Re: [OpenSIPS-Users] CDRTool - disable or minimize normalization

[Jeff Pyle]

In case someone finds this later on with similar issues, I'll tell the rest of 
the story.

Setting the skipNormalize field to 1 or true in each data source did stop the 
normalization.  This stopped the problem of fields changing when I didn't want 
them too.  But, it also caused CDRTool to not display everything the same way 
as a normalized call.

For example, failed calls (486, 503, etc) displayed as "in progress".  I have 
other processes that add Rating info and Prices, and they weren't displayed 
anymore.  Neither was the KBIn or KBOut of a call tha went through a Mediaproxy 
relay.  Hmm.

I solved the issue by normalizing the calls in the stored procedures.  This 
really isn't the same "bringing them into a normal format" the normalizing 
procedure normally does, but rather only setting the flag to 1 once the call 
was complete.  I updated the insert_radacct_record procedure to set 
Normalized='1', unless SipResponseCode=200, in which case we'll set the field 
with the update_radacct_record.  I modified the update_radacct_record procedure 
to set Normalized='1' instead of the '0' it already had.

So, problem solved.  Of course maintaining these updates will require 
re-modifying the procedures each time the a new version of CDRTool requires 
updating the procedures in MySQL.  Definitely worth it, though.  CDRTool is 
still a fantastic utility when used only as a display front-end for the 
database.


- Jeff


From: Jeff Pyle mailto:jp...@fidelityvoice.com>>
Reply-To: OpenSIPS users mailling list 
mailto:users@lists.opensips.org>>
Date: Tue, 8 Mar 2011 12:28:27 -0500
To: OpenSIPS users mailling list 
mailto:users@lists.opensips.org>>
Subject: Re: [OpenSIPS-Users] CDRTool - disable or minimize normalization

Yep, that did it.  Excellent.

From: Adrian Georgescu mailto:a...@ag-projects.com>>
Reply-To: OpenSIPS users mailling list 
mailto:users@lists.opensips.org>>
Date: Tue, 8 Mar 2011 03:40:57 -0500
To: OpenSIPS users mailling list 
mailto:users@lists.opensips.org>>
Subject: Re: [OpenSIPS-Users] CDRTool - disable or minimize normalization

Actually I found it, there is an undocumented variable skipNormalize:

Set skipNormalize to 1 or true in your data source to disable the normalization 
process.

Adrian

On Mar 8, 2011, at 3:32 AM, Jeff Pyle wrote:

Adrian,

I did as you said but it is still normalizing the data.  The top of 
cdr_generic.php now looks like this:

class CDRS {

var $CDR_class   = 'CDR';
var $intAccessCode   = '00';
var $natAccessCode   = '0';
var $maxrowsperpage  = 15;
var $status  = array();
//var $normalizedField = 'Normalized';
var $DestinationIdField  = 'DestinationId';
var $BillingIdField  = 'UserName';

And there is no mention of normalizedField in /etc/cdrtool/global.inc.


- Jeff


From: Adrian Georgescu mailto:a...@ag-projects.com>>
Reply-To: OpenSIPS users mailling list 
mailto:users@lists.opensips.org>>
Date: Wed, 2 Mar 2011 03:52:24 -0500
To: OpenSIPS users mailling list 
mailto:users@lists.opensips.org>>
Subject: Re: [OpenSIPS-Users] CDRTool - disable or minimize normalization

You may comment out any definition of $normalizedField variable in 
cdr_generic.php and global.inc, without it the normalization process will not 
save anything in the database.

Adrian

On Mar 1, 2011, at 11:25 PM, Jeff Pyle wrote:

Hello,

Is there a simple way to disable normalization in CDRTool?  We use it only to 
view the CDRs, not do any billing or rating.  And the Normalization process is 
doing some strange things to some of the fields.

Or perhaps, if it must run, to minimize the impact it has?

Specifically, it is mangling international numbers, then reporting "No 
destination for number ".  It changes some other fields that don't 
necessarily need to be changed, but less harmful.

I've played with the E164 field in global.inc for the data source but it 
doesn't seem to have any effect.


- Jeff
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Re: [OpenSIPS-Users] rtp_proxy_stream2uac and opensips-1.6.4

[Flavio Goncalves]

Hi Kamen,

Thanks for the indication, I found this in the source code of makeann.c

+then
+  echo 
"***
$ECHO_C" 1>&6
+  echo "* Please contact sa...@sippysoft.com
 if you need G.729 support in makeann * $ECHO_C" 1>&6
+  echo 
"***
$ECHO_C" 1>&6

I've sent an email, probably there is a fee for the codec.

