Hello,
On 06/13/08 05:29, Gang Liu (VSC) wrote:
> Hi,
> I have 3 trunk gateways, two are cisco 5300 with 4E1, the other is
> 16E1 audio codes.
> Could I use dispatch module to load balance those gateways based
> weight I set.
> I successfully setup dispatcher module to load 50%, 50% for two gate
False alarm. I cleaned out some debug logs and it works the same as 1.3.1.
It's strange though. The debugs were not in the exec or avpops modules.
On Thursday 12 June 2008, Robert Dyck wrote:
> With the following in cfg --
>
> exec_avp("uname -s","$avp(s:test)");
> xlog("L_NOTICE","test result is
With the following in cfg --
exec_avp("uname -s","$avp(s:test)");
xlog("L_NOTICE","test result is <$avp(s:test)>\n");
1.3.1 yields "test result is "
1.3.2 yields "test result is <>"
I also noticed that 1.3.2 handles an uninitialized $var differently than
1.3.1. with the following in cfg --
if
Hi,
I have 3 trunk gateways, two are cisco 5300 with 4E1, the other is 16E1
audio codes.
Could I use dispatch module to load balance those gateways based weight I
set.
I successfully setup dispatcher module to load 50%, 50% for two gateway.
regards,
Danile___
On Thu, Jun 12, 2008 at 09:48:17PM +0200, David Villasmil wrote:
> What exactly is 3000 flows?
> 1500 calls?
No, speaking on "media-proxy context" 1 flow = 1 call, because there
is no transcoding, only traffic flowing in and out
--
Saludos
Raúl Alexis Betancor Santana
Dimensión virtual S.L.
___
Hello everyone, i am testing the carrierroute module to do the following:
-If destination starts with 1760, the call is sent to PSTN GW.
-If destination starts with 1305, the call is blocked.
The lab works as desired, however i am getting this error:
ERROR:carrierroute:carrier_rewrite_msg: during
Now I have the following question:
to be able to make sure you get ALL accounting info with media-proxy, you
need to use it on ALL CALLS, is this correct?
cheers,
David
On Thu, Jun 12, 2008 at 9:48 PM, David Villasmil <
[EMAIL PROTECTED]> wrote:
> What exactly is 3000 flows?
> 1500 calls?
>
>
What exactly is 3000 flows?
1500 calls?
On Thu, Jun 12, 2008 at 9:29 PM, <[EMAIL PROTECTED]> wrote:
> On Thu, Jun 12, 2008 at 07:05:07PM +0200, David Villasmil wrote:
> > >
> > > > This all seems very disorganized. Co-operation is needed to build a
> > > > comprehensive solution. Perhaps a modula
On Thu, Jun 12, 2008 at 07:05:07PM +0200, David Villasmil wrote:
> >
> > > This all seems very disorganized. Co-operation is needed to build a
> > > comprehensive solution. Perhaps a modular session border controller where
> > > one module is Openser and the others are a B2BUA and a media proxy. T
David Villasmil wrote:
> Then...
>
> Excuse my asking: What IS the point of doing accounting with
> openser, if it will never be accurate?
Just because someone wrote an accounting module for OpenSER, doesn't
mean it will be accurate or complete.
I do all my accounting via my media gate
You can limit per contract. Check if the user belongs to a contract
and then perform contract profiling instead of caller/callee
profiling. That's the big thing about OpenSER: very flexible config.
Regards,
Ovidiu Sas
On Thu, Jun 12, 2008 at 1:01 PM, Patrick Miccio <[EMAIL PROTECTED]> wrote:
>
On Thursday 12 June 2008, Iñaki Baz Castillo wrote:
> El Thursday 12 June 2008 18:39:33 Robert Dyck escribió:
> > On Thursday 12 June 2008, Iñaki Baz Castillo wrote:
> > > El Thursday 12 June 2008 16:38:11 Juha Heinanen escribió:
> > > > Iñaki Baz Castillo writes:
> > > > > You can take a look at
>
> > This all seems very disorganized. Co-operation is needed to build a
> > comprehensive solution. Perhaps a modular session border controller where
> > one module is Openser and the others are a B2BUA and a media proxy. They
> > would be optimized to work well together, possibly using non-sip
Hi,
that sounds promising, I like the 2nd example, limiting calls to PSTN is the
idea.
