Re: [OpenSIPS-Users] Route timeout

[Pasan Meemaduma]

Hi Wesley,

Hope this is what you seeking.

modparam("tm", "fr_inv_timer_avp", "$avp(s:timeout)")

and you can load different timeout to avp using like below, this is per username

avp_db_load("$ru/username","$avp(s:timeout)");


and you can trigger failure route as follow,

 t_on_failure("1");

you can use this in your route to trigger next failure route.


failure_route[1]
{
#alternate route
}




From: Wesley Volcov 
To: OpenSIPS users mailling list 
Sent: Friday, August 13, 2010 1:17:43
Subject: [OpenSIPS-Users] Route timeout

Hello All,

Is there any way to configure a timeout per route?
What I mean is: If my first route doesn't send a reply(100 Trying or any other) 
in some seconds, the opensips sends the the call to failure route, and then I 
use failure route to try another route. How can I do that?

Regards,

-- 
Wesley Volcov
Email: wesleyvol...@gmail.com
Messenger: vol...@live.com
Mobile: +55 11 9989-5348
Website: http://volcov.blogspot.com


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Re: [OpenSIPS-Users] Route timeout

[Wesley Volcov]

Hey Antonio,

It seems to work great in my test environment!
Thank you very much!

Cheers

On 12 August 2010 17:09, Antonio Anderson Souza <
anto...@voicetechnology.com.br> wrote:

> Wesley,
>
> Yes, it's possible to control the timeout per route, this could be made by
> the fr_timer_avp and fr_inv_timer_avp, you need just set a valeu in seconds
> in those avps to control the timeout per branch.
>
> Have a look in the documentation of TM module [1].
>
> [1] -
> http://www.opensips.org/html/docs/modules/1.6.2/tm.html#timer-based-failover
>
> Best Regards,
>
> Antonio Anderson Souza
>
> Call me for free! 
> 
> Voice Technology  - Blog 
>  - Twitter  - 
> LinkedIn  - Facebook 
> 
>
>
>
>
>
> On Thu, Aug 12, 2010 at 4:47 PM, Wesley Volcov wrote:
>
>> Hello All,
>>
>> Is there any way to configure a timeout per route?
>> What I mean is: If my first route doesn't send a reply(100 Trying or any
>> other) in some seconds, the opensips sends the the call to failure route,
>> and then I use failure route to try another route. How can I do that?
>>
>> Regards,
>>
>> --
>> Wesley Volcov
>> Email: wesleyvol...@gmail.com
>> Messenger: vol...@live.com
>> Mobile: +55 11 9989-5348
>> Website: http://volcov.blogspot.com
>>
>> ___
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>>
>>
>


-- 
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Email: wesleyvol...@gmail.com
Messenger: vol...@live.com
Mobile: +55 11 9989-5348
Website: http://volcov.blogspot.com
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Re: [OpenSIPS-Users] Route timeout

[Antonio Anderson Souza]

Wesley,

Yes, it's possible to control the timeout per route, this could be made by
the fr_timer_avp and fr_inv_timer_avp, you need just set a valeu in seconds
in those avps to control the timeout per branch.

Have a look in the documentation of TM module [1].

[1] -
http://www.opensips.org/html/docs/modules/1.6.2/tm.html#timer-based-failover

Best Regards,

Antonio Anderson Souza

Call me for free!

Voice Technology  - Blog
 - Twitter 
- LinkedIn  - Facebook





On Thu, Aug 12, 2010 at 4:47 PM, Wesley Volcov wrote:

> Hello All,
>
> Is there any way to configure a timeout per route?
> What I mean is: If my first route doesn't send a reply(100 Trying or any
> other) in some seconds, the opensips sends the the call to failure route,
> and then I use failure route to try another route. How can I do that?
>
> Regards,
>
> --
> Wesley Volcov
> Email: wesleyvol...@gmail.com
> Messenger: vol...@live.com
> Mobile: +55 11 9989-5348
> Website: http://volcov.blogspot.com
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
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[OpenSIPS-Users] Route timeout

[Wesley Volcov]

Hello All,

Is there any way to configure a timeout per route?
What I mean is: If my first route doesn't send a reply(100 Trying or any
other) in some seconds, the opensips sends the the call to failure route,
and then I use failure route to try another route. How can I do that?

