Re: [OpenSIPS-Users] Opensips-mi-proxy error with xcap-diff

[Saúl Ibarra Corretgé]

On 7/1/11 12:01 AM, Duane Larson wrote:

I just installed the SVN version 7639 and I am still getting the
HTTP/1.1 500 Internal Server Error



I checked the source and looks like the xcap-diff event is not defined, 
so it's not recognized. I'll make a patch for this.



Thanks for the report!

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AG Projects

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[OpenSIPS-Users] CRITICAL syslog mesages

[Jayesh Nambiar]

Hi Bogdan,
Thanks for the reply. I use the fr_timer as 3 seconds in my script and I am
also using failure routes extensively to failover my calls to multiple
destinations. Can this be a cause for the race event?
Is there anything else that I can look at to avoid this race event?

Thanks for all your help !!

--- Jayesh



> Hi Jayesh,
>
> the number refers to a timer list (type):
>   0  FR_TIMER_LIST
>   1  FR_INV_TIMER_LIST
>   2  WT_TIMER_LIST,
>   3  DELETE_LIST,
>   4  RT_T1_TO_1,
>   5  RT_T1_TO_2,
>   6  RT_T1_TO_3,
>   7  RT_T2,
>
> 4 - is first retransmission timer ,  while 0,1 are final response timers.
>
> The message meas that transaction module tried (in one process) to arm
> again a timer which was just reset (by other process) - it is a race
> between 2 events - re-arming and reseting the timer, Ex: re-arming
> retransmission timer while a reply came and stop retransmissions.
>
> Regards,
> Bogdan
>
> Jayesh Nambiar wrote:
> > Hi All,
> > I see a lot of similar messages in my syslog:
> > CRITICAL:tm:set_timer: set_timer for 4 list called on a "detached"
> > timer -- ignoring: 0x2d32f650.
> >
> > Although i don't see any problems in my call processing and I
> > understand I can safely ignore it, but can someone please make me
> > understand the significance of the integers used in these messages.
> > Like in the above message the integer is "4", i have earlier seen
> > these messages with integers 0 and 1 and at that time I used to have
> > serious problems of opensips not processing requests because of a lag
> > in DB queries !!
> >
> > Explanation of these integers and their significance will be very
> > helpful.
> > Thanks,
> >
> > --- Jayesh
> >
>
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Re: [OpenSIPS-Users] Users Digest, Vol 30, Issue 12

[Jayesh Nambiar]

Hi Bogdan,
Thanks for the reply. I use the fr_timer as 3 seconds in my script and I am
also using failure routes extensively to failover my calls to multiple
destinations. Can this be a cause for the race event?
Is there anything else that I can look at to avoid this race event?

Thanks for all your help !!

--- Jayesh



> Hi Jayesh,
>
> the number refers to a timer list (type):
>   0  FR_TIMER_LIST
>   1  FR_INV_TIMER_LIST
>   2  WT_TIMER_LIST,
>   3  DELETE_LIST,
>   4  RT_T1_TO_1,
>   5  RT_T1_TO_2,
>   6  RT_T1_TO_3,
>   7  RT_T2,
>
> 4 - is first retransmission timer ,  while 0,1 are final response timers.
>
> The message meas that transaction module tried (in one process) to arm
> again a timer which was just reset (by other process) - it is a race
> between 2 events - re-arming and reseting the timer, Ex: re-arming
> retransmission timer while a reply came and stop retransmissions.
>
> Regards,
> Bogdan
>
> Jayesh Nambiar wrote:
> > Hi All,
> > I see a lot of similar messages in my syslog:
> > CRITICAL:tm:set_timer: set_timer for 4 list called on a "detached"
> > timer -- ignoring: 0x2d32f650.
> >
> > Although i don't see any problems in my call processing and I
> > understand I can safely ignore it, but can someone please make me
> > understand the significance of the integers used in these messages.
> > Like in the above message the integer is "4", i have earlier seen
> > these messages with integers 0 and 1 and at that time I used to have
> > serious problems of opensips not processing requests because of a lag
> > in DB queries !!
> >
> > Explanation of these integers and their significance will be very
> > helpful.
> > Thanks,
> >
> > --- Jayesh
> >
>
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[OpenSIPS-Users] Prepay without RADIUS ?

[rad bogdan]

Hi everyone,
 
Is it possible to configure a prepay system that doesn't use the RADIUS server 
but only OpenSIPS, CallControl and CDRTool ?
Maybe I haven't searched enough but the documentations that I've read on 
internet rely on Freeradius being configured for 
authentication-accounting-authorization and don't give other alternatives.
Couldn't OpenSIPS be used to accomplish all these tasks ?
 
Thanks,
Bogdan


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Re: [OpenSIPS-Users] Opensips-mi-proxy error with xcap-diff

[Duane Larson]

I just installed the SVN version 7639 and I am still getting the
HTTP/1.1 500 Internal Server Error




On Tue, Jan 4, 2011 at 10:49 AM, Duane Larson wrote:

> I am running OpenSIPS 1.6.4.  Should I install the latest SVN version or
> was the fix included in 1.6.4?
>
>
>
> On Tue, Jan 4, 2011 at 1:59 AM, Saúl Ibarra Corretgé  > wrote:
>
>> Hi,
>>
>>
>> On 28/12/10 6:09 PM, Duane Larson wrote:
>>
>>> I am getting the following error from Opensips-mi-proxy when it receives
>>> an XMLRPC request from OpenXCAP
>>>
>>>
>> That error is produced when parsing the reply from OpenSIPS when executing
>> the pua_publish MI command to publish an xcap-diff event. Looks like
>> OpenSIPS is not identifying xcap-diff as a valid event even if you have
>> loaded the presence_xcapdiff module, I spotted these in you debug trace:
>>
>> ERROR:presence:handle_publish: Missing or unsupported event header field
>> value
>> ERROR:presence:handle_publish: #011event=[xcap-diff]
>>
>> IIRC Anca made a fix not so long ago addressing this. Please try latest
>> OpenSIPS version.
>>
>>
>> Regards,
>>
>> --
>> Saúl Ibarra Corretgé
>> AG Projects
>>
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>>
>
>
>
> --
> --
> *--*--*--*--*--*
> Duane
> *--*--*--*--*--*
> --
>