The only problem is that OpenSER does only know about username/uuid, you cannot
control if a customer with 1 contract
uses 2 or more usernames. Then again we could limit calls based on the source
IP, but cus
I was hoping that Bodgan would chip on this conversation...
:S
On Thu, Jun 12, 2008 at 6:34 PM, Iñaki Baz Castillo <[EMAIL PROTECTED]>
wrote:
> El Thursday 12 June 2008 18:25:28 Jesus Rodriguez escribió:
>
> > This thread seems the "spanish army" accounting support :-)
>
> Doesn't exist a Ope
El Thursday 12 June 2008 18:39:33 Robert Dyck escribió:
> On Thursday 12 June 2008, Iñaki Baz Castillo wrote:
> > El Thursday 12 June 2008 16:38:11 Juha Heinanen escribió:
> > > Iñaki Baz Castillo writes:
> > > > You can take a look at those projects:
> > > >
> > > > http://www.b2bua.org/
> > >
On Thursday 12 June 2008, Iñaki Baz Castillo wrote:
> El Thursday 12 June 2008 16:38:11 Juha Heinanen escribió:
> > Iñaki Baz Castillo writes:
> > > You can take a look at those projects:
> > >
> > > http://www.b2bua.org/
> > > http://www.resiprocate.org/Main_Page
> > > https://sailfin.dev.jav
El Thursday 12 June 2008 18:25:28 Jesus Rodriguez escribió:
> This thread seems the "spanish army" accounting support :-)
Doesn't exist a OpenSer spanish maillist? XD
What are we doing here speaking between us in English? XD
--
Iñaki Baz Castillo
[EMAIL PROTECTED]
Hola,
> El Thursday 12 June 2008 18:15:07 Raúl Alexis Betancor Santana
> escribió:
>> Why you don't want to use RTPProxies ?, I insists that if you do not
>> control de RTP path .. you could not do accurate accounting only
>> with the
>> signaling path, not without having a lot of complains f
El Thursday 12 June 2008 18:15:07 Raúl Alexis Betancor Santana escribió:
> Why you don't want to use RTPProxies ?, I insists that if you do not
> control de RTP path .. you could not do accurate accounting only with the
> signaling path, not without having a lot of complains from your customers
> .
El Jueves, 12 de Junio de 2008 17:03, David Villasmil escribió:
> In which case, we STILL depend on a third party NOT BEING openser...
That is because with ONLY a proxy, you could not do accurate accounting, thats
all.
> Just a question:
>
> Is there any plan to implement this? i.e. sending an O
El Thursday 12 June 2008 18:03:26 David Villasmil escribió:
> Just a question:
>
> Is there any plan to implement this? i.e. sending an OPTIONS to the UACs to
> make sure they are online whilst in a call and ending the dialog if there
> is not answer?
IMHO OpenSer is a very good SIP stateful prox
Hi Patrick,
Take a look at the dialog module and dialog profiling support - I think
it will help you:
http://lists.openser.org/pipermail/users/2008-June/017710.html
There is even an example similar to what you want to do.
Regards,
Bogdan
Patrick Miccio wrote:
> Hi @ all,
>
> I was wonderin
On Thu, Jun 12, 2008 at 5:51 PM, Iñaki Baz Castillo <[EMAIL PROTECTED]>
wrote:
> El Thursday 12 June 2008 17:38:35 David Villasmil escribió:
> > Then...
> >
> > Excuse my asking: What IS the point of doing accounting with
> openser,
> > if it will never be accurate?
>
> That's not true. If yo
On Thu, Jun 12, 2008 at 5:51 PM, Iñaki Baz Castillo <[EMAIL PROTECTED]>
wrote:
> El Thursday 12 June 2008 17:38:35 David Villasmil escribió:
> > Then...
> >
> > Excuse my asking: What IS the point of doing accounting with
> openser,
> > if it will never be accurate?
>
> That's not true. If yo
Hi @ all,
I was wondering if anyone of you guys already implemented a system where one
could limit the maximum number of
concurrent calls? With ISDN-terminaladapters or analog-telefon-adapters it is
no problem because the hardware itself is
the limiting device, but with the growing amount of IP
On Thursday 12 June 2008, David Villasmil wrote:
> Then...
>
> Excuse my asking: What IS the point of doing accounting with openser,
> if it will never be accurate?
