil
> Khamlichi
> *Sent:* Tuesday, September 19, 2017 8:54 AM
> *To:* OpenSIPS users mailling list <users@lists.opensips.org>
> *Subject:* Re: [OpenSIPS-Users] trying to understand the
> E_DLG_STATE_CHANGED
>
>
>
> Hi Răzvan,
>
>
>
> Thanks for your answer.
>
>
>
> I ended up using onreply_route and checking if ($rs == "200") to increment
> connect_calls and then event_route[E_ACC_CDR]  to decrement
> connected_calls.
>
>
>
> multiple local tests are giving expected behavior, will need to test on
> production to confirm though.
>
>
>
> Thanks a lot for your help.
>
>
>
> On Mon, Sep 18, 2017 at 9:47 AM, Răzvan Crainea <raz...@opensips.org>
> wrote:
>
> Hi, Khalil!
>
> To be honest, I think this event was initially made to be used with the MI
> dlg_end_dlg command, which only terminates a dialog. However, you could run
> 'opensipsctl fifo dlg_list' and match the hash_id and hash_entry against
> the returned values, and then identify the callid.
>
> If you would also like to receive the callid in the event, please open a
> feature request[1].
>
> [1] https://github.com/OpenSIPS/opensips/issues
> <https://na01.safelinks.protection.outlook.com/?url=https%3A%2F%2Fgithub.com%2FOpenSIPS%2Fopensips%2Fissues=02%7C01%7Crmundkowsky%40ets.org%7Cf9c8eab9cdfd4fae4aa608d4ff5dedd6%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636414226270892544=FECgaKsHHG0bJ27vaswQ3Kv8mIs%2B%2FEIWA0BhlexCT5w%3D=0>
>
> Best regards,
>
> Răzvan Crainea
>
> OpenSIPS Developer
>
> www.opensips-solutions.com 
> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.opensips-solutions.com=02%7C01%7Crmundkowsky%40ets.org%7Cf9c8eab9cdfd4fae4aa608d4ff5dedd6%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636414226270892544=FfiJVQW07fhERDG5EmsgIAKTnzaXJ5B4Hiq%2BK69mtXk%3D=0>
>
> On 09/15/2017 10:46 PM, Khalil Khamlichi wrote:
>
> Hi everyone,
>
>
>
> I am trying to understand dialog module eventing system.
>
>
>
> I

Re: [OpenSIPS-Users] trying to understand the E_DLG_STATE_CHANGED

[Khalil Khamlichi]

Hi Răzvan,

Thanks for your answer.

I ended up using onreply_route and checking if ($rs == "200") to increment
connect_calls and then event_route[E_ACC_CDR]  to decrement
connected_calls.

multiple local tests are giving expected behavior, will need to test on
production to confirm though.

Thanks a lot for your help.

On Mon, Sep 18, 2017 at 9:47 AM, Răzvan Crainea <raz...@opensips.org> wrote:

> Hi, Khalil!
>
> To be honest, I think this event was initially made to be used with the MI
> dlg_end_dlg command, which only terminates a dialog. However, you could run
> 'opensipsctl fifo dlg_list' and match the hash_id and hash_entry against
> the returned values, and then identify the callid.
>
> If you would also like to receive the callid in the event, please open a
> feature request[1].
>
> [1] https://github.com/OpenSIPS/opensips/issues
>
> Best regards,
>
> Răzvan Crainea
> OpenSIPS Developerwww.opensips-solutions.com
>
> On 09/15/2017 10:46 PM, Khalil Khamlichi wrote:
>
> Hi everyone,
>
> I am trying to understand dialog module eventing system.
>
> I have added this route :
>
>
> event_route[E_DLG_STATE_CHANGED] {
>
> fetch_event_params("$avp(a);$avp(b);$avp(c);$avp(d);$avp(e);$avp(f)");
>
> cache_raw_query("redis:0", "PUBLISH serv1 fetch_event_params=$avp(a),$
> avp(b),$avp(c),$avp(d),$avp(e),$avp(f)", "$avp(res)");
>
> }
>
>
> so for each event I can watch an entry
>
> 1505503997.413642 [0 127.0.0.1:39734] "PUBLISH" "serv1"
> "fetch_event_params=3917,339471624,1,3,,"
> 1505503997.524535 [0 127.0.0.1:39762] "PUBLISH" "serv1"
> "fetch_event_params=3917,339471624,3,4,,"
> 1505504018.809746 [0 127.0.0.1:39840] "PUBLISH" "serv1"
> "fetch_event_params=3917,339471624,4,5,,"
>
>
> I understand that
>
> a = hash_id of dialog
> b= hash_entry of dialog
> c = old_state
> d = new_state
> e & f = do not exist and here just to prove that they do not exist.
>
> but I don't understand how to use the information of a & b to track
> dialogs ? if there a function that when fed those parameters gives us
> callid for example ?
>
> Thanks in advance.
>
> Kkh
>
>
>
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[OpenSIPS-Users] trying to understand the E_DLG_STATE_CHANGED

[Khalil Khamlichi]

Hi everyone,

I am trying to understand dialog module eventing system.

I have added this route :


event_route[E_DLG_STATE_CHANGED] {

fetch_event_params("$avp(a);$avp(b);$avp(c);$avp(d);$avp(e);$avp(f)");

cache_raw_query("redis:0", "PUBLISH serv1
fetch_event_params=$avp(a),$avp(b),$avp(c),$avp(d),$avp(e),$avp(f)",
"$avp(res)");

}


so for each event I can watch an entry

1505503997.413642 [0 127.0.0.1:39734] "PUBLISH" "serv1"
"fetch_event_params=3917,339471624,1,3,,"
1505503997.524535 [0 127.0.0.1:39762] "PUBLISH" "serv1"
"fetch_event_params=3917,339471624,3,4,,"
1505504018.809746 [0 127.0.0.1:39840] "PUBLISH" "serv1"
"fetch_event_params=3917,339471624,4,5,,"


I understand that

a = hash_id of dialog
b= hash_entry of dialog
c = old_state
d = new_state
e & f = do not exist and here just to prove that they do not exist.

but I don't understand how to use the information of a & b to track dialogs
? if there a function that when fed those parameters gives us callid for
example ?