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Re: [OpenSIPS-Users] prepaid with call plan

[Dani Popa]

i know,

Thanks,
Dani

On Thu, Jan 6, 2011 at 7:21 PM, Bogdan-Andrei Iancu
wrote:

> yes, whatever backend (radius, db), you end up with 2 radius request at
> call setup. Mainly because they take place at different moments in call
> setup (so, you cannot combine them in a single request). The auth is done at
> INVITE time, before sending the call to termination, while the ACC start is
> done at 200 OK INVITE, when the call is established.
>
> BTW, the acc module in opensips can automatically do RADIUS accounting (you
> do not need to do it manually).
>
>
> Regards,
> Bogdan
>
> Dani Popa wrote:
>
>> Ok,
>> I think i know how can i do this setup, but this involve to make 2
>> radius request(one for authorize and one for accounting) for each first
>> invite and one for bye or cancel(for accounting).
>>
>>
>> Thanks,
>> Dani
>>
>> Bogdan-Andrei Iancu wrote:
>>
>>
>>> Hi Dani,
>>>
>>> never used the call_control module, but if need custom RADIUS
>>> interaction with RADIUS server, you can directly use the aaa_radius
>>> module to fire your custom RADIUS requests and to access the replies.
>>> See :
>>>
>>> http://www.opensips.org/html/docs/modules/1.6.x/aaa_radius.html#id249962
>>>
>>> http://www.opensips.org/html/docs/modules/1.6.x/aaa_radius.html#id249105
>>>
>>> Regards,
>>> bogdan
>>>
>>> Dani Popa wrote:
>>>
>>>
 Hi Bogdan,

 I know that opensips care just about SIP part. But my question is if
 somebody tried to make this setup with call_control and opensips. I did
 prepaid part, but now, i'm stuck in billing part. I have to tell radius
 that the call belongs to some service plans(i have to setup some flag
 and then pass it to radius somehow ) and the price for that call is 0. I
 just try to avoid to make changes in call_control  module to return
 (other value then  2 - call with no limit, 1 - call with call control
 limit, -1, and so on..) that mean in my case, call belongs to some
 service plan and to setup flag for billing in that case.


 Thanks,
 Dani

 Bogdan-Andrei Iancu wrote:


> Hi Dani,
>
> OpenSIPS is a SIP server so it does not care at all about rating
> plans, billing profiles, etc...It is doing only the SIP part.
>
> So, from PrePaid perspective, opensips is a SIP call controller, which
> keeps that state of the call and it is able to terminate (from middle)
> an ongoing call when instructed (from outside).
>
> Typically you integrate opensips with a billing/rating engine. Most
> used approaches are:
>   - when call is established, opensips queries the billing engine
> (DB, RADIUS, custom ) to see what's the maximum duration for that call
> ; so opensips will terminate the call if this max duration is
> exceeded. (you can use the dialog module)
>   - opensips informs the billing when a new call is established
> (again, DB, RADIUS, etc) and allows the billing to trigger the call
> termination from outside (like billing is keep computing costs and
> when there is no more credit, it notifies opensips to terminate the
> call) - again, you can use here the dialog module with the dlg_end_dlg
> command via XMLRPC .
>
> Regards,
> Bogdan
>
> Dani Popa wrote:
>
>
>> Hi,
>>
>> I wonder, how can be implemented with opensips prepaid system with
>> service plans with included minutes. Can someone to give me some
>> hints ?
>>
>> Thanks,
>> Dani Popa
>>
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://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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>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>>
>
>
> --
> Bogdan-Andrei Iancu
> OpenSIPS Event - expo, conf, social, bootcamp
> 2 - 4 February 2011, ITExpo, Miami,  USA
> www.voice-system.ro
>
>
> ___
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> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>



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[OpenSIPS-Users] Problem with load balancer module

[Diego Barberio]

Diego Sebastián Barberio

www.redmondsoftware.com
+54 11 48153511 (Ext 143)


-Original Message-
From: Diego Barberio [mailto:diego.barbe...@redmondsoftware.com] 
Sent: jueves, 06 de enero de 2011 03:50 p.m.
To: 'users@lists.opensips.org'
Subject: Problem with load balancer module

Hello,

I’m testing the Load Balancing module and I have a problem I can’t fix by
myself. The INVITE message is routed correctly to one of the destinations.
However the subsequent ACK and the BYE messages are not sent to the
destinations.

I set up opensips to run only in the udp 5060 port. Then I have two
identical applications: one running on port 5061 and the other on port 5062,
the 3 components are running in the same server which has a single IP
address: 192.168.1.195.
The application is very simple:
1. Receives the INVITE, starts streaming the RTP, and sends the OK
2. When the ACK is received injects some music in the streaming
3. Waits until de BYE is received. Then stops the streaming and
sends the OK.

This is the configuration of the load_balancer table:

mysql> select * from load_balancer;
++--++---++-
+
| id | group_id | dst_uri| resources | probe_mode |
description |
++--++---++-
+
|  1 |0 | sip:192.168.1.195:5061 | pstn=1|  0 |
|
|  2 |0 | sip:192.168.1.195:5062 | pstn=1|  0 |
|
++--++---++-
+
I've configured only one resource in each application because I'm just
testing.