Flavio E. Goncalves




2011/3/11 Kamen Petrov 

> Flavio:
>
> http://en.wikipedia.org/wiki/G.729
> "G.729 includes 
> patents from
> several companies and is licensed by Sipro Lab Telecom. "
>
> http://en.wikipedia.org/wiki/GSM#Voice_codecs
>  I think, patent issues
> here too.
>
> *
> Thanks
> -- Kamen*
>
>
>
> On 11 March 2011 16:41, Flavio Goncalves  wrote:
>
>> Razvan,
>>
>> I need to implement this feature with GSM and G729, but I see that this
>> files do not appear in the rtpproxy distribution. Any tip?
>>
>> Best regards,
>>
>> Flavio E. Goncalves
>>
>>
>>
>>
>> 2011/3/11 Razvan Crainea 
>>
>>>  Hello Flavio,
>>>
>>> I don't know why you have to take the check out. Probably this is a
>>> little bug. I will dig into this and let you know as soon as I solve the
>>> problem.
>>>
>>> Regards,
>>> Razvan
>>>
>>>
>>> On 03/11/2011 04:35 PM, Flavio Goncalves wrote:
>>>
>>> Hi Razvan,
>>>
>>>  I got to make it work using a version downloaded from GIT. There was a
>>> mistake in the name of the file, it is working, I have removed the check
>>> from rtpproxy_stream.c
>>>
>>>  Thanks for helping.
>>>
>>> Flavio E. Goncalves
>>>
>>>
>>>
>>>
>>> 2011/3/11 Razvan Crainea 
>>>
  Hi Flavio,

 Can you please give me the output of the command "rtpproxy -v"?

 The word "session" specifies RTPProxy to search for the codec list in
 the initial Update command (when OpenSIPS calls rtpproxy_offer), so there 
 is
 nothing wrong with it.

 This is how RTPProxy works when it receives a Play command:
 Assuming your initial codec list is "3,8,101", your command is
 interpreted like this:
 it checks in the '/var/rtpproxy/prompts/' folder for the 'dmcaller.3'
 file (which should be a GSM encoded file). If found, it starts to play it 
 to
 the uac/uas. Otherwise it checks further for the 'dmcaller.8' file, and so
 on until it finds a suitable media file. If it doesn't, it should log an
 error.
 I guess your problem is that RTPProxy is unable to find a suitable file
 to open.

 Regards,
 Razvan


 On 03/11/2011 03:23 PM, Flavio Goncalves wrote:

  Hi,

  Is there anyone with experience using rtpproxy_stream2uac command. I'm
 trying to use with OpenSIPS 1.6.4 with rtpproxy-1.2.1. I'm getting the
 following error:

  ERROR:nathelper:rtpproxy_stream: required functionality is not
 supported by the version of the RTPproxy running on the selected node.
  Please upgrade the RTPproxy and try again.

  1. I tried to upgrade the RTPPROXY with the on in GIT, but it crashes
 as soon as I start OpenSIPS.

  2. I tried to disabe the check in the rtpproxy_stream2uac, I don't get
 the error but still don't work. In this case:

  OpenSIPS is sending to the RTPPROXY

  DBUG:handle_command: received command "23907_6 P10
 f765564a-317a422b@192.168.1.175 /var/rtpproxy/prompts/dmcaller session
 5cc2dfbec1d32747o2;1 as1dc4b372;1"

  The RTPPROXY is expecting according to the RTPPROXY protocol

  P[args] callid play_name codecs from_tag to_tag

  The mismatch seems to be in the word "session" in this place rtpproxy
 is expecting the codecs.

  I tried everything I could.

  Flavio E. Goncalves



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Re: [OpenSIPS-Users] rtp_proxy_stream2uac and opensips-1.6.4

[Kamen Petrov]

Flavio:

http://en.wikipedia.org/wiki/G.729
"G.729 includes
patents from
several companies and is licensed by Sipro Lab Telecom. "

http://en.wikipedia.org/wiki/GSM#Voice_codecs
 I think, patent issues here
too.