Thanks in advance.

Kkh
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Re: [OpenSIPS-Users] OpenSIPS 2.3 - release date 16th of March

[Khalil Khamlichi]

Guys, you are amazing !


Sent from my Samsung Note

On Mar 14, 2017 9:27 AM, "Bogdan-Andrei Iancu"  wrote:

> Hi all,
>
> Heads up, the date for the 2.3 major release is set - Thursday 16th of
> March,. 2017 of course :).
>
> There is a tremendous effort do finalize all the work, in terms, of
> coding, testing, putting together docs, manuals and overviews. A detailed
> description of all new goodies in 2.3, as well as a list of changes, will
> be shortly available (before the release) - we are still compiling the
> commit logs.
>
> Still, the overview of what 2.3 is about is available here:
> https://blog.opensips.org/2017/01/12/introducing-opensips-2-3/
>
> So, finger cross and stay tuned ! Thursday is the day !
>
> Best regards,
>
> --
> Bogdan-Andrei Iancu
>   OpenSIPS Founder and Developer
>   http://www.opensips-solutions.com
>
> OpenSIPS Summit May 2017 Amsterdam
>   http://www.opensips.org/events/Summit-2017Amsterdam.html
>
>
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Re: [OpenSIPS-Users] How to quit debug mode ?

[Khalil Khamlichi]

you probably missed this in opensips.cfg

/* uncomment the following lines to enable debugging */
#debug_mode=yes


On Fri, Feb 10, 2017 at 11:01 AM, Manar Boukhari <boukhari.ma...@gmail.com>
wrote:

> I've already restarted with ctrl-c and : /etc/init.d/opensips restart
> Tried to stop it : /etc/init.d/opensips stop and starting again :
> /etc/init.d/opensips start
>
> But it's not working, the logs are being displayed continuously ...
>
> 2017-02-10 10:32 GMT+00:00 Khalil Khamlichi <khamlichi.kha...@gmail.com>:
>
>> I guess ctrl-c and then restart normally.
>>
>> On Feb 10, 2017 10:09 AM, "Manar Boukhari" <boukhari.ma...@gmail.com>
>> wrote:
>>
>>> Hi,
>>>
>>> I'm using opensips debug mode to display logs from the server.
>>> by typing : /etc/init.d/opensips debug
>>>
>>> But after that, i want to quit this mode so that logs can be saved in
>>> /var/log/opensips.log
>>>
>>> Any ideas on how to achieve this ?
>>>
>>> Best regards,
>>>
>>> Manar
>>>
>>>
>>>
>>> ___
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>>>
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>>
>>
>
>
> --
> Cordialement
> --
> BOUKHARI Manar
>
>
>
>
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Re: [OpenSIPS-Users] How to quit debug mode ?

[Khalil Khamlichi]

I guess ctrl-c and then restart normally.

On Feb 10, 2017 10:09 AM, "Manar Boukhari"  wrote:

> Hi,
>
> I'm using opensips debug mode to display logs from the server.
> by typing : /etc/init.d/opensips debug
>
> But after that, i want to quit this mode so that logs can be saved in
> /var/log/opensips.log
>
> Any ideas on how to achieve this ?
>
> Best regards,
>
> Manar
>
>
>
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Re: [OpenSIPS-Users] scheduled concurrent call reporting

[Khalil Khamlichi]

I used Liviu technique, works beautifully, we are getting very clean
reliable data right from the core of opensips into redis and from there the
world is the limit :)

On Thu, Jan 26, 2017 at 2:11 PM, Bogdan-Andrei Iancu <bog...@opensips.org>
wrote:

> Hi,
>
> You can do get_profile_size via MI interface, see:
> http://www.opensips.org/html/docs/modules/2.2.x/dialog.html#id297085
>
> And you can use xmlrpc, jsonrpc, fifo and more for the MI backend.
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 01/26/2017 04:06 PM, Khalil Khamlichi wrote:
>
> snmpstat data seems a little cryptic and difficult to parse, it might be
> just me, so please tell me about your experience using snmpstat and how
> easy was it for you to make it useful for your business case ?
>
> On Wed, Jan 25, 2017 at 9:08 PM, Abdul Basit <basit.e...@gmail.com> wrote:
>
>> Khalil,
>>
>> Why don't you consider snmpstat module for stats collection from
>> opensips?
>>
>>
>> On 25 Jan 2017 17:12, "Khalil Khamlichi" <khamlichi.kha...@gmail.com>
>> wrote:
>>
>>> Liviu, you just saved my day. my plan is to push information to redis
>>> every 10s maybe 5s if need there is, when dashboard is complete I might
>>> consider saving the stream of data for plotting and further analysis.
>>>
>>>
>>> On Wed, Jan 25, 2017 at 11:45 AM, Liviu Chircu < <li...@opensips.org>
>>> li...@opensips.org> wrote:
>>>
>>>> I do not see any problems with calling "get_profile_size()" from timer
>>>> routes, event routes, etc. As such, you can safely bypass the block
>>>> restriction using the following example:
>>>>
>>>> route [get_prof_sz_inboundCalls] {
>>>> get_profile_size("inboundCalls",,"$var(size)");
>>>> }
>>>>
>>>> route [get_prof_sz_outboundCalls] {
>>>> get_profile_size("outboundCalls",,"$var(size)");
>>>> }
>>>>
>>>> timer_route [mytimer, 1]
>>>> {
>>>> route(get_prof_sz_inboundCalls);
>>>> xlog("inboundCalls: $var(size)\n");
>>>>
>>>> route(get_prof_sz_outboundCalls);
>>>> xlog("outboundCalls: $var(size)\n");
>>>> }
>>>>
>>>> Regards,
>>>>
>>>> Liviu Chircu
>>>> OpenSIPS Developerhttp://www.opensips-solutions.com
>>>>
>>>> On 24.01.2017 01:05, Khalil Khamlichi wrote:
>>>>
>>>> the idea is that I don't need to get the number of calls 30 times per
>>>> second which would happens if I save to redis on each invite. scheduling
>>>> the saving to redis every 10 seconds serves  well my purpose but the
>>>> timer_route doesn't seem to accept get_profile_size()
>>>> any hints would be appreciated.
>>>> On Mon, Jan 23, 2017 at 10:39 PM, Khalil Khamlichi <
>>>> khamlichi.kha...@gmail.com> wrote:
>>>>>
>>>>> Hi,
>>>>> I am trying to get call load information out of opensips at regular
>>>>> intervals to get some sort of realtime view of the load on the server.
>>>>> After applying profiles on the the INVITEs, I have tried to call
>>>>> get_profile_size from a timer_route.
>>>>> I am getting error :
>>>>> CRITICAL:core:yyerror: parse error in config file
>>>>> /usr/local//etc/opensips/opensips.cfg, line 483, column 38-39:
>>>>> Command cannot be used in the block
>>>>> Any alternative solutions ?
>>>>> Thanks in advance.
>>>>> kkh
>>>>>
>>>> ___
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>>>>
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>>>> bin/mailman/listinfo/users
>>>
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Re: [OpenSIPS-Users] scheduled concurrent call reporting