Finally, this is the configuration script, which is the one from the
tutorial on the website:

debug=3
log_facility=LOG_LOCAL6

fork=yes
children=4

/* uncomment the following lines to enable debugging */
debug=6
fork=no
#log_stderror=yes

/* uncomment the next line to disable TCP (default on) */
disable_tcp=yes

port=5060

/* uncomment and configure the following line if you want opensips to 
   bind on a specific interface/port/proto (default bind on all available)
*/
listen=udp:192.168.1.195:5060

### Modules Section 

#set module path
mpath="/usr/local/lib/opensips/modules/"

loadmodule "maxfwd.so"
loadmodule "sl.so"
loadmodule "db_mysql.so"
loadmodule "tm.so"
loadmodule "uri.so"
loadmodule "rr.so"
loadmodule "dialog.so"
loadmodule "mi_fifo.so"
loadmodule "signaling.so"
loadmodule "textops.so"
loadmodule "load_balancer.so"

# - setting module-specific parameters ---
# - mi_fifo params -
modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo")


# - rr params -
# add value to ;lr param to cope with most of the UAs
#modparam("rr", "enable_full_lr", 1)
# do not append from tag to the RR (no need for this script)
#modparam("rr", "append_fromtag", 0)
modparam("rr","enable_double_rr",1)
modparam("rr","append_fromtag",1)


# - uri params -
modparam("uri", "use_uri_table", 0)

modparam("dialog", "dlg_flag", 13)
modparam("dialog", "db_mode", 1)
modparam("dialog", "db_url", "mysql://root:viamonte1...@localhost/opensips")


modparam("load_balancer",
"db_url","mysql://root:viamonte1...@localhost/opensips")

### Routing Logic 


# main request routing logic
route{
if (!mf_process_maxfwd_header("3")) {
sl_send_reply("483","looping");
exit;
}


if (!has_totag()) {
xlog("[Redmond] Hast'n to tag\n");
# initial request
record_route();
} else {
# sequential request -> obey Route indication
xlog("[Redmond] Has to tag\n");
loose_route();
t_relay();
exit;
}

# handle cancel and re-transmissions
if ( is_method("CANCEL") ) {
if ( t_check_trans() )
t_relay();
exit;
}


# from now on we have only the initial requests
if (!is_method("INVITE")) {
xlog("[Redmond] Not invite\n");
if ( t_check_trans() )
t_relay();
exit;
#send_reply("405","Method Not Allowed");
#exit;
}

load_balance("0","pstn");

# LB function returns negative if no suitable destination (for
requested resources) is found,
# or if all destinations are full
if ($retcode<0) {
xlog("[Redmond] Service full\n");
sl_send_reply("500","Service full");
exit;
}

xlog("[Redmond] Selected destination is: $du\n");

# send it out
if (!t_relay()) {
sl_reply_error();
}
}





It seems that the route is not saved, because the To tag is sent in the ACK
and BYE messages, also the logs I've added are written as expected for each
mes

Re: [OpenSIPS-Users] rtp and fake hangup

[mancyb...@gmail.com]

Very interesting, thank you very much Bogdan and Dani :)



On Thu, 06 Jan 2011 19:24:54 +0200
Bogdan-Andrei Iancu  wrote:

> Hi Mike,
> 
> it is not the only one. You can detect such ghost calls also at 
> signalling level (via SST).
> 
> But detection at media level:
> - is the most accurate as time error
> - it adds the biggest load on the system (as you need to put all RTP 
> streams on your servers).
> 
> So, which one to use (signalling versus media), is about making a 
> compromise between accuracy and load (considering that such ghost calls 
> are corner cases).
> 
> Regards,
> Bogdan
> 
> mancyb...@gmail.com wrote:
> > Hi All,
> >
> > in a scenario where opensips routes calls from sip user agents to voip 
> > carriers:
> > can you please confirm that the only way to be sure to prevent fraud false 
> > hangups
> > is to force the voice (rtp) to pass through opensips ?
> >
> >
> > Thanks and regards,
> > Mike
> > ___
> > Users mailing list
> > Users@lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >
> >   
> 
> 
> -- 
> Bogdan-Andrei Iancu
> OpenSIPS Event - expo, conf, social, bootcamp
> 2 - 4 February 2011, ITExpo, Miami,  USA
> www.voice-system.ro
> 
> 
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Re: [OpenSIPS-Users] prepaid with call plan

[Adrian Georgescu]

You must also deal with concurrent sessions sharing the same balance and 
updating the remaining time for each session depending on the duration of 
ringing time between the INVITE and 200 OK/ACK. Combining these with set of 
free minutes per destination is a mathematical equation hard to solve. You will 
most probably not find a system that does all these by default,  you must 
customize whatever you find close to your needs writing code that deals with 
your particular requirements .

Adrian

On Jan 6, 2011, at 5:40 PM, Dani Popa wrote:

> Hi,
> 
> I'm afraid i can not use call_control, because after DebitBalance or
> MaxSessiontime,  I can not categorize the call as belonging to a call
> plan with included minutes or to deduct from customer balance(for prepaid)
> 
> Dani
> 
> Tijmen de Mes wrote:
>> Hi ,
>> 
>> You can use the call-control application for this. See
>> http://callcontrol.ag-projects.com.
>> 
>> Best Regards,
>> 
>> Tijmen de Mes
>> AG Projects
>> 
>> Op 1/6/11 12:21 PM, Dani Popa schreef:
>>> Hi,
>>> 
>>> I wonder, how can be implemented with opensips prepaid system with
>>> service plans with included minutes. Can someone to give me some hints ?
>>> 
>>> Thanks,
>>> Dani Popa
>>> 
>>> ___
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>> 
>> 
>> ___
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>> 
> 
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Re: [OpenSIPS-Users] new install and INVITES not being forwarded

[Bogdan-Andrei Iancu]

Hi Gareth,

hard to say without some logs - do you see errors in the opensips log ? 
if not, increase the debug level to 6 and post the logs corresponding 
the call.


Regards,
Bogdan

Gareth Blades wrote:

Setup:-
Server installed at a hosting facility with its owm public IP address.
A desk phone in the office with a couple of accounts configured 
registered to opensips through a NAT firewall.
The opensips server has been installed and configured using the guide 
in the Opensips PACT book together with database support (no 
multidomain).


The two lines on the phone are able to register fine and I can see 
them as being registered in the database and 'opensipsctrl ul show' 
displays what I would expect.


However when I call between the two lines and minitoring via ngrep I 
see the initial INVITE go out and then a '407 proxy authentication 
required' comes back and the phone sends the invite again with the 
authentication but opensips doesnt send any reply. The phone 
retransmits the packet a couple of times and displays call failed.