*
Thanks
-- Kamen*


On 11 March 2011 16:41, Flavio Goncalves  wrote:

> Razvan,
>
> I need to implement this feature with GSM and G729, but I see that this
> files do not appear in the rtpproxy distribution. Any tip?
>
> Best regards,
>
> Flavio E. Goncalves
>
>
>
>
> 2011/3/11 Razvan Crainea 
>
>>  Hello Flavio,
>>
>> I don't know why you have to take the check out. Probably this is a little
>> bug. I will dig into this and let you know as soon as I solve the problem.
>>
>> Regards,
>> Razvan
>>
>>
>> On 03/11/2011 04:35 PM, Flavio Goncalves wrote:
>>
>> Hi Razvan,
>>
>>  I got to make it work using a version downloaded from GIT. There was a
>> mistake in the name of the file, it is working, I have removed the check
>> from rtpproxy_stream.c
>>
>>  Thanks for helping.
>>
>> Flavio E. Goncalves
>>
>>
>>
>>
>> 2011/3/11 Razvan Crainea 
>>
>>>  Hi Flavio,
>>>
>>> Can you please give me the output of the command "rtpproxy -v"?
>>>
>>> The word "session" specifies RTPProxy to search for the codec list in the
>>> initial Update command (when OpenSIPS calls rtpproxy_offer), so there is
>>> nothing wrong with it.
>>>
>>> This is how RTPProxy works when it receives a Play command:
>>> Assuming your initial codec list is "3,8,101", your command is
>>> interpreted like this:
>>> it checks in the '/var/rtpproxy/prompts/' folder for the 'dmcaller.3'
>>> file (which should be a GSM encoded file). If found, it starts to play it to
>>> the uac/uas. Otherwise it checks further for the 'dmcaller.8' file, and so
>>> on until it finds a suitable media file. If it doesn't, it should log an
>>> error.
>>> I guess your problem is that RTPProxy is unable to find a suitable file
>>> to open.
>>>
>>> Regards,
>>> Razvan
>>>
>>>
>>> On 03/11/2011 03:23 PM, Flavio Goncalves wrote:
>>>
>>>  Hi,
>>>
>>>  Is there anyone with experience using rtpproxy_stream2uac command. I'm
>>> trying to use with OpenSIPS 1.6.4 with rtpproxy-1.2.1. I'm getting the
>>> following error:
>>>
>>>  ERROR:nathelper:rtpproxy_stream: required functionality is not
>>> supported by the version of the RTPproxy running on the selected node.
>>>  Please upgrade the RTPproxy and try again.
>>>
>>>  1. I tried to upgrade the RTPPROXY with the on in GIT, but it crashes
>>> as soon as I start OpenSIPS.
>>>
>>>  2. I tried to disabe the check in the rtpproxy_stream2uac, I don't get
>>> the error but still don't work. In this case:
>>>
>>>  OpenSIPS is sending to the RTPPROXY
>>>
>>>  DBUG:handle_command: received command "23907_6 P10
>>> f765564a-317a422b@192.168.1.175 /var/rtpproxy/prompts/dmcaller session
>>> 5cc2dfbec1d32747o2;1 as1dc4b372;1"
>>>
>>>  The RTPPROXY is expecting according to the RTPPROXY protocol
>>>
>>>  P[args] callid play_name codecs from_tag to_tag
>>>
>>>  The mismatch seems to be in the word "session" in this place rtpproxy
>>> is expecting the codecs.
>>>
>>>  I tried everything I could.
>>>
>>>  Flavio E. Goncalves
>>>
>>>
>>>
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>>> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
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>>>
>>
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Re: [OpenSIPS-Users] New module: registrant

[Ovidiu Sas]

Each AOR should be checked once during a timer interval.
When the timer interval is configured, an internal timer interval is
computed by dividing the configured timer interval over the number of
hash tables.
For large number of hash tables and a small timer interval, more then
one hash tables needs to be checked (this is not implemented yet).

For example, if you have a hash_size set to 3 it means that you have 8
hash tables.
If the timer interval is set to 15, then the internal timer will be
set to 1 second (15/8=1) and every second a hash table will be checked
(in round robin order).

For a large number of registrations, it is better to use a hash_size
that will lead to very few AORs inside the same hash table.
Hash-es are computed over AOR.


