[OpenSIPS-Users] How to put mediaproxy and opensips on 2 different machine?

Thu Jan 30 14:29:33 CET 2014

Hi,

I'm trying to put opensips on one and mediaproxy on second server.

Is it possible ?

If yes ! how we can configure mediaproxy "config.ini" for opensips and what
we need to put in opensips.cfg for mediaproxy ?

Pls help

Regds
Chandra Prakash

-----Original Message-----
From: users-bounces at lists.opensips.org
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Sent: Thursday, January 30, 2014 4:30 PM
To: users at lists.opensips.org
Subject: Users Digest, Vol 66, Issue 48

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Today's Topics:

   1. Re: call control module (Muhammad Shahzad Shafi)
   2. Store $DLG_dir in acc Database (Wilmar Campos)
   3. Re: call control module (Eddie Chan)
   4. mediaproxy (dotnetdub)
   5. Load_credentials => output to xlog(); (Alectronic)
   6. Re: Store $DLG_dir in acc Database (R?zvan Crainea)

----------------------------------------------------------------------

Message: 1
Date: Wed, 29 Jan 2014 12:36:24 +0100
From: Muhammad Shahzad Shafi <shahzad at voip-demos.com>
Subject: Re: [OpenSIPS-Users] call control module
To: <users at lists.opensips.org>
Message-ID: <7844762b2c461fec7022a1f2d8336e52 at voip-demos.com>
Content-Type: text/plain; charset="utf-8"

All SIP requests generated by OpenSIPS appear in local_route. So you can
filter BYE generated by opensips there. 

Also using is_direction
method you can determine who sent BYE (caller or callee).

http://www.opensips.org/html/docs/modules/devel/rr.html#id293720

Since BYE generated by opensips is sent in both directions, so you probably
want to do accounting only for one BYE (to avoid duplication).

Thank you. 

On 2014-01-27 23:39, Eddie Chan wrote: 

> Hi all,
> 
>
I am having problem triggering CDR when the max call duration in call
control module timeout. 
> 
> For a normal call, I use setflag in the
main routing logic to trigger CDRs. 
> 
> route(
> 
> ? 
> 
> If
(is_method("BYE")) { 
> 
> ? 
> 
> setflag(AAA_DO);
> 
> }
> 
>
However, when the call control timer expired, it will generate two BYE
messages to each endpoints. Since the BYE messages were not originated by
the endpoints, the main routing loop cannot detect the BYE message and thus
failed to generate CDR. 
> 
> Can anyone give me some idea on
this problem? Where should I put the setflag if the BYE messages were
originated by Opensip itself? 
> 
> Thanks,
> 
> Eddie

--
Mit
freundlichen Gr??en
Muhammad
Shahzad
-----------------------------------
CISCO Rich Media
Communication Specialist (CRMCS)
CISCO Certified Network Associate
(CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_786pk at hotmail.com
Email:
shaheryarkh at googlemail.com

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Message: 2
Date: Wed, 29 Jan 2014 14:15:03 -0500
From: Wilmar Campos <wilmar.campos at gmail.com>
Subject: [OpenSIPS-Users] Store $DLG_dir in acc Database
To: OpenSIPS users mailling list <users at lists.opensips.org>
Message-ID:
	<CAH_cxxHS0rHm-_Kejc78e=eVCiRun6nHrA4yTpzGsNbUqUCHUg at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"

Hi All,

Can someone please point me to the right direction on how to store the
$DLG_dir into the accounting cdr?

Thanks,

Wilmar
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Message: 3
Date: Thu, 30 Jan 2014 08:59:45 +1100
From: "Eddie Chan" <e.chan at ivstel.com>
Subject: Re: [OpenSIPS-Users] call control module
To: <shahzad at voip-demos.com>,	"'OpenSIPS users mailling list'"
	<users at lists.opensips.org>
Message-ID: <52e979e0.2742440a.3e34.6a87 at mx.google.com>
Content-Type: text/plain; charset="utf-8"

Thanks Muhammad.   It works.

From: users-bounces at lists.opensips.org
[mailto:users-bounces at lists.opensips.org] On Behalf Of Muhammad Shahzad
Shafi
Sent: Wednesday, January 29, 2014 10:36 PM
To: users at lists.opensips.org
Subject: Re: [OpenSIPS-Users] call control module

All SIP requests generated by OpenSIPS appear in local_route. So you can
filter BYE generated by opensips there.