Any idea what may be wrong?


I know the call audio is unlikely to work but that will be the next 
step. I am just trying to get each step working as I go at the moment.



Thanks
Gareth

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2 - 4 February 2011, ITExpo, Miami,  USA
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Re: [OpenSIPS-Users] rtp and fake hangup

[Bogdan-Andrei Iancu]

Hi Mike,

it is not the only one. You can detect such ghost calls also at 
signalling level (via SST).


But detection at media level:
   - is the most accurate as time error
   - it adds the biggest load on the system (as you need to put all RTP 
streams on your servers).


So, which one to use (signalling versus media), is about making a 
compromise between accuracy and load (considering that such ghost calls 
are corner cases).


Regards,
Bogdan

mancyb...@gmail.com wrote:

Hi All,

in a scenario where opensips routes calls from sip user agents to voip carriers:
can you please confirm that the only way to be sure to prevent fraud false 
hangups
is to force the voice (rtp) to pass through opensips ?


Thanks and regards,
Mike
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--
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OpenSIPS Event - expo, conf, social, bootcamp
2 - 4 February 2011, ITExpo, Miami,  USA
www.voice-system.ro


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Re: [OpenSIPS-Users] prepaid with call plan

[Bogdan-Andrei Iancu]

yes, whatever backend (radius, db), you end up with 2 radius request at 
call setup. Mainly because they take place at different moments in call 
setup (so, you cannot combine them in a single request). The auth is 
done at INVITE time, before sending the call to termination, while the 
ACC start is done at 200 OK INVITE, when the call is established.


BTW, the acc module in opensips can automatically do RADIUS accounting 
(you do not need to do it manually).


Regards,
Bogdan

Dani Popa wrote:

Ok,
I think i know how can i do this setup, but this involve to make 2
radius request(one for authorize and one for accounting) for each first
invite and one for bye or cancel(for accounting).


Thanks,
Dani

Bogdan-Andrei Iancu wrote:
  

Hi Dani,

never used the call_control module, but if need custom RADIUS
interaction with RADIUS server, you can directly use the aaa_radius
module to fire your custom RADIUS requests and to access the replies.
See :
 
http://www.opensips.org/html/docs/modules/1.6.x/aaa_radius.html#id249962
 
http://www.opensips.org/html/docs/modules/1.6.x/aaa_radius.html#id249105


Regards,
bogdan

Dani Popa wrote:


Hi Bogdan,

I know that opensips care just about SIP part. But my question is if
somebody tried to make this setup with call_control and opensips. I did
prepaid part, but now, i'm stuck in billing part. I have to tell radius
that the call belongs to some service plans(i have to setup some flag
and then pass it to radius somehow ) and the price for that call is 0. I
just try to avoid to make changes in call_control  module to return
(other value then  2 - call with no limit, 1 - call with call control
limit, -1, and so on..) that mean in my case, call belongs to some
service plan and to setup flag for billing in that case.


Thanks,
Dani

Bogdan-Andrei Iancu wrote:
 
  

Hi Dani,

OpenSIPS is a SIP server so it does not care at all about rating
plans, billing profiles, etc...It is doing only the SIP part.

So, from PrePaid perspective, opensips is a SIP call controller, which
keeps that state of the call and it is able to terminate (from middle)
an ongoing call when instructed (from outside).

Typically you integrate opensips with a billing/rating engine. Most
used approaches are:
   - when call is established, opensips queries the billing engine
(DB, RADIUS, custom ) to see what's the maximum duration for that call
; so opensips will terminate the call if this max duration is
exceeded. (you can use the dialog module)
   - opensips informs the billing when a new call is established
(again, DB, RADIUS, etc) and allows the billing to trigger the call
termination from outside (like billing is keep computing costs and
when there is no more credit, it notifies opensips to terminate the
call) - again, you can use here the dialog module with the dlg_end_dlg
command via XMLRPC .

Regards,
Bogdan

Dani Popa wrote:
   


Hi,

I wonder, how can be implemented with opensips prepaid system with
service plans with included minutes. Can someone to give me some
hints ?

Thanks,
Dani Popa

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[OpenSIPS-Users] new install and INVITES not being forwarded

[Gareth Blades]

Setup:-
Server installed at a hosting facility with its owm public IP address.
A desk phone in the office with a couple of accounts configured 
registered to opensips through a NAT firewall.
The opensips server has been installed and configured using the guide in 
the Opensips PACT book together with database support (no multidomain).


The two lines on the phone are able to register fine and I can see them 
as being registered in the database and 'opensipsctrl ul show' displays 
what I would expect.


However when I call between the two lines and minitoring via ngrep I see 
the initial INVITE go out and then a '407 proxy authentication required' 
comes back and the phone sends the invite again with the authentication 
but opensips doesnt send any reply. The phone retransmits the packet a 
couple of times and displays call failed.


Any idea what may be wrong?


I know the call audio is unlikely to work but that will be the next 
step. I am just trying to get each step working as I go at the moment.



Thanks
Gareth

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Re: [OpenSIPS-Users] rtp and fake hangup

[Dani Popa]

hi,

you have to use mediaproxy or rtpproxy for media timeout.

Dani

mancyb...@gmail.com wrote:
> Hi All,
>
> in a scenario where opensips routes calls from sip user agents to voip 
> carriers:
> can you please confirm that the only way to be sure to prevent fraud false 
> hangups
> is to force the voice (rtp) to pass through opensips ?
>
>
> Thanks and regards,
> Mike
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Re: [OpenSIPS-Users] prepaid with call plan

[Dani Popa]

Hi,

I'm afraid i can not use call_control, because after DebitBalance or
MaxSessiontime,  I can not categorize the call as belonging to a call
plan with included minutes or to deduct from customer balance(for prepaid)

Dani

Tijmen de Mes wrote:
> Hi ,
>
> You can use the call-control application for this. See
> http://callcontrol.ag-projects.com.
>
> Best Regards,
>
> Tijmen de Mes
> AG Projects
>
> Op 1/6/11 12:21 PM, Dani Popa schreef:
>> Hi,
>>
>> I wonder, how can be implemented with opensips prepaid system with
>> service plans with included minutes. Can someone to give me some hints ?
>>
>> Thanks,
>> Dani Popa
>>
>> ___
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>
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[OpenSIPS-Users] rtp and fake hangup

[mancyb...@gmail.com]

Hi All,

in a scenario where opensips routes calls from sip user agents to voip carriers:
can you please confirm that the only way to be sure to prevent fraud false 
hangups
is to force the voice (rtp) to pass through opensips ?