Hi,
i would be really glad if you could sell me a 100% accurate accounting
solution. ;-) In real world scenarios there will be
Hola,
> El Jueves, 12 de Junio de 2008 16:38, David Villasmil escribió:
>> Then...
>>
>> Excuse my asking: What IS the point of doing accounting with
>> openser,
>> if it will never be accurate?
>>
>> david
>
> Hi David, you should not rely only on OpenSer to do accounting, if
> you are
>
El Thursday 12 June 2008 17:38:35 David Villasmil escribió:
> Then...
>
> Excuse my asking: What IS the point of doing accounting with openser,
> if it will never be accurate?
That's not true. If you use OpenSer + Radius ACC + MediaProxy + CDRTool
(optional) you can get 99% accurate: if a UA
El Thursday 12 June 2008 16:38:11 Juha Heinanen escribió:
> Iñaki Baz Castillo writes:
> > You can take a look at those projects:
> >
> > http://www.b2bua.org/
> > http://www.resiprocate.org/Main_Page
> > https://sailfin.dev.java.net/
> > http://www.opensipstack.org/sbc_man_content.html
>
> a
Hola David,
> Then...
>
> Excuse my asking: What IS the point of doing accounting with
> openser, if it will never be accurate?
The implementation of different "keepalive" methods like Session
Timers, OPTIONS in-dialog, stop receiving RTP (some gateways like
Cisco AS5xxx have an opti
El Jueves, 12 de Junio de 2008 16:38, David Villasmil escribió:
> Then...
>
> Excuse my asking: What IS the point of doing accounting with openser,
> if it will never be accurate?
>
> david
Hi David, you should not rely only on OpenSer to do accounting, if you are
doing so .. you are in trou
Then...
Excuse my asking: What IS the point of doing accounting with openser,
if it will never be accurate?
david
On Thu, Jun 12, 2008 at 5:15 PM, Jesus Rodriguez <[EMAIL PROTECTED]> wrote:
> Hola David,
>
>
> Now I got a question: I understand that sometimes, when a UAC droppes
>> ou
Hi Kionez,
you do not need to integrate the patch anymore - same functionality is
on the SVN trunk of openser.
Regards,
Bogdan
kionez wrote:
> #include// created 12/06/2008 14:51
>
>
>> so, isn't a configuration issue? :)
>>
>
> ok.. auto-answer.. I suppose is a configuration mistak
Hola David,
> Now I got a question: I understand that sometimes, when a UAC
> droppes out of the internet, a BYE message will be lost. But, if I
> were to use Openser as a Carrier switch, this is NOT supposed to
> happen, right? as all VoIP Carriers MUST be on the internet
> permanen
Hello all,
Now I got a question: I understand that sometimes, when a UAC droppes
out of the internet, a BYE message will be lost. But, if I were to use
Openser as a Carrier switch, this is NOT supposed to happen, right? as all
VoIP Carriers MUST be on the internet permanently.
Any thoughts?
Iñaki Baz Castillo writes:
> You can take a look at those projects:
>
> http://www.b2bua.org/
> http://www.resiprocate.org/Main_Page
> https://sailfin.dev.java.net/
> http://www.opensipstack.org/sbc_man_content.html
also sems supports session-timer.
-- juha
__
El Thursday 12 June 2008 14:38:44 David Villasmil escribió:
> > The problem is that you are addressing the problem in a privative way
> > while there are RFC's and techniques for that. For your proposal using
> > Session Timers should be the best option,
>
> Are you talking, i.e. Asterisk?
No, As
#include// created 12/06/2008 14:51
> so, isn't a configuration issue? :)
ok.. auto-answer.. I suppose is a configuration mistake...
if I set avp before record_route, everything works fine..
-8<---
if(!is_method("REGISTER"))
{
$avp
#include// created 12/06/2008 10:55
> i just commited this additional log messages yesterday to the trunk version,
> rev 4373. I thought this could be useful in debugging this problem. :-)
I added LOG function to dialog module in 1.3.2-tls (editing
dlg_handlers.c) and now i have this debug i
On Thu, Jun 12, 2008 at 2:22 PM, Iñaki Baz Castillo <[EMAIL PROTECTED]>
wrote:
> El Thursday 12 June 2008 14:06:22 David Villasmil escribió:
>
> > Regarding this, I know we could simply use the Dialog module to store al
> > dialogs on the dialog table, and use an external script to end the dialog
Can anyone give me suggestion for storing and manipulating multi-leg
accounting records?