Also using is_direction method you can determine who sent BYE (caller or
callee).

http://www.opensips.org/html/docs/modules/devel/rr.html#id293720

Since BYE generated by opensips is sent in both directions, so you probably
want to do accounting only for one BYE (to avoid duplication).

Thank you.

On 2014-01-27 23:39, Eddie Chan wrote:

Hi all,

I am having problem triggering CDR when the max call duration in call
control module timeout.

For a normal call, I use setflag in the main routing logic to trigger CDRs.

route(

?

                If (is_method(?BYE?)) {

                                ?

                                setflag(AAA_DO);

                }

However, when the call control timer expired, it will generate two BYE
messages to each endpoints.   Since the BYE messages were not originated by
the endpoints, the main routing loop cannot detect the BYE message and thus
failed to generate CDR.

Can anyone give me some idea on this problem?  Where should I put the
setflag if the BYE messages were originated by Opensip itself?

Thanks,

Eddie

--
Mit freundlichen Gr??en
Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network
Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_786pk at hotmail.com
Email: shaheryarkh at googlemail.com
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Message: 4
Date: Thu, 30 Jan 2014 00:02:28 +0000
From: dotnetdub <dotnetdub at gmail.com>
Subject: [OpenSIPS-Users] mediaproxy
To: OpenSIPS users mailling list <users at lists.opensips.org>
Message-ID:
	<CACzrF4BCijdDC4BxLsSL_DL_Qx0tg_YX2Q2n9ucvaWuOa_d9Vw at mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1

Hi AG Team,

I have a very kind of rare issue.

We have one particular customer who is using an obscure UA that gets one way
audio when we have mediaproxy in the path.

We are running opensips 1.8.2 --> mediaproxy-dispatcher latest -->
mediaproxy-relay latest --> freeswitch

I have increased debug levels on the relay and dispatcher to maximum but
can't see anything pertinent in the logs.

If I remove the engage media proxy directive from opensips and let the audio
flow from freeswitch to the troublesome ua there is never an issue. Once
media proxy is in the media path there is no audio from the troublesome Ua
50% of the time. The SDP is identical if the audio flows or does not.

I have the logs at debug on dispatcher and relay right now but they don't
seem to reveal anything that points in the direction of where the issue is.
I'm sure its the remote UA that is the problem.

What information can I provide that we may debug this ?

Thanks
Shena

------------------------------

Message: 5
Date: Wed, 29 Jan 2014 20:40:58 -0800 (PST)
From: Alectronic <a.dorantwyford at ivstel.com>
Subject: [OpenSIPS-Users] Load_credentials => output to xlog();
To: users at lists.opensips.org
Message-ID: <1391056858246-7589370.post at n2.nabble.com>
Content-Type: text/plain; charset=us-ascii

Hi,

I'm trying to do authentication on my version of opensips(1.10) and want to
be able to allow it to view the username, password and domain then match it
with the database. however I can't seem to view the output of the database
from a xlog("");

I'm trying to to do it using 

modparam("auth_db", "load_credentials", "username;password;domain")

and then output it using

xlog("$avp(username)");

where am i going wrong? 

Thanks
Alec

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View this message in context:
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------------------------------

Message: 6
Date: Thu, 30 Jan 2014 11:13:33 +0200
From: R?zvan Crainea <razvan at opensips.org>
Subject: Re: [OpenSIPS-Users] Store $DLG_dir in acc Database
To: users at lists.opensips.org
Message-ID: <52EA17BD.3050807 at opensips.org>
Content-Type: text/plain; charset=UTF-8; format=flowed

Hi, Wilmar!

Have you tried setting the $DLG_dir pvar in the db_extra? For example:

modparam("acc", "db_extra", "direction=$DLG_dir")

Don't forget you have to add the direction column to your database table,
both acc and missed_calls.

Best regards,

Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com

On 01/29/2014 09:15 PM, Wilmar Campos wrote:
> Hi All,
>
> Can someone please point me to the right direction on how to store the 
> $DLG_dir into the accounting cdr?
>
> Thanks,
>
> Wilmar
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>

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