Thanks and regards,
Mike
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Re: [OpenSIPS-Users] prepaid with call plan

[Dani Popa]

Ok,
I think i know how can i do this setup, but this involve to make 2
radius request(one for authorize and one for accounting) for each first
invite and one for bye or cancel(for accounting).


Thanks,
Dani

Bogdan-Andrei Iancu wrote:
> Hi Dani,
>
> never used the call_control module, but if need custom RADIUS
> interaction with RADIUS server, you can directly use the aaa_radius
> module to fire your custom RADIUS requests and to access the replies.
> See :
>  
> http://www.opensips.org/html/docs/modules/1.6.x/aaa_radius.html#id249962
>  
> http://www.opensips.org/html/docs/modules/1.6.x/aaa_radius.html#id249105
>
> Regards,
> bogdan
>
> Dani Popa wrote:
>> Hi Bogdan,
>>
>> I know that opensips care just about SIP part. But my question is if
>> somebody tried to make this setup with call_control and opensips. I did
>> prepaid part, but now, i'm stuck in billing part. I have to tell radius
>> that the call belongs to some service plans(i have to setup some flag
>> and then pass it to radius somehow ) and the price for that call is 0. I
>> just try to avoid to make changes in call_control  module to return
>> (other value then  2 - call with no limit, 1 - call with call control
>> limit, -1, and so on..) that mean in my case, call belongs to some
>> service plan and to setup flag for billing in that case.
>>
>>
>> Thanks,
>> Dani
>>
>> Bogdan-Andrei Iancu wrote:
>>  
>>> Hi Dani,
>>>
>>> OpenSIPS is a SIP server so it does not care at all about rating
>>> plans, billing profiles, etc...It is doing only the SIP part.
>>>
>>> So, from PrePaid perspective, opensips is a SIP call controller, which
>>> keeps that state of the call and it is able to terminate (from middle)
>>> an ongoing call when instructed (from outside).
>>>
>>> Typically you integrate opensips with a billing/rating engine. Most
>>> used approaches are:
>>>- when call is established, opensips queries the billing engine
>>> (DB, RADIUS, custom ) to see what's the maximum duration for that call
>>> ; so opensips will terminate the call if this max duration is
>>> exceeded. (you can use the dialog module)
>>>- opensips informs the billing when a new call is established
>>> (again, DB, RADIUS, etc) and allows the billing to trigger the call
>>> termination from outside (like billing is keep computing costs and
>>> when there is no more credit, it notifies opensips to terminate the
>>> call) - again, you can use here the dialog module with the dlg_end_dlg
>>> command via XMLRPC .
>>>
>>> Regards,
>>> Bogdan
>>>
>>> Dani Popa wrote:
>>>
 Hi,

 I wonder, how can be implemented with opensips prepaid system with
 service plans with included minutes. Can someone to give me some
 hints ?

 Thanks,
 Dani Popa

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 Users@lists.opensips.org
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>>> 
>>
>> ___
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>
>

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Re: [OpenSIPS-Users] prepaid with call plan

[Tijmen de Mes]

Hi ,

You can use the call-control application for this. See 
http://callcontrol.ag-projects.com.


Best Regards,

Tijmen de Mes
AG Projects

Op 1/6/11 12:21 PM, Dani Popa schreef:

Hi,

I wonder, how can be implemented with opensips prepaid system with
service plans with included minutes. Can someone to give me some hints ?

Thanks,
Dani Popa

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Re: [OpenSIPS-Users] prepaid with call plan

[Bogdan-Andrei Iancu]

Hi Dani,

never used the call_control module, but if need custom RADIUS 
interaction with RADIUS server, you can directly use the aaa_radius 
module to fire your custom RADIUS requests and to access the replies. See :
  
http://www.opensips.org/html/docs/modules/1.6.x/aaa_radius.html#id249962
  
http://www.opensips.org/html/docs/modules/1.6.x/aaa_radius.html#id249105


Regards,
bogdan

Dani Popa wrote:

Hi Bogdan,

I know that opensips care just about SIP part. But my question is if
somebody tried to make this setup with call_control and opensips. I did
prepaid part, but now, i'm stuck in billing part. I have to tell radius
that the call belongs to some service plans(i have to setup some flag
and then pass it to radius somehow ) and the price for that call is 0. I
just try to avoid to make changes in call_control  module to return
(other value then  2 - call with no limit, 1 - call with call control
limit, -1, and so on..) that mean in my case, call belongs to some
service plan and to setup flag for billing in that case.


Thanks,
Dani

Bogdan-Andrei Iancu wrote:
  

Hi Dani,

OpenSIPS is a SIP server so it does not care at all about rating
plans, billing profiles, etc...It is doing only the SIP part.

So, from PrePaid perspective, opensips is a SIP call controller, which
keeps that state of the call and it is able to terminate (from middle)
an ongoing call when instructed (from outside).

Typically you integrate opensips with a billing/rating engine. Most
used approaches are:
   - when call is established, opensips queries the billing engine
(DB, RADIUS, custom ) to see what's the maximum duration for that call
; so opensips will terminate the call if this max duration is
exceeded. (you can use the dialog module)
   - opensips informs the billing when a new call is established
(again, DB, RADIUS, etc) and allows the billing to trigger the call
termination from outside (like billing is keep computing costs and
when there is no more credit, it notifies opensips to terminate the
call) - again, you can use here the dialog module with the dlg_end_dlg
command via XMLRPC .