[Khalil Khamlichi]

snmpstat data seems a little cryptic and difficult to parse, it might be
just me, so please tell me about your experience using snmpstat and how
easy was it for you to make it useful for your business case ?

On Wed, Jan 25, 2017 at 9:08 PM, Abdul Basit <basit.e...@gmail.com> wrote:

> Khalil,
>
> Why don't you consider snmpstat module for stats collection from opensips?
>
>
> On 25 Jan 2017 17:12, "Khalil Khamlichi" <khamlichi.kha...@gmail.com>
> wrote:
>
>> Liviu, you just saved my day. my plan is to push information to redis
>> every 10s maybe 5s if need there is, when dashboard is complete I might
>> consider saving the stream of data for plotting and further analysis.
>>
>>
>> On Wed, Jan 25, 2017 at 11:45 AM, Liviu Chircu <li...@opensips.org>
>> wrote:
>>
>>> I do not see any problems with calling "get_profile_size()" from timer
>>> routes, event routes, etc. As such, you can safely bypass the block
>>> restriction using the following example:
>>>
>>> route [get_prof_sz_inboundCalls] {
>>> get_profile_size("inboundCalls",,"$var(size)");
>>> }
>>>
>>> route [get_prof_sz_outboundCalls] {
>>> get_profile_size("outboundCalls",,"$var(size)");
>>> }
>>>
>>> timer_route [mytimer, 1]
>>> {
>>> route(get_prof_sz_inboundCalls);
>>> xlog("inboundCalls: $var(size)\n");
>>>
>>> route(get_prof_sz_outboundCalls);
>>> xlog("outboundCalls: $var(size)\n");
>>> }
>>>
>>> Regards,
>>>
>>> Liviu Chircu
>>> OpenSIPS Developerhttp://www.opensips-solutions.com
>>>
>>> On 24.01.2017 01:05, Khalil Khamlichi wrote:
>>>
>>> the idea is that I don't need to get the number of calls 30 times per
>>> second which would happens if I save to redis on each invite. scheduling
>>> the saving to redis every 10 seconds serves  well my purpose but the
>>> timer_route doesn't seem to accept get_profile_size()
>>>
>>> any hints would be appreciated.
>>>
>>> On Mon, Jan 23, 2017 at 10:39 PM, Khalil Khamlichi <
>>> <khamlichi.kha...@gmail.com>khamlichi.kha...@gmail.com> wrote:
>>>
>>>> Hi,
>>>>
>>>> I am trying to get call load information out of opensips at regular
>>>> intervals to get some sort of realtime view of the load on the server.
>>>>
>>>> After applying profiles on the the INVITEs, I have tried to call
>>>> get_profile_size from a timer_route.
>>>> I am getting error :
>>>>
>>>> CRITICAL:core:yyerror: parse error in config file
>>>> /usr/local//etc/opensips/opensips.cfg, line 483, column 38-39: Command
>>>> cannot be used in the block
>>>>
>>>> Any alternative solutions ?
>>>>
>>>> Thanks in advance.
>>>>
>>>> kkh
>>>>
>>>
>>>
>>>
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>>>
>>>
>>>
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Re: [OpenSIPS-Users] how to catch hangup event inside opensips

[Khalil Khamlichi]

sorry the expression "run a route" was misleading, actually we are working
on a dashboard, we need to catch events happening inside opensips to feed
our counters, we hope to be able to catch hangup event and probably the
sip_code so that we can have counters for every sip code specific to every
gateway. the other way around this is to catch the cdr insert and feed
counters.

On Wed, Jan 25, 2017 at 12:32 PM, Stefano Pisani <stefano.pis...@omnianet.it
> wrote:

> please explain better
>
> Il 25 Gennaio 2017 13:24:57 CET, Khalil Khamlichi <
> khamlichi.kha...@gmail.com> ha scritto:
>>
>> there is one more question we still struggling with, how to run a route
>> at each hangup, we never found any way to do it inside opensips. right now
>> we catching the event upon cdr insert in db.
>>
>
> -- Inviato dal mio cellulare Android con K-9 Mail.
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[OpenSIPS-Users] how to catch hangup event inside opensips

[Khalil Khamlichi]

there is one more question we still struggling with, how to run a route at
each hangup, we never found any way to do it inside opensips. right now we
catching the event upon cdr insert in db.
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Re: [OpenSIPS-Users] scheduled concurrent call reporting

[Khalil Khamlichi]

Liviu, you just saved my day. my plan is to push information to redis every
10s maybe 5s if need there is, when dashboard is complete I might consider
saving the stream of data for plotting and further analysis.