Regards,
Ovidiu Sas

On Fri, Mar 11, 2011 at 12:17 AM, Andrew Pogrebennyk
 wrote:
> Ovidiu,
> great news! This is exactly what I've been looking for. But I'm not sure,
> does timer_interval affect the distribution of registration load in time or
> it affects only re-register? In what units if the hash_size given, do you
> have any examples e.g. how big hash_size we need to distribute in time (a
> few seconds apart) 10, 100 and 1000 registrations?
> Thank you.
>
> On 11.03.2011 07:13, Ovidiu Sas wrote:
>>
>> Hello all,
>>
>> There is a new module available for opensips: registrant.
>> This module allows opensips to register itself on a remote registrar
>> server.
>> For more info, check the README file:
>> http://www.opensips.org/html/docs/modules/devel/registrant.html
>>
>>
>> Regards,
>> Ovidiu Sas
>
> --
> Sincerely,
> Andrew Pogrebennyk
>

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Re: [OpenSIPS-Users] rtp_proxy_stream2uac and opensips-1.6.4

[Flavio Goncalves]

Razvan,

I need to implement this feature with GSM and G729, but I see that this
files do not appear in the rtpproxy distribution. Any tip?

Best regards,

Flavio E. Goncalves




2011/3/11 Razvan Crainea 

>  Hello Flavio,
>
> I don't know why you have to take the check out. Probably this is a little
> bug. I will dig into this and let you know as soon as I solve the problem.
>
> Regards,
> Razvan
>
>
> On 03/11/2011 04:35 PM, Flavio Goncalves wrote:
>
> Hi Razvan,
>
>  I got to make it work using a version downloaded from GIT. There was a
> mistake in the name of the file, it is working, I have removed the check
> from rtpproxy_stream.c
>
>  Thanks for helping.
>
> Flavio E. Goncalves
>
>
>
>
> 2011/3/11 Razvan Crainea 
>
>>  Hi Flavio,
>>
>> Can you please give me the output of the command "rtpproxy -v"?
>>
>> The word "session" specifies RTPProxy to search for the codec list in the
>> initial Update command (when OpenSIPS calls rtpproxy_offer), so there is
>> nothing wrong with it.
>>
>> This is how RTPProxy works when it receives a Play command:
>> Assuming your initial codec list is "3,8,101", your command is interpreted
>> like this:
>> it checks in the '/var/rtpproxy/prompts/' folder for the 'dmcaller.3' file
>> (which should be a GSM encoded file). If found, it starts to play it to the
>> uac/uas. Otherwise it checks further for the 'dmcaller.8' file, and so on
>> until it finds a suitable media file. If it doesn't, it should log an error.
>> I guess your problem is that RTPProxy is unable to find a suitable file to
>> open.
>>
>> Regards,
>> Razvan
>>
>>
>> On 03/11/2011 03:23 PM, Flavio Goncalves wrote:
>>
>>  Hi,
>>
>>  Is there anyone with experience using rtpproxy_stream2uac command. I'm
>> trying to use with OpenSIPS 1.6.4 with rtpproxy-1.2.1. I'm getting the
>> following error:
>>
>>  ERROR:nathelper:rtpproxy_stream: required functionality is not supported
>> by the version of the RTPproxy running on the selected node.  Please upgrade
>> the RTPproxy and try again.
>>
>>  1. I tried to upgrade the RTPPROXY with the on in GIT, but it crashes as
>> soon as I start OpenSIPS.
>>
>>  2. I tried to disabe the check in the rtpproxy_stream2uac, I don't get
>> the error but still don't work. In this case:
>>
>>  OpenSIPS is sending to the RTPPROXY
>>
>>  DBUG:handle_command: received command "23907_6 P10
>> f765564a-317a422b@192.168.1.175 /var/rtpproxy/prompts/dmcaller session
>> 5cc2dfbec1d32747o2;1 as1dc4b372;1"
>>
>>  The RTPPROXY is expecting according to the RTPPROXY protocol
>>
>>  P[args] callid play_name codecs from_tag to_tag
>>
>>  The mismatch seems to be in the word "session" in this place rtpproxy is
>> expecting the codecs.
>>
>>  I tried everything I could.
>>
>>  Flavio E. Goncalves
>>
>>
>>
>> ___
>> Users mailing 
>> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
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>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>
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Re: [OpenSIPS-Users] rtp_proxy_stream2uac and opensips-1.6.4

[Razvan Crainea]

Hello Flavio,

I don't know why you have to take the check out. Probably this is a 
little bug. I will dig into this and let you know as soon as I solve the 
problem.