On Thu, Jun 12, 2008 at 2:03 PM, Ruchir <[EMAIL PROTECTED]> wrote:
> Any suggestions from experts? :)
>
>
> On Thu, Jun 12, 2008 at 1:27 PM, Bogdan-Andrei Iancu <
> [EMAIL PROTECTED]> wrote:
>
>> Hi Ruchir,
>
El Thursday 12 June 2008 14:06:22 David Villasmil escribió:
> Regarding this, I know we could simply use the Dialog module to store al
> dialogs on the dialog table, and use an external script to end the dialog
How knows the external script where to generate the BYE?
> On the other hand, Bogdan
>
>
> > Instruct the UAC to send a message every 60 senconds to our server,
>
> And how will you do it?
>
> > and
> > reseting that timer when we receive it, so that we know that the call
> > hasn't ended.
>
>
>
> > If we don't receive that "keep-alive" message we end the dialog with a
> BYE
> > (f
Hi Bastien,
Thank you for your help. I have an other question please: Can I use
dynamique string ($avp(...)) as parameters in remove_hf( ) or append_hf( )?
because when I execute for example remove_hf ("$avp(s:headername)") ($avp
(s:headerName) = "Allow" ), I have not error but the header is not r
Hi,
On Thursday 12 June 2008, Yazid Hadj Said wrote:
> Please, i want to know why when i try to use remove_hf() and append_hf()
> functions with Perl (moduleFunction(func,string1,string2)), I have this
> message:
>
> Jun 12 11:03:53 [12838] ERROR:core:moduleFunc: Module function 'remove_hf'
> is u
Hi,
Please, i want to know why when i try to use remove_hf() and append_hf()
functions with Perl (moduleFunction(func,string1,string2)), I have this
message:
Jun 12 11:03:53 [12838] ERROR:core:moduleFunc: Module function 'remove_hf'
is unsafe. Call is refused.
Jun 12 11:03:53 [12838] ERROR:core:X
On Thursday 12 June 2008, kionez wrote:
> > is used. Some logs (from trunk):
>
> [cut]
>
> > Jun 11 14:37:52 ca ../openser[23446]: INFO:dialog:get_dlg_timeout:
>
> > invalid AVP value, use default timeout
>
> I try to set my debug level to 9, but i can't reproduce this behaviour,
> i never seen "i
Any suggestions from experts? :)
On Thu, Jun 12, 2008 at 1:27 PM, Bogdan-Andrei Iancu <[EMAIL PROTECTED]>
wrote:
> Hi Ruchir,
>
> with multi-leg accounting, you can match all the records belonging to the
> same call by using call_id, to_tag and from_tag. How you format and store
> the CDRS is you
El Thursday 12 June 2008 02:27:05 David Villasmil escribió:
> Instruct the UAC to send a message every 60 senconds to our server,
And how will you do it?
> and
> reseting that timer when we receive it, so that we know that the call
> hasn't ended.
> If we don't receive that "keep-alive" mes
Hi Ruchir,
with multi-leg accounting, you can match all the records belonging to
the same call by using call_id, to_tag and from_tag. How you format and
store the CDRS is your choice :).
Regards,
Bogdan
Ruchir wrote:
> Yeah I just noticed that INVITE & BYE time difference gives accurate
> dur
Hi David,
This should work, if you manage to convince the UAC to periodically send
some with-in the dialog requests.
When sending the BYE via MI command, you could use a local_route to
catch the BYE and do accounting for it from the script.
Regards,
Bogdan
David Villasmil wrote:
> Hello all,
#include// created 11/06/2008 14:41
[cut]
> It seems that in this case the AVP value is not valid, and the
default timeout
> is used. Some logs (from trunk):
[cut]
> Jun 11 14:37:52 ca ../openser[23446]: INFO:dialog:get_dlg_timeout:
invalid AVP
> value, use default timeout
I try to set my
El Thursday 12 June 2008 08:59:55 Pezhman Lali escribió:
> Dear,
> I find that stun and symmetric nat can not work together ,
> but some big sip providers like sipgate or fwd, added stun settings to
> their ata as default without, any question about customer's nat type.
>
> where is the key ?
AFAI
Dear,
I find that stun and symmetric nat can not work together ,
but some big sip providers like sipgate or fwd, added stun settings to their
ata as default without, any question about customer's nat type.
where is the key ?
___
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