On Wed, Jan 25, 2017 at 11:45 AM, Liviu Chircu <li...@opensips.org> wrote:

> I do not see any problems with calling "get_profile_size()" from timer
> routes, event routes, etc. As such, you can safely bypass the block
> restriction using the following example:
>
> route [get_prof_sz_inboundCalls] {
> get_profile_size("inboundCalls",,"$var(size)");
> }
>
> route [get_prof_sz_outboundCalls] {
> get_profile_size("outboundCalls",,"$var(size)");
> }
>
> timer_route [mytimer, 1]
> {
> route(get_prof_sz_inboundCalls);
> xlog("inboundCalls: $var(size)\n");
>
> route(get_prof_sz_outboundCalls);
> xlog("outboundCalls: $var(size)\n");
> }
>
> Regards,
>
> Liviu Chircu
> OpenSIPS Developerhttp://www.opensips-solutions.com
>
> On 24.01.2017 01:05, Khalil Khamlichi wrote:
>
> the idea is that I don't need to get the number of calls 30 times per
> second which would happens if I save to redis on each invite. scheduling
> the saving to redis every 10 seconds serves  well my purpose but the
> timer_route doesn't seem to accept get_profile_size()
>
> any hints would be appreciated.
>
> On Mon, Jan 23, 2017 at 10:39 PM, Khalil Khamlichi <
> <khamlichi.kha...@gmail.com>khamlichi.kha...@gmail.com> wrote:
>
>> Hi,
>>
>> I am trying to get call load information out of opensips at regular
>> intervals to get some sort of realtime view of the load on the server.
>>
>> After applying profiles on the the INVITEs, I have tried to call
>> get_profile_size from a timer_route.
>> I am getting error :
>>
>> CRITICAL:core:yyerror: parse error in config file
>> /usr/local//etc/opensips/opensips.cfg, line 483, column 38-39: Command
>> cannot be used in the block
>>
>> Any alternative solutions ?
>>
>> Thanks in advance.
>>
>> kkh
>>
>
>
>
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Re: [OpenSIPS-Users] topology hiding in opensips

[Khalil Khamlichi]

sorry :

# record routing
   record_route();

   # create dialog with timeout
   if ( !create_dialog("B") ) {
   send_reply("500","Internal Server Error");
   exit;
   }

# apply transformations from dialplan table
   dp_translate("0","$rU/$rU");

   # route calls based on prefix
   if ( !do_routing("1","W",,"$var(rule_attrs)","$var(gw_attrs)") ) {
   send_reply("404","No Route found");
   exit;
   }

   $acc_extra(gwid)=$avp(gw_id);

   t_on_failure("GW_FAILOVER");

   do_accounting("db|log","cdr|missed",);
#NAT
   if (isbflagset(NAT)) setflag(NAT);
#NAT
#TOPOHIDE
toopology_hiding("UC");
#TOPOHIDE
   route(RELAY);

}
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Re: [OpenSIPS-Users] scheduled concurrent call reporting

[Khalil Khamlichi]

I looked up at the docs, $stat(active_dialogs)  is for the whole system,
If there is no way to call get_profile_size() automatically from whithin
opensips I will need to fall back to MI fifo from an external script.

On Tue, Jan 24, 2017 at 8:49 AM, Khalil Khamlichi <
khamlichi.kha...@gmail.com> wrote:

> Hi Răzvan, would $stat(active_dialogs) give detailed stats ? I mean for
> each label I put on the calls ? example if I am labeling by IP would I get
> from $stat(active_dialogs) stats by label ?
>
> On Tue, Jan 24, 2017 at 8:38 AM, Răzvan Crainea <raz...@opensips.org>
> wrote:
>
>> Hello, Khalil!
>>
>> You could check the $stat(active_dialogs) pvar, which will return the
>> number of ongoing dialogs.
>>
>> Best regards,
>>
>> Răzvan Crainea
>> OpenSIPS Core Developer
>> http://www.opensips-solutions.com
>>
>> On 01/24/2017 01:05 AM, Khalil Khamlichi wrote:
>>
>>> the idea is that I don't need to get the number of calls 30 times per
>>> second which would happens if I save to redis on each invite. scheduling
>>> the saving to redis every 10 seconds serves  well my purpose but the
>>> timer_route doesn't seem to accept get_profile_size()
>>>
>>> any hints would be appreciated.
>>>
>>> On Mon, Jan 23, 2017 at 10:39 PM, Khalil Khamlichi
>>> <khamlichi.kha...@gmail.com <mailto:khamlichi.kha...@gmail.com>> wrote:
>>>
>>> Hi,
>>>
>>> I am trying to get call load information out of opensips at regular
>>> intervals to get some sort of realtime view of the load on the
>>> server.
>>>
>>> After applying profiles on the the INVITEs, I have tried to call
>>> get_profile_size from a timer_route.
>>> I am getting error :
>>>
>>> CRITICAL:core:yyerror: parse error in config file
>>> /usr/local//etc/opensips/opensips.cfg, line 483, column 38-39:
>>> Command cannot be used in the block
>>>
>>> Any alternative solutions ?
>>>
>>> Thanks in advance.
>>>
>>> kkh
>>>
>>>
>>>
>>>
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>>>
>>>
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Re: [OpenSIPS-Users] scheduled concurrent call reporting

[Khalil Khamlichi]

Hi Răzvan, would $stat(active_dialogs) give detailed stats ? I mean for
each label I put on the calls ? example if I am labeling by IP would I get
from $stat(active_dialogs) stats by label ?