Regards,
Razvan

On 03/11/2011 04:35 PM, Flavio Goncalves wrote:

Hi Razvan,

I got to make it work using a version downloaded from GIT. There was a 
mistake in the name of the file, it is working, I have removed the 
check from rtpproxy_stream.c


Thanks for helping.

Flavio E. Goncalves



2011/3/11 Razvan Crainea >


Hi Flavio,

Can you please give me the output of the command "rtpproxy -v"?

The word "session" specifies RTPProxy to search for the codec list
in the initial Update command (when OpenSIPS calls
rtpproxy_offer), so there is nothing wrong with it.

This is how RTPProxy works when it receives a Play command:
Assuming your initial codec list is "3,8,101", your command is
interpreted like this:
it checks in the '/var/rtpproxy/prompts/' folder for the
'dmcaller.3' file (which should be a GSM encoded file). If found,
it starts to play it to the uac/uas. Otherwise it checks further
for the 'dmcaller.8' file, and so on until it finds a suitable
media file. If it doesn't, it should log an error.
I guess your problem is that RTPProxy is unable to find a suitable
file to open.

Regards,
Razvan


On 03/11/2011 03:23 PM, Flavio Goncalves wrote:

Hi,

Is there anyone with experience using rtpproxy_stream2uac
command. I'm trying to use with OpenSIPS 1.6.4 with
rtpproxy-1.2.1. I'm getting the following error:

ERROR:nathelper:rtpproxy_stream: required functionality is not
supported by the version of the RTPproxy running on the selected
node.  Please upgrade the RTPproxy and try again.

1. I tried to upgrade the RTPPROXY with the on in GIT, but it
crashes as soon as I start OpenSIPS.

2. I tried to disabe the check in the rtpproxy_stream2uac, I
don't get the error but still don't work. In this case:

OpenSIPS is sending to the RTPPROXY

DBUG:handle_command: received command "23907_6 P10
f765564a-317a422b@192.168.1.175

/var/rtpproxy/prompts/dmcaller session 5cc2dfbec1d32747o2;1
as1dc4b372;1"

The RTPPROXY is expecting according to the RTPPROXY protocol

P[args] callid play_name codecs from_tag to_tag

The mismatch seems to be in the word "session" in this place
rtpproxy is expecting the codecs.

I tried everything I could.

Flavio E. Goncalves


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Re: [OpenSIPS-Users] rtp_proxy_stream2uac and opensips-1.6.4

[Flavio Goncalves]

Hi Razvan,

I got to make it work using a version downloaded from GIT. There was a
mistake in the name of the file, it is working, I have removed the check
from rtpproxy_stream.c

Thanks for helping.

Flavio E. Goncalves




2011/3/11 Razvan Crainea 

>  Hi Flavio,
>
> Can you please give me the output of the command "rtpproxy -v"?
>
> The word "session" specifies RTPProxy to search for the codec list in the
> initial Update command (when OpenSIPS calls rtpproxy_offer), so there is
> nothing wrong with it.
>
> This is how RTPProxy works when it receives a Play command:
> Assuming your initial codec list is "3,8,101", your command is interpreted
> like this:
> it checks in the '/var/rtpproxy/prompts/' folder for the 'dmcaller.3' file
> (which should be a GSM encoded file). If found, it starts to play it to the
> uac/uas. Otherwise it checks further for the 'dmcaller.8' file, and so on
> until it finds a suitable media file. If it doesn't, it should log an error.
> I guess your problem is that RTPProxy is unable to find a suitable file to
> open.
>
> Regards,
> Razvan
>
>
> On 03/11/2011 03:23 PM, Flavio Goncalves wrote:
>
> Hi,
>
>  Is there anyone with experience using rtpproxy_stream2uac command. I'm
> trying to use with OpenSIPS 1.6.4 with rtpproxy-1.2.1. I'm getting the
> following error:
>
>  ERROR:nathelper:rtpproxy_stream: required functionality is not supported
> by the version of the RTPproxy running on the selected node.  Please upgrade
> the RTPproxy and try again.
>
>  1. I tried to upgrade the RTPPROXY with the on in GIT, but it crashes as
> soon as I start OpenSIPS.
>
>  2. I tried to disabe the check in the rtpproxy_stream2uac, I don't get
> the error but still don't work. In this case:
>
>  OpenSIPS is sending to the RTPPROXY
>
>  DBUG:handle_command: received command "23907_6 P10
> f765564a-317a422b@192.168.1.175 /var/rtpproxy/prompts/dmcaller session
> 5cc2dfbec1d32747o2;1 as1dc4b372;1"
>
>  The RTPPROXY is expecting according to the RTPPROXY protocol
>
>  P[args] callid play_name codecs from_tag to_tag
>
>  The mismatch seems to be in the word "session" in this place rtpproxy is
> expecting the codecs.
>
>  I tried everything I could.
>
>  Flavio E. Goncalves
>
>
>
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Re: [OpenSIPS-Users] RPID or FROM?

[Jeff Pyle]

All depends on the carrier you're sending it to.  I've got some where this is 
the case.  I've got others I have to use an RPID, and others still I have to 
use a PAI.  I build AVPs on the inbound portion my proxy called $avp(s:id_name) 
and $avp(s:id_num) with a message flag to indicate privacy.

Then I apply either an RPID or a PAI/Privacy combo in the branch route 
depending on what the carrier I'm about to send to want to see.  If I serial 
fork to another carrier, I'll build those headers on the next round out, again 
in the branch route.  That way any carrier can see what they want to see.


- Jeff


From: Toyima Dias mailto:toyim...@gmail.com>>
Reply-To: OpenSIPS users mailling list 
mailto:users@lists.opensips.org>>
Date: Fri, 11 Mar 2011 09:14:43 -0500
To: OpenSIPS users mailling list 
mailto:users@lists.opensips.org>>
Subject: Re: [OpenSIPS-Users] RPID or FROM?

Ok, so the RPID is not mandatory...

2011/3/11 Laszlo mailto:las...@voipfreak.net>>


2011/3/11 Toyima Dias mailto:toyim...@gmail.com>>

so, the RPID will identify the originator of the call (in terms of the PSNT), 
right? what if i do not set any RPID or PAI, what will be presented as teh 
originator of the call on the pstn destiny?

Then it will be taken from the "From"

($fU)

2011/3/11 Jeff Pyle mailto:jp...@fidelityvoice.com>>
Let me reword that…

Most prefer at least an RPID header.

Alternatively, some prefer a P-Asserted-Identity header.

If you use a PAI header, and want to indicate caller privacy, you'd use the 
Privacy header in conjunction with it.



- Jeff

From: Jeff Pyle mailto:jp...@fidelityvoice.com>>

Reply-To: OpenSIPS users mailling list 
mailto:users@lists.opensips.org>>
Date: Fri, 11 Mar 2011 08:55:00 -0500

To: OpenSIPS users mailling list 
mailto:users@lists.opensips.org>>
Subject: Re: [OpenSIPS-Users] RPID or FROM?

Depends what the terminating carrier gateway wants to see.  At least here in 
the US my experience is most prefer at least an RPID, some a 
P-Asserted-Identity header with a Privacy header if you want to indicate 
restricted caller ID.


- Jeff


From: Toyima Dias mailto:toyim...@gmail.com>>
Reply-To: OpenSIPS users mailling list 
mailto:users@lists.opensips.org>>
Date: Fri, 11 Mar 2011 08:53:08 -0500
To: OpenSIPS users mailling list 
mailto:users@lists.opensips.org>>
Subject: [OpenSIPS-Users] RPID or FROM?

Hello,

I have a question about connectivity to the public network, if a user wants to 
make a call to the pstn, where does he define the originator of the call? in 
the from header? or using a RPID? i'm quite confuse aboout this?

Many thanks!


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Re: [OpenSIPS-Users] rtp_proxy_stream2uac and opensips-1.6.4

[Flavio Goncalves]

Hi Razvan,

rtpproxy -v
Basic version: 20040107
Extension 20050322: Support for multiple RTP streams and MOH
Extension 20060704: Support for extra parameter in the V command
Extension 20071116: Support for RTP re-packetization
Extension 20071218: Support for forking (copying) RTP stream
Extension 20080403: Support for RTP statistics querying
Extension 20081102: Support for setting codecs in the update/lookup command
Extension 20081224: Support for session timeout notifications