Regards,

-- 
Wesley Volcov
Email: wesleyvol...@gmail.com
Messenger: vol...@live.com
Mobile: +55 11 9989-5348
Website: http://volcov.blogspot.com
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Re: [OpenSIPS-Users] recvfrom does not return

[Bogdan-Andrei Iancu]

Hi Erik,

do you develop your own module? or where do you use the recvfrom ?

Regards,
Bogdan

erik.buel...@telenet.be wrote:
> Hello,
>
> In my setup recvfrom does not return with data while I can see the 
> data arriving on the specified IP:port with tcpdump.
>
> Any idea what could be verified?
>
> kind regards,
>
> Erik Buelens
> 
>
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-- 
Bogdan-Andrei Iancu
OpenSIPS Bootcamp
20 - 24 September 2010, Frankfurt, Germany
www.voice-system.ro


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Re: [OpenSIPS-Users] How to get call duration ( AcctSessionTime ) in OPENSIPS

[Bogdan-Andrei Iancu]

Brett Nemeroff wrote:
>
> On Tue, Aug 10, 2010 at 7:57 AM, Bogdan-Andrei Iancu 
> mailto:bog...@voice-system.ro>> wrote:
>
> Hi Alex,
>
> Of course, if you use the dialog support - anyhow, we have a ready
> patch
> for acc module to do accounting based on dialog support (directly
> CDRs).
>
> Regards,
> Bogdan
>
>
>  Bogdan,
> When can we see that? I've been anxiously hoping to see it. :) Can I 
> add custom avps? :) :)
Next days and yes, you can do extra_acc :)

Regards,
Bogdan

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Re: [OpenSIPS-Users] How to get call duration ( AcctSessionTime ) in OPENSIPS

[Bogdan-Andrei Iancu]

Matt,

I was looking at the code and the  $DLG_lifetime return the difference 
between the current time and the "start_time" (which the moment when the 
dialog was confirmed with 200 OK ).

Regards,
bogdan

Matt lehner wrote:
> I will have to look at this again. This was not the result I saw when
> playing with dialog for doing accounting. $DLG_liftime always showed a
> timeout value. I actually wrote my own module to do accounting using
> hooks from the dialog module.
>
> Matt
>
> On Tue, Aug 10, 2010 at 8:58 AM, Bogdan-Andrei Iancu
>  wrote:
>   
>> Hi Matt,
>>
>> not at all , $DLG_lifetime returns the time passed since the dialog was
>> established -> at BYE time, it will return the dialog duration.
>>
>> Regards,
>> Bogdan
>>
>>
>> Matt lehner wrote:
>> 
>>> $DLG_lifetime is actually the number of seconds before the dialog will
>>> timeout, or something along those lines. I also thought it might be
>>> the actual session duration.
>>>
>>> Matt
>>>
>>> On Tue, Aug 10, 2010 at 8:26 AM, Alex Massover  wrote:
>>>
>>>   
 Hi,

 $DLG_lifetime will not work?

 -Original Message-
 From: users-boun...@lists.opensips.org 
 [mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu
 Sent: Tuesday, August 10, 2010 1:17 PM
 To: OpenSIPS users mailling list
 Subject: Re: [OpenSIPS-Users] How to get call duration ( AcctSessionTime ) 
 in OPENSIPS

 Hi Tao,

 the accounting in opensips is transaction based and not call base -
 which means opensips will generate (for each call) a START and STOP acc
 events (when the call starts and when the call ends).

 There is no variable to automatically provide the call duration (as
 there is no call state) - each acc event has its own timestamp, so the
 RADIUS server can calculate the duration based on the INVITE and BYE
 timestamps.

 Regards,
 Bogdan

 Tao Vu Hoang wrote:

 
> Hi All!
>
>  I configure Opensips send Accounting to Steel Belt Radius 4, but i
> can't get call duration time ( AcctSessionTime ) , i don't know variable
> of opensips container call duration time to definition in file :
> opensips.cfg like this :
>  modparam("acc", "aaa_extra",   "User-Name=$Au; \
> Calling-Station-Id=$from; \
> Called-Station-Id=$to; \
> Sip-Translated-Request-URI=$ru; \
> Sip-RPid=$avp(s:rpid); \
> Source-IP=$avp(s:source_ip); \
> Source-Port=$avp(s:source_port); \
> 
> SIP-Proxy-IP=$avp(s:sip_proxy_ip); \
> Canonical-URI=$avp(s:can_uri); \
>
> Billing-Party=$avp(s:billing_party); \
>
> Divert-Reason=$avp(s:divert_reason); \
> User-Agent=$hdr(user-agent); \
> Contact=$hdr(contact); \
> Event=$hdr(event); \
> ENUM-TLD=$avp(s:enum_tld)")
>
> I use Opensips 1.6.3 to use function :
> get_dialog_info("start_time","$avp(s:starttime)","callid","$ci");
> to get Startime&  Stoptime of one Call from table DIALOG in database
> OPENIPS  but don't success, maybe i don't know how to use it.
>  I also use module : EXEC to get  Starttime&  Stoptime  from
> external cammand but don't success.
> Can someone help me ?  Or recommend me how to get  call duaration
> (AcctSessionTime).
> Thanks a lot.
>
>
>
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>
>
>   
 --
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 OpenSIPS Bootcamp
 20 - 24 September 2010, Frankfurt, Germany
 www.voice-system.ro


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>>> ___
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>>>
>>>   
>> --
>> Bogdan-Andrei Iancu
>> OpenSIPS Bootcamp
>> 20 - 24 September 2010, Frankfurt, Germany
>> www.voice-sys

Re: [OpenSIPS-Users] Hosting multiple Companies PBX on one OpenSIPS Server

[Deon Vermeulen]

Hi Mark

Thanks for the feedback.

Appreciate.

Will investigate into the mentioned systems.

Kind Regards

Deon

On Aug 12, 2010, at 2:46 PM, Mark Sayer wrote:

> The simple answer is yes, but not with OpenSIPS alone. You'll need
> something like Asterisk or FreeSWITCH to handle the PBX functions.
> 
> Mark
> 
> On Thu, Aug 12, 2010 at 8:48 PM, Deon Vermeulen
>  wrote:
>> Good Day List
>> Hope you well.
>> I would like to find out if someone could assist me.
>> I am in the process of setting up a SIP Server hosting a few companies PBX
>> Functionality, but require some assistance/guidance.
>> What I am looking for is:
>> 1) Support for Multiple Domains - Each Company with their own domain and
>> Extensions (Users).
>> 2) Support for Multiple Servers - LDAP integration with each Companies AD
>> Server. This should also be in a Secure way that each Company can only view,
>> search within their own Domain. Domains should be transparent to each other.
>> 3) Support for multiple PSTN/LCR connections with each company having their
>> own auto attendant console, Dial-Plans, etc.
>> Can this be done with OpenSIPS?
>> Thanks
>> 
>> Deon
>> 
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>> 
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Re: [OpenSIPS-Users] Hosting multiple Companies PBX on one OpenSIPS Server

[Mark Sayer]

The simple answer is yes, but not with OpenSIPS alone. You'll need
something like Asterisk or FreeSWITCH to handle the PBX functions.

Mark

On Thu, Aug 12, 2010 at 8:48 PM, Deon Vermeulen
 wrote:
> Good Day List
> Hope you well.
> I would like to find out if someone could assist me.
> I am in the process of setting up a SIP Server hosting a few companies PBX
> Functionality, but require some assistance/guidance.
> What I am looking for is:
> 1) Support for Multiple Domains - Each Company with their own domain and
> Extensions (Users).
> 2) Support for Multiple Servers - LDAP integration with each Companies AD
> Server. This should also be in a Secure way that each Company can only view,
> search within their own Domain. Domains should be transparent to each other.
> 3) Support for multiple PSTN/LCR connections with each company having their
> own auto attendant console, Dial-Plans, etc.
> Can this be done with OpenSIPS?
> Thanks
>
> Deon
>
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>

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[OpenSIPS-Users] Hosting multiple Companies PBX on one OpenSIPS Server

[Deon Vermeulen]

Good Day List

Hope you well.

I would like to find out if someone could assist me.

I am in the process of setting up a SIP Server hosting a few companies PBX 
Functionality, but require some assistance/guidance.
What I am looking for is:
1) Support for Multiple Domains - Each Company with their own domain and 
Extensions (Users).
2) Support for Multiple Servers - LDAP integration with each Companies AD 
Server. This should also be in a Secure way that each Company can only view, 
search within their own Domain. Domains should be transparent to each other.
3) Support for multiple PSTN/LCR connections with each company having their own 
auto attendant console, Dial-Plans, etc.

Can this be done with OpenSIPS?

Thanks

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