On Tue, Jan 24, 2017 at 8:38 AM, Răzvan Crainea <raz...@opensips.org> wrote:

> Hello, Khalil!
>
> You could check the $stat(active_dialogs) pvar, which will return the
> number of ongoing dialogs.
>
> Best regards,
>
> Răzvan Crainea
> OpenSIPS Core Developer
> http://www.opensips-solutions.com
>
> On 01/24/2017 01:05 AM, Khalil Khamlichi wrote:
>
>> the idea is that I don't need to get the number of calls 30 times per
>> second which would happens if I save to redis on each invite. scheduling
>> the saving to redis every 10 seconds serves  well my purpose but the
>> timer_route doesn't seem to accept get_profile_size()
>>
>> any hints would be appreciated.
>>
>> On Mon, Jan 23, 2017 at 10:39 PM, Khalil Khamlichi
>> <khamlichi.kha...@gmail.com <mailto:khamlichi.kha...@gmail.com>> wrote:
>>
>> Hi,
>>
>> I am trying to get call load information out of opensips at regular
>> intervals to get some sort of realtime view of the load on the server.
>>
>> After applying profiles on the the INVITEs, I have tried to call
>> get_profile_size from a timer_route.
>> I am getting error :
>>
>> CRITICAL:core:yyerror: parse error in config file
>> /usr/local//etc/opensips/opensips.cfg, line 483, column 38-39:
>> Command cannot be used in the block
>>
>> Any alternative solutions ?
>>
>> Thanks in advance.
>>
>> kkh
>>
>>
>>
>>
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Re: [OpenSIPS-Users] scheduled concurrent call reporting

[Khalil Khamlichi]

the idea is that I don't need to get the number of calls 30 times per
second which would happens if I save to redis on each invite. scheduling
the saving to redis every 10 seconds serves  well my purpose but the
timer_route doesn't seem to accept get_profile_size()

any hints would be appreciated.

On Mon, Jan 23, 2017 at 10:39 PM, Khalil Khamlichi <
khamlichi.kha...@gmail.com> wrote:

> Hi,
>
> I am trying to get call load information out of opensips at regular
> intervals to get some sort of realtime view of the load on the server.
>
> After applying profiles on the the INVITEs, I have tried to call
> get_profile_size from a timer_route.
> I am getting error :
>
> CRITICAL:core:yyerror: parse error in config file 
> /usr/local//etc/opensips/opensips.cfg,
> line 483, column 38-39: Command cannot be used in the block
>
> Any alternative solutions ?
>
> Thanks in advance.
>
> kkh
>
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[OpenSIPS-Users] scheduled concurrent call reporting

[Khalil Khamlichi]

Hi,

I am trying to get call load information out of opensips at regular
intervals to get some sort of realtime view of the load on the server.

After applying profiles on the the INVITEs, I have tried to call
get_profile_size from a timer_route.
I am getting error :

CRITICAL:core:yyerror: parse error in config file
/usr/local//etc/opensips/opensips.cfg, line 483, column 38-39: Command
cannot be used in the block

Any alternative solutions ?

Thanks in advance.

kkh
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Re: [OpenSIPS-Users] opensips $Revision: 4448 $

[Khalil Khamlichi]

I confirm, your suggestion works perfect :

failure_route[GW_FAILOVER] {

   if (t_was_cancelled()) {
   exit;
   }

   # detect failure and redirect to next available GW
   if (t_check_status("(408)|488|([56][0-9][0-9])")) {
   xlog("Failed GW $rd detected \n");

   if ( use_next_gw() ) {
   t_on_failure("GW_FAILOVER");
   *do_accounting("db|log","failed|missed",);*
   t_relay();
   exit;
   }

   send_reply("500","All GW are down");
   }
}




On Mon, Jan 16, 2017 at 11:35 AM, Răzvan Crainea <raz...@opensips.org>
wrote:

> Yes, please try that. To be honest, I've never done it using the new
> accounting module, but this might do the trick.
> Also, did you take a look at the multi-legging accounting?
>
> Best regards,
>
> Răzvan Crainea
> OpenSIPS Solutionswww.opensips-solutions.com
>
> On 01/16/2017 01:18 PM, Khalil Khamlichi wrote:
>
> Oki, I will modify my old failure_route :
>
>
> failure_route[GW_FAILOVER] {
>
>if (t_was_cancelled()) {
>exit;
>}
>
># detect failure and redirect to next available GW
>if (t_check_status("(408)|488|([56][0-9][0-9])")) {
> #xlog("Failed GW $rd detected \n");
>
>if ( use_next_gw() ) {
>t_on_failure("GW_FAILOVER");
>t_relay();
>exit;
>}
>
>send_reply("500","All GW are down");
>}
> }
>
>
> to :
>
>
> failure_route[GW_FAILOVER] {
>
>if (t_was_cancelled()) {
>exit;
>}
>
># detect failure and redirect to next available GW
>if (t_check_status("(408)|488|([56][0-9][0-9])")) {
>xlog("Failed GW $rd detected \n");
>
>if ( use_next_gw() ) {
>t_on_failure("GW_FAILOVER");
>*do_accounting("db|log","failed|missed",);*
>t_relay();
>exit;
>}
>
>send_reply("500","All GW are down");
>}
> }
>
>
> right ?
>
> On Mon, Jan 16, 2017 at 11:07 AM, Răzvan Crainea <raz...@opensips.org>
> wrote:
>
>> I was asking you to call do_accounting() in failure route, for each leg.
>>
>> Best regards,
>>
>> Răzvan Crainea
>> OpenSIPS Solutionswww.opensips-solutions.com
>>
>> On 01/16/2017 12:02 PM, Khalil Khamlichi wrote:
>>
>> sorry mistype, I am calling do_accounting() twice.
>>
>> On Mon, Jan 16, 2017 at 9:47 AM, Khalil Khamlichi <
>> khamlichi.kha...@gmail.com> wrote:
>>
>>> thanks for your much appreciated help,
>>> I am calling do_routing twice.
>>>
>>> Here is my actual opensips.cfg :
>>>
>>> route {
>>> ...
>>> ...
>>>
>>> ...
>>> ...
>>> if (has_totag()) {
>>># sequential request withing a dialog should
>>># take the path determined by record-routing
>>>if (loose_route()) {
>>>
>>># validate the sequential request against dialog
>>>if ( $DLG_status!=NULL && !validate_dialog() ) {
>>>xlog("In-Dialog $rm from $si (callid=$ci)
>>> is not valid according to dialog\n");
>>>}
>>>
>>>if (is_method("BYE")) {
>>># do accounting even if the transaction
>>> fails
>>>do_accounting("db|log","failed|missed",);
>>>} else if (is_method("INVITE")) {
>>># even if in most of the cases is
>>> useless, do RR for
>>># re-INVITEs alos, as some buggy clients
>>> do change route set
>>># during the dialog.
>>>record_route();
>>>}
>>> ...
>>> ...
>>>
>>> ...
>>> ...
>>> $acc_extra(gwid)=$avp(gw_id);
>>>
>>>t_on_fail