Regards,
Bogdan

Dani Popa wrote:


Hi,

I wonder, how can be implemented with opensips prepaid system with
service plans with included minutes. Can someone to give me some hints ?

Thanks,
Dani Popa

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2 - 4 February 2011, ITExpo, Miami,  USA
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Re: [OpenSIPS-Users] opensips listen on port not 5060

[Bogdan-Andrei Iancu]

fengbin wrote:

Thank you for your quick reply.

But actually the request is sent to 5064 not 5060.
this has no relevance  - if you sent a request to a SIP server does not 
mean that the request targets that server (final destination) - it SIP 
server may act as a proxy if the request does not target it.

When opensips checks RURI is it possible to ignore port number?
that's up to your script logic...you can decide if a URI point to you or 
not based on whatever logic (condition is script). The default script 
detects a URI as local only if matches the listening interfaces (net 
level) - there is a  (uri==myself) test in script.


Regards,
Bogdan
Or maybe opensips can forward this registration to another instance of 
server with listening on 5060?


Again ,thank you  very much.

Best Regards,




On Thu, Jan 6, 2011 at 7:49 PM, Bogdan-Andrei Iancu 
mailto:bog...@voice-system.ro>> wrote:


Hi,

well, this happens because the REGISTER RURI points to
"sip:siptest.org " (or  "sip:siptest.org:5060
" ) which is not your opensips -> your
opensips is "sip:siptest.org:5064 "...

So your opensips says that the REGISTER is not for itself (by
looking at RURI) and simply does forward based on RURI (trying to
deliver the REGISTER to the server responsible for
"sip:siptest.org:5060 ".

You need to configure in the UAC to point to siptest.org:5064
 as REGISTRAR server.

Regards,
Bogdan


fengbin wrote:

Hi,all

I am using the default opensip.cfg and it's working.

However,when I set port=5064  and listen=udp.siptest.org
  I found
opensips forwarded REGISTER request to itself with port 5060.


Did I miss some configuration?

The enclosed is the pcap file.

Thanks!

Regards,

-- 
arithdon



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OpenSIPS Event - expo, conf, social, bootcamp
2 - 4 February 2011, ITExpo, Miami,  USA
www.voice-system.ro 


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Re: [OpenSIPS-Users] prepaid with call plan

[Dani Popa]

Hi Bogdan,

I know that opensips care just about SIP part. But my question is if
somebody tried to make this setup with call_control and opensips. I did
prepaid part, but now, i'm stuck in billing part. I have to tell radius
that the call belongs to some service plans(i have to setup some flag
and then pass it to radius somehow ) and the price for that call is 0. I
just try to avoid to make changes in call_control  module to return
(other value then  2 - call with no limit, 1 - call with call control
limit, -1, and so on..) that mean in my case, call belongs to some
service plan and to setup flag for billing in that case.


Thanks,
Dani

Bogdan-Andrei Iancu wrote:
> Hi Dani,
>
> OpenSIPS is a SIP server so it does not care at all about rating
> plans, billing profiles, etc...It is doing only the SIP part.
>
> So, from PrePaid perspective, opensips is a SIP call controller, which
> keeps that state of the call and it is able to terminate (from middle)
> an ongoing call when instructed (from outside).
>
> Typically you integrate opensips with a billing/rating engine. Most
> used approaches are:
>- when call is established, opensips queries the billing engine
> (DB, RADIUS, custom ) to see what's the maximum duration for that call
> ; so opensips will terminate the call if this max duration is
> exceeded. (you can use the dialog module)
>- opensips informs the billing when a new call is established
> (again, DB, RADIUS, etc) and allows the billing to trigger the call
> termination from outside (like billing is keep computing costs and
> when there is no more credit, it notifies opensips to terminate the
> call) - again, you can use here the dialog module with the dlg_end_dlg
> command via XMLRPC .
>
> Regards,
> Bogdan
>
> Dani Popa wrote:
>> Hi,
>>
>> I wonder, how can be implemented with opensips prepaid system with
>> service plans with included minutes. Can someone to give me some hints ?
>>
>> Thanks,
>> Dani Popa
>>
>> ___
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>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>   
>
>

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[OpenSIPS-Users] nat_traversal samples?

[Jeff Chua]

If anyone has working opensips.cfg using nat_traversal, please kindly
show me how to set one up correctly.

I'm looking for something like this ...

   SIP Client A -- OpenSIPS B -- VPN (only TCP/IP allowed) -- OpenSIPS
C -- SIP Client D


Thank you,
Jeff

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Re: [OpenSIPS-Users] opensips listen on port not 5060

[fengbin]

Thank you for your quick reply.

But actually the request is sent to 5064 not 5060. When opensips checks RURI
is it possible to ignore port number?
Or maybe opensips can forward this registration to another instance of
server with listening on 5060?

Again ,thank you  very much.

Best Regards,




On Thu, Jan 6, 2011 at 7:49 PM, Bogdan-Andrei Iancu
wrote:

> Hi,
>
> well, this happens because the REGISTER RURI points to "sip:siptest.org"
> (or  "sip:siptest.org:5060" ) which is not your opensips -> your opensips
> is "sip:siptest.org:5064"...
>
> So your opensips says that the REGISTER is not for itself (by looking at
> RURI) and simply does forward based on RURI (trying to deliver the REGISTER
> to the server responsible for "sip:siptest.org:5060".
>
> You need to configure in the UAC to point to siptest.org:5064 as REGISTRAR
> server.
>
> Regards,
> Bogdan
>
>
> fengbin wrote:
>
>> Hi,all
>>
>> I am using the default opensip.cfg and it's working.
>>
>> However,when I set port=5064  and listen=udp.siptest.org <
>> http://udp.siptest.org> I found opensips forwarded REGISTER request to
>> itself with port 5060.
>>
>>
>> Did I miss some configuration?
>>
>> The enclosed is the pcap file.
>>
>> Thanks!
>>
>> Regards,
>>
>> --
>> arithdon
>> 
>>
>> ___
>> Users mailing list
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>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>
>
> --
> Bogdan-Andrei Iancu
> OpenSIPS Event - expo, conf, social, bootcamp
> 2 - 4 February 2011, ITExpo, Miami,  USA
> www.voice-system.ro
>
>
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>



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Re: [OpenSIPS-Users] opensips listen on port not 5060

[Bogdan-Andrei Iancu]

Hi,

well, this happens because the REGISTER RURI points to "sip:siptest.org" 
(or  "sip:siptest.org:5060" ) which is not your opensips -> your 
opensips is "sip:siptest.org:5064"...