Flavio E. Goncalves





2011/3/11 Razvan Crainea 

>  Hi Flavio,
>
> Can you please give me the output of the command "rtpproxy -v"?
>
> The word "session" specifies RTPProxy to search for the codec list in the
> initial Update command (when OpenSIPS calls rtpproxy_offer), so there is
> nothing wrong with it.
>
> This is how RTPProxy works when it receives a Play command:
> Assuming your initial codec list is "3,8,101", your command is interpreted
> like this:
> it checks in the '/var/rtpproxy/prompts/' folder for the 'dmcaller.3' file
> (which should be a GSM encoded file). If found, it starts to play it to the
> uac/uas. Otherwise it checks further for the 'dmcaller.8' file, and so on
> until it finds a suitable media file. If it doesn't, it should log an error.
> I guess your problem is that RTPProxy is unable to find a suitable file to
> open.
>
> Regards,
> Razvan
>
>
> On 03/11/2011 03:23 PM, Flavio Goncalves wrote:
>
> Hi,
>
>  Is there anyone with experience using rtpproxy_stream2uac command. I'm
> trying to use with OpenSIPS 1.6.4 with rtpproxy-1.2.1. I'm getting the
> following error:
>
>  ERROR:nathelper:rtpproxy_stream: required functionality is not supported
> by the version of the RTPproxy running on the selected node.  Please upgrade
> the RTPproxy and try again.
>
>  1. I tried to upgrade the RTPPROXY with the on in GIT, but it crashes as
> soon as I start OpenSIPS.
>
>  2. I tried to disabe the check in the rtpproxy_stream2uac, I don't get
> the error but still don't work. In this case:
>
>  OpenSIPS is sending to the RTPPROXY
>
>  DBUG:handle_command: received command "23907_6 P10
> f765564a-317a422b@192.168.1.175 /var/rtpproxy/prompts/dmcaller session
> 5cc2dfbec1d32747o2;1 as1dc4b372;1"
>
>  The RTPPROXY is expecting according to the RTPPROXY protocol
>
>  P[args] callid play_name codecs from_tag to_tag
>
>  The mismatch seems to be in the word "session" in this place rtpproxy is
> expecting the codecs.
>
>  I tried everything I could.
>
>  Flavio E. Goncalves
>
>
>
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Re: [OpenSIPS-Users] RPID or FROM?

[Toyima Dias]

Ok, so the RPID is not mandatory...

2011/3/11 Laszlo 

>
>
> 2011/3/11 Toyima Dias 
>
> so, the RPID will identify the originator of the call (in terms of the
>> PSNT), right? what if i do not set any RPID or PAI, what will be presented
>> as teh originator of the call on the pstn destiny?
>>
>
> Then it will be taken from the "From"
>
> ($fU)
>
>>
>> 2011/3/11 Jeff Pyle 
>>
>>>   Let me reword that…
>>>
>>> Most prefer at least an RPID header.
>>>
>>> Alternatively, some prefer a P-Asserted-Identity header.
>>>
>>> If you use a PAI header, and want to indicate caller privacy, you'd use
>>> the Privacy header in conjunction with it.
>>>
>>>
>>>
>>> - Jeff
>>>
>>> From: Jeff Pyle 
>>>
>>> Reply-To: OpenSIPS users mailling list 
>>> Date: Fri, 11 Mar 2011 08:55:00 -0500
>>>
>>> To: OpenSIPS users mailling list 
>>> Subject: Re: [OpenSIPS-Users] RPID or FROM?
>>>
>>>  Depends what the terminating carrier gateway wants to see.  At least
>>> here in the US my experience is most prefer at least an RPID, some a
>>> P-Asserted-Identity header with a Privacy header if you want to indicate
>>> restricted caller ID.
>>>
>>>
>>> - Jeff
>>>
>>>
>>> From: Toyima Dias 
>>> Reply-To: OpenSIPS users mailling list 
>>> Date: Fri, 11 Mar 2011 08:53:08 -0500
>>> To: OpenSIPS users mailling list 
>>> Subject: [OpenSIPS-Users] RPID or FROM?
>>>
>>> Hello,
>>>
>>> I have a question about connectivity to the public network, if a user
>>> wants to make a call to the pstn, where does he define the originator of the
>>> call? in the from header? or using a RPID? i'm quite confuse aboout this?
>>>
>>> Many thanks!
>>>
>>>
>>> ___
>>> Users mailing list
>>> 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
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Re: [OpenSIPS-Users] rtp_proxy_stream2uac and opensips-1.6.4

[Razvan Crainea]

Hi Flavio,

Can you please give me the output of the command "rtpproxy -v"?