Re: [OpenSIPS-Users] opensips $Revision: 4448 $

[Khalil Khamlichi]

multi-legging accounting is the next step in my planning, will let you
updated about it.
I will be testing this later on today, I just hope $avp(gw_id) gets updated
too on the cdr.

On Mon, Jan 16, 2017 at 11:35 AM, Răzvan Crainea <raz...@opensips.org>
wrote:

> Yes, please try that. To be honest, I've never done it using the new
> accounting module, but this might do the trick.
> Also, did you take a look at the multi-legging accounting?
>
> Best regards,
>
> Răzvan Crainea
> OpenSIPS Solutionswww.opensips-solutions.com
>
> On 01/16/2017 01:18 PM, Khalil Khamlichi wrote:
>
> Oki, I will modify my old failure_route :
>
>
> failure_route[GW_FAILOVER] {
>
>if (t_was_cancelled()) {
>exit;
>}
>
># detect failure and redirect to next available GW
>if (t_check_status("(408)|488|([56][0-9][0-9])")) {
> #xlog("Failed GW $rd detected \n");
>
>if ( use_next_gw() ) {
>t_on_failure("GW_FAILOVER");
>t_relay();
>exit;
>}
>
>send_reply("500","All GW are down");
>}
> }
>
>
> to :
>
>
> failure_route[GW_FAILOVER] {
>
>if (t_was_cancelled()) {
>exit;
>}
>
># detect failure and redirect to next available GW
>if (t_check_status("(408)|488|([56][0-9][0-9])")) {
>xlog("Failed GW $rd detected \n");
>
>if ( use_next_gw() ) {
>t_on_failure("GW_FAILOVER");
>*do_accounting("db|log","failed|missed",);*
>t_relay();
>exit;
>}
>
>send_reply("500","All GW are down");
>}
> }
>
>
> right ?
>
> On Mon, Jan 16, 2017 at 11:07 AM, Răzvan Crainea <raz...@opensips.org>
> wrote:
>
>> I was asking you to call do_accounting() in failure route, for each leg.
>>
>> Best regards,
>>
>> Răzvan Crainea
>> OpenSIPS Solutionswww.opensips-solutions.com
>>
>> On 01/16/2017 12:02 PM, Khalil Khamlichi wrote:
>>
>> sorry mistype, I am calling do_accounting() twice.
>>
>> On Mon, Jan 16, 2017 at 9:47 AM, Khalil Khamlichi <
>> khamlichi.kha...@gmail.com> wrote:
>>
>>> thanks for your much appreciated help,
>>> I am calling do_routing twice.
>>>
>>> Here is my actual opensips.cfg :
>>>
>>> route {
>>> ...
>>> ...
>>>
>>> ...
>>> ...
>>> if (has_totag()) {
>>># sequential request withing a dialog should
>>># take the path determined by record-routing
>>>if (loose_route()) {
>>>
>>># validate the sequential request against dialog
>>>if ( $DLG_status!=NULL && !validate_dialog() ) {
>>>xlog("In-Dialog $rm from $si (callid=$ci)
>>> is not valid according to dialog\n");
>>>}
>>>
>>>if (is_method("BYE")) {
>>># do accounting even if the transaction
>>> fails
>>>do_accounting("db|log","failed|missed",);
>>>} else if (is_method("INVITE")) {
>>># even if in most of the cases is
>>> useless, do RR for
>>># re-INVITEs alos, as some buggy clients
>>> do change route set
>>>    # during the dialog.
>>>record_route();
>>>}
>>> ...
>>> ...
>>>
>>> ...
>>> ...
>>> $acc_extra(gwid)=$avp(gw_id);
>>>
>>>t_on_failure("GW_FAILOVER");
>>>
>>>do_accounting("db|log","cdr|missed",);
>>> #NAT
>>>if (isbflagset(NAT)) setflag(NAT);
>>> #NAT
>>>route(RELAY);
>>> }
>>>
>>> # END OF MAIN ROUTE
>>>
>>>
>>> On Mon, Jan 16, 2017 at 8:51 AM, Răzvan Crainea <raz...@opensips.org>
>>> wrote:
>>>
>>>> Hi, Khalil!
>>>>
>&

Re: [OpenSIPS-Users] opensips $Revision: 4448 $

[Khalil Khamlichi]