So your opensips says that the REGISTER is not for itself (by looking at 
RURI) and simply does forward based on RURI (trying to deliver the 
REGISTER to the server responsible for "sip:siptest.org:5060".


You need to configure in the UAC to point to siptest.org:5064 as 
REGISTRAR server.


Regards,
Bogdan


fengbin wrote:

Hi,all

I am using the default opensip.cfg and it's working.

However,when I set port=5064  and listen=udp.siptest.org 
 I found opensips forwarded REGISTER request 
to itself with port 5060.


Did I miss some configuration?

The enclosed is the pcap file.

Thanks!

Regards,

--
arithdon


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2 - 4 February 2011, ITExpo, Miami,  USA
www.voice-system.ro


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Re: [OpenSIPS-Users] prepaid with call plan

[Bogdan-Andrei Iancu]

Hi Dani,

OpenSIPS is a SIP server so it does not care at all about rating plans, 
billing profiles, etc...It is doing only the SIP part.


So, from PrePaid perspective, opensips is a SIP call controller, which 
keeps that state of the call and it is able to terminate (from middle) 
an ongoing call when instructed (from outside).


Typically you integrate opensips with a billing/rating engine. Most used 
approaches are:
   - when call is established, opensips queries the billing engine (DB, 
RADIUS, custom ) to see what's the maximum duration for that call ; so 
opensips will terminate the call if this max duration is exceeded. (you 
can use the dialog module)
   - opensips informs the billing when a new call is established 
(again, DB, RADIUS, etc) and allows the billing to trigger the call 
termination from outside (like billing is keep computing costs and when 
there is no more credit, it notifies opensips to terminate the call) - 
again, you can use here the dialog module with the dlg_end_dlg command 
via XMLRPC .


Regards,
Bogdan

Dani Popa wrote:

Hi,

I wonder, how can be implemented with opensips prepaid system with
service plans with included minutes. Can someone to give me some hints ?

Thanks,
Dani Popa

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2 - 4 February 2011, ITExpo, Miami,  USA
www.voice-system.ro


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Re: [OpenSIPS-Users] ACK incorrect handle when use domain

[Bogdan-Andrei Iancu]

First, as a note - opensips is doing what the config file tells to do.

Second, ACK is a sequential requests and RFC defines different routing 
algs for initial and sequential requests.


I suggest you to follow this webinar where the SIP routing is explained:
  http://www.opensips.org/Resources/Webinars#toc11

Regards,
Bogdan


fengbin wrote:

Hi,Bogdan

Thank you for your reply.

I have a question on your comment.

in RFC3261 16.12 (in the following)
The first step of proxy should lookup location table to replace 
request-uri if there is.
In my case I think proxy should first replace ACK's request uri to 
callee's sip identity and then go to loose_route


I added lookup script before loose_route then it works.

   1.  The proxy will inspect the Request-URI.  If it indicates a
  resource owned by this proxy, the proxy will replace it with
  the results of running a location service.  Otherwise, the
  proxy will not change the Request-URI.

  2.  The proxy will inspect the URI in the topmost Route header
  field value.  If it indicates this proxy, the proxy removes it
  from the Route header field (this route node has been
  reached).

  3.  The proxy will forward the request to the resource indicated
  by the URI in the topmost Route header field value or in the
  Request-URI if no Route header field is present.  The proxy
  determines the address, port and transport to use when
  forwarding the request by applying the procedures in [4] to
  that URI.
  





On Thu, Jan 6, 2011 at 7:27 PM, Bogdan-Andrei Iancu 
mailto:bog...@voice-system.ro>> wrote:


Hi,

The ACK gets to our opensips because of the Route hdr (Route:
); opensips will further route the ACK
based on RURI which is "sip:stct_1...@siptest.org
"...is this domain resolvable
by DNS and pointing to callee (10.61.20.200:5060
) ? Callee has to place in 200 OK its
contact address that can be used to route back to it...so if its
contact "sip:stct_1...@siptest.org
" does not point back to
callee, it is a callee issue.

Regards,
Bogdan

fengbin wrote:

Hi,Mogdan,

Thank you for your help.

The enclosed is the pcap file.


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OpenSIPS Event - expo, conf, social, bootcamp
2 - 4 February 2011, ITExpo, Miami,  USA
www.voice-system.ro 


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Re: [OpenSIPS-Users] CRITICAL syslog mesages

[Bogdan-Andrei Iancu]

Hi Jayesh,

the number refers to a timer list (type):
  0  FR_TIMER_LIST
  1  FR_INV_TIMER_LIST
  2  WT_TIMER_LIST,
  3  DELETE_LIST,
  4  RT_T1_TO_1,
  5  RT_T1_TO_2,
  6  RT_T1_TO_3,
  7  RT_T2,

4 - is first retransmission timer ,  while 0,1 are final response timers.

The message meas that transaction module tried (in one process) to arm 
again a timer which was just reset (by other process) - it is a race 
between 2 events - re-arming and reseting the timer, Ex: re-arming 
retransmission timer while a reply came and stop retransmissions.


Regards,
Bogdan

Jayesh Nambiar wrote:

Hi All,
I see a lot of similar messages in my syslog:
CRITICAL:tm:set_timer: set_timer for 4 list called on a "detached" 
timer -- ignoring: 0x2d32f650.


Although i don't see any problems in my call processing and I 
understand I can safely ignore it, but can someone please make me 
understand the significance of the integers used in these messages. 
Like in the above message the integer is "4", i have earlier seen 
these messages with integers 0 and 1 and at that time I used to have 
serious problems of opensips not processing requests because of a lag 
in DB queries !!


Explanation of these integers and their significance will be very 
helpful. 
Thanks,


--- Jayesh



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Bogdan-Andrei Iancu
OpenSIPS Event - expo, conf, social, bootcamp
2 - 4 February 2011, ITExpo, Miami,  USA
www.voice-system.ro


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Re: [OpenSIPS-Users] ACK incorrect handle when use domain

[fengbin]

Hi,Bogdan

Thank you for your reply.

I have a question on your comment.

in RFC3261 16.12 (in the following)
The first step of proxy should lookup location table to replace request-uri
if there is.
In my case I think proxy should first replace ACK's request uri to callee's
sip identity and then go to loose_route

I added lookup script before loose_route then it works.

   1.  The proxy will inspect the Request-URI.  If it indicates a
  resource owned by this proxy, the proxy will replace it with
  the results of running a location service.  Otherwise, the
  proxy will not change the Request-URI.

  2.  The proxy will inspect the URI in the topmost Route header
  field value.  If it indicates this proxy, the proxy removes it
  from the Route header field (this route node has been
  reached).

  3.  The proxy will forward the request to the resource indicated
  by the URI in the topmost Route header field value or in the
  Request-URI if no Route header field is present.  The proxy
  determines the address, port and transport to use when
  forwarding the request by applying the procedures in [4] to
  that URI.





On Thu, Jan 6, 2011 at 7:27 PM, Bogdan-Andrei Iancu
wrote:

> Hi,
>
> The ACK gets to our opensips because of the Route hdr (Route:
> ); opensips will further route the ACK based on RURI
> which is "sip:stct_1...@siptest.org "...is
> this domain resolvable by DNS and pointing to callee (10.61.20.200:5060) ?
> Callee has to place in 200 OK its contact address that can be used to route
> back to it...so if its contact 
> "sip:stct_1...@siptest.org"
> does not point back to callee, it is a callee issue.
>
> Regards,
> Bogdan
>
> fengbin wrote:
>
>> Hi,Mogdan,
>>
>> Thank you for your help.
>>
>> The enclosed is the pcap file.
>> 
>>
>> ___
>> Users mailing list
>> Users@lists.opensips.org
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>>
>>
>
>
> --
> Bogdan-Andrei Iancu
> OpenSIPS Event - expo, conf, social, bootcamp
> 2 - 4 February 2011, ITExpo, Miami,  USA
> www.voice-system.ro
>
>
> ___
> Users mailing list
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> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>



-- 
arithdon
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Re: [OpenSIPS-Users] How to implement a SIP Trunk in between two SIP servers.

[Bogdan-Andrei Iancu]

Hi Steven,

If you use the opensips default script, your opensips will accept calls 
from any other external SIP entities (call targeting a local opensips 
subscriber).


If you want to configure your opensips to accept foreign calls only form 
a specific IP address, you can use the permission module, with address 
table to implement IP-based authentication.


Best regards,
Bogdan

steven chew wrote:

Hi everyone,

I am a newbie with SIP-Trunk in OpenSips. 

I have a Cisco Communication Unified Manager and a OpenSips Server 
running in two different Virtual Machines.


I would like to have a SIP trunk in between them "Cisco Communication 
Unified Manager and OpenSips Server". 

Therefore, I can make a call from OpenSips Server's SIP Clients to 
Cisco IP Phone. 


What should I need to add into opensips.cfg configuration file?

I hope you can give some simple examples how to do it. 


I look forward to hearing from your advise asap.

Thanks
Regards,
-Steven.



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--
Bogdan-Andrei Iancu
OpenSIPS Event - expo, conf, social, bootcamp
2 - 4 February 2011, ITExpo, Miami,  USA
www.voice-system.ro


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[OpenSIPS-Users] opensips listen on port not 5060

[fengbin]

Hi,all

I am using the default opensip.cfg and it's working.

However,when I set port=5064  and listen=udp.siptest.org I found opensips
forwarded REGISTER request to itself with port 5060.

Did I miss some configuration?

The enclosed is the pcap file.

Thanks!

Regards,

-- 
arithdon


siptest_xport_reg_KO.pcap
Description: application/extension-pcap
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Re: [OpenSIPS-Users] ACK incorrect handle when use domain

[Bogdan-Andrei Iancu]

Hi,

The ACK gets to our opensips because of the Route hdr (Route: 
); opensips will further route the ACK based on 
RURI which is "sip:stct_1...@siptest.org"...is this domain resolvable by 
DNS and pointing to callee (10.61.20.200:5060) ? Callee has to place in 
200 OK its contact address that can be used to route back to it...so if 
its contact "sip:stct_1...@siptest.org" does not point back to callee, 
it is a callee issue.


Regards,
Bogdan

fengbin wrote:

Hi,Mogdan,

Thank you for your help.

The enclosed is the pcap file.


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--
Bogdan-Andrei Iancu
OpenSIPS Event - expo, conf, social, bootcamp
2 - 4 February 2011, ITExpo, Miami,  USA
www.voice-system.ro


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[OpenSIPS-Users] prepaid with call plan

[Dani Popa]

Hi,

I wonder, how can be implemented with opensips prepaid system with
service plans with included minutes. Can someone to give me some hints ?

Thanks,
Dani Popa

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[OpenSIPS-Users] CRITICAL syslog mesages

[Jayesh Nambiar]

Hi All,
I see a lot of similar messages in my syslog:
CRITICAL:tm:set_timer: set_timer for 4 list called on a "detached" timer --
ignoring: 0x2d32f650.

Although i don't see any problems in my call processing and I understand I
can safely ignore it, but can someone please make me understand the
significance of the integers used in these messages. Like in the above
message the integer is "4", i have earlier seen these messages with integers
0 and 1 and at that time I used to have serious problems of opensips not
processing requests because of a lag in DB queries !!

Explanation of these integers and their significance will be very helpful.
Thanks,

--- Jayesh
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