Oki, I will modify my old failure_route :


failure_route[GW_FAILOVER] {

   if (t_was_cancelled()) {
   exit;
   }

   # detect failure and redirect to next available GW
   if (t_check_status("(408)|488|([56][0-9][0-9])")) {
#xlog("Failed GW $rd detected \n");

   if ( use_next_gw() ) {
   t_on_failure("GW_FAILOVER");
   t_relay();
   exit;
   }

   send_reply("500","All GW are down");
   }
}


to :


failure_route[GW_FAILOVER] {

   if (t_was_cancelled()) {
   exit;
   }

   # detect failure and redirect to next available GW
   if (t_check_status("(408)|488|([56][0-9][0-9])")) {
   xlog("Failed GW $rd detected \n");

   if ( use_next_gw() ) {
   t_on_failure("GW_FAILOVER");
   *do_accounting("db|log","failed|missed",);*
   t_relay();
   exit;
   }

   send_reply("500","All GW are down");
   }
}


right ?

On Mon, Jan 16, 2017 at 11:07 AM, Răzvan Crainea <raz...@opensips.org>
wrote:

> I was asking you to call do_accounting() in failure route, for each leg.
>
> Best regards,
>
> Răzvan Crainea
> OpenSIPS Solutionswww.opensips-solutions.com
>
> On 01/16/2017 12:02 PM, Khalil Khamlichi wrote:
>
> sorry mistype, I am calling do_accounting() twice.
>
> On Mon, Jan 16, 2017 at 9:47 AM, Khalil Khamlichi <
> khamlichi.kha...@gmail.com> wrote:
>
>> thanks for your much appreciated help,
>> I am calling do_routing twice.
>>
>> Here is my actual opensips.cfg :
>>
>> route {
>> ...
>> ...
>>
>> ...
>> ...
>> if (has_totag()) {
>># sequential request withing a dialog should
>># take the path determined by record-routing
>>if (loose_route()) {
>>
>># validate the sequential request against dialog
>>if ( $DLG_status!=NULL && !validate_dialog() ) {
>>xlog("In-Dialog $rm from $si (callid=$ci)
>> is not valid according to dialog\n");
>>}
>>
>>if (is_method("BYE")) {
>># do accounting even if the transaction
>> fails
>>do_accounting("db|log","failed|missed",);
>>} else if (is_method("INVITE")) {
>># even if in most of the cases is
>> useless, do RR for
>># re-INVITEs alos, as some buggy clients
>> do change route set
>># during the dialog.
>>record_route();
>>}
>> ...
>> ...
>>
>> ...
>> ...
>> $acc_extra(gwid)=$avp(gw_id);
>>
>>t_on_failure("GW_FAILOVER");
>>
>>do_accounting("db|log","cdr|missed",);
>> #NAT
>>if (isbflagset(NAT)) setflag(NAT);
>> #NAT
>>route(RELAY);
>> }
>>
>> # END OF MAIN ROUTE
>>
>>
>> On Mon, Jan 16, 2017 at 8:51 AM, Răzvan Crainea <raz...@opensips.org>
>> wrote:
>>
>>> Hi, Khalil!
>>>
>>> Did you try to call the do_accounting() function for each leg going to
>>> the next gateway?
>>>
>>> Best regards,
>>>
>>> Răzvan Crainea
>>> OpenSIPS Solutionswww.opensips-solutions.com
>>>
>>> On 01/15/2017 12:24 AM, Khalil Khamlichi wrote:
>>>
>>> Hi,
>>> I am testing opensips 2.2, 2.3 I have tried to configure acc module to
>>> save to db failed calls on drouting configuration, I have found that it
>>> does save only the first failed call (that is the first gateway) it does
>>> not save to database any other failures on second and third gateways that
>>> are tried for the call.
>>> is this the expected behavioure ?
>>> Thanks for your help.
>>>
>>> regards,
>>> kh
>>>
>>>
>>> ___
>>> Users mailing 
>>> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
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>>> Users@lists.opensips.org http://lists.opensips.org/cgi-
>>> bin/mailman/listinfo/users
>>
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Re: [OpenSIPS-Users] opensips $Revision: 4448 $

[Khalil Khamlichi]

sorry mistype, I am calling do_accounting() twice.

On Mon, Jan 16, 2017 at 9:47 AM, Khalil Khamlichi <
khamlichi.kha...@gmail.com> wrote:

> thanks for your much appreciated help,
> I am calling do_routing twice.
>
> Here is my actual opensips.cfg :
>
> route {
> ...
> ...
>
> ...
> ...
> if (has_totag()) {
># sequential request withing a dialog should
># take the path determined by record-routing
>if (loose_route()) {
>
># validate the sequential request against dialog
>if ( $DLG_status!=NULL && !validate_dialog() ) {
>xlog("In-Dialog $rm from $si (callid=$ci)
> is not valid according to dialog\n");
>}
>
>if (is_method("BYE")) {
># do accounting even if the transaction
> fails
>do_accounting("db|log","failed|missed",);
>} else if (is_method("INVITE")) {
># even if in most of the cases is useless,
> do RR for
># re-INVITEs alos, as some buggy clients
> do change route set
># during the dialog.
>record_route();
>}
> ...
> ...
>
> ...
> ...
> $acc_extra(gwid)=$avp(gw_id);
>
>t_on_failure("GW_FAILOVER");
>
>do_accounting("db|log","cdr|missed",);
> #NAT
>if (isbflagset(NAT)) setflag(NAT);
> #NAT
>route(RELAY);
> }
>
> # END OF MAIN ROUTE
>
>
> On Mon, Jan 16, 2017 at 8:51 AM, Răzvan Crainea <raz...@opensips.org>
> wrote:
>
>> Hi, Khalil!
>>
>> Did you try to call the do_accounting() function for each leg going to
>> the next gateway?
>>
>> Best regards,
>>
>> Răzvan Crainea
>> OpenSIPS Solutionswww.opensips-solutions.com
>>
>> On 01/15/2017 12:24 AM, Khalil Khamlichi wrote:
>>
>> Hi,
>> I am testing opensips 2.2, 2.3 I have tried to configure acc module to
>> save to db failed calls on drouting configuration, I have found that it
>> does save only the first failed call (that is the first gateway) it does
>> not save to database any other failures on second and third gateways that
>> are tried for the call.
>> is this the expected behavioure ?
>> Thanks for your help.
>>
>> regards,
>> kh
>>
>>
>> ___
>> Users mailing 
>> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>>
>> ___
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>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>
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Re: [OpenSIPS-Users] opensips $Revision: 4448 $

[Khalil Khamlichi]

thanks for your much appreciated help,
I am calling do_routing twice.

Here is my actual opensips.cfg :

route {
...
...

...
...
if (has_totag()) {
   # sequential request withing a dialog should
   # take the path determined by record-routing
   if (loose_route()) {

   # validate the sequential request against dialog
   if ( $DLG_status!=NULL && !validate_dialog() ) {
   xlog("In-Dialog $rm from $si (callid=$ci) is
not valid according to dialog\n");
   }

   if (is_method("BYE")) {
   # do accounting even if the transaction fails
   do_accounting("db|log","failed|missed",);
   } else if (is_method("INVITE")) {
   # even if in most of the cases is useless,
do RR for
   # re-INVITEs alos, as some buggy clients do
change route set
   # during the dialog.
   record_route();
   }
...
...

...
...
$acc_extra(gwid)=$avp(gw_id);

   t_on_failure("GW_FAILOVER");

   do_accounting("db|log","cdr|missed",);
#NAT
   if (isbflagset(NAT)) setflag(NAT);
#NAT
   route(RELAY);
}

# END OF MAIN ROUTE


On Mon, Jan 16, 2017 at 8:51 AM, Răzvan Crainea <raz...@opensips.org> wrote:

> Hi, Khalil!
>
> Did you try to call the do_accounting() function for each leg going to the
> next gateway?
>
> Best regards,
>
> Răzvan Crainea
> OpenSIPS Solutionswww.opensips-solutions.com
>
> On 01/15/2017 12:24 AM, Khalil Khamlichi wrote:
>
> Hi,
> I am testing opensips 2.2, 2.3 I have tried to configure acc module to
> save to db failed calls on drouting configuration, I have found that it
> does save only the first failed call (that is the first gateway) it does
> not save to database any other failures on second and third gateways that
> are tried for the call.
> is this the expected behavioure ?
> Thanks for your help.
>
> regards,
> kh
>
>
> ___
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> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
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[OpenSIPS-Users] opensips $Revision: 4448 $

[Khalil Khamlichi]

Hi,
I am testing opensips 2.2, 2.3 I have tried to configure acc module to save
to db failed calls on drouting configuration, I have found that it does
save only the first failed call (that is the first gateway) it does not
save to database any other failures on second and third gateways that are
tried for the call.
is this the expected behavioure ?
Thanks for your help.

regards,
kh
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Re: [OpenSIPS-Users] OpenSips 2.2 Error

[Khalil Khamlichi]

you need to set this parameter for the module to load correctly

Jan  9 12:30:34 dev opensips: ERROR:core:set_mod_param_regex: no module
matching aaa_radius found

On Mon, Jan 9, 2017 at 10:36 AM, Dragomir Haralambiev 
wrote:

> Hello,
>
> Here is my problem:
>
> OpenSips 2.2.2 - latest
> OS - CentOS Linux release 7.3.1611
>
> opensips.cfg
> .
> loadmodule "aaa_radius.so"
> ..
>
> aaa_radius.so exists in /usr/local/lib64/opensips/modules
>
> Here is list of Eroors
>
>
> Jan  9 12:30:34 dev opensips: ERROR:core:sr_load_module: could not open
> module :
> /usr/local/lib64/opensips/modules/aaa_radius.so: undefined symbol:
> rc_acct_resume
> Jan  9 12:30:34 dev opensips: ERROR:core:load_module: failed to load module
> Jan  9 12:30:34 dev opensips: CRITICAL:core:yyerror: parse error in config
> file /etc/opensips/opensips.cfg, line 53, column 13-14: failed to load
> module aaa_radius.so
> Jan  9 12:30:34 dev opensips: ERROR:core:set_mod_param_regex: no module
> matching aaa_radius found
>
>
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Re: [OpenSIPS-Users] do_routing function first argument

[Khalil Khamlichi]

Well you get to my point, shouldn't it be in the database then ? whats the
point in having it hard coded inside the config file ?

On Sun, Jan 8, 2017 at 1:24 PM, Johan De Clercq <jo...@democon.be> wrote:

> I think that's route set id.
>
> On 08 Jan 2017 12:12 PM, "Khalil Khamlichi" <khamlichi.kha...@gmail.com>
> wrote:
>
>> Hi eveyone,
>>
>> I have been trying to understand the first argument to the d
>> o_routing("1","",,"$var(rule_attrs)")
>>
>> Can anyone please give some explanation of the first argument to this
>> function ? what does the "1" stands for and where does it fit in the logic
>> back in the database setup (in my scenario it's mostly drouting tables)
>>
>> Thanks in advance.
>>
>>
>>
>> ___
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>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
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[OpenSIPS-Users] do_routing function first argument

[Khalil Khamlichi]

Hi eveyone,

I have been trying to understand the first argument to the d
o_routing("1","",,"$var(rule_attrs)")

Can anyone please give some explanation of the first argument to this
function ? what does the "1" stands for and where does it fit in the logic
back in the database setup (in my scenario it's mostly drouting tables)

Thanks in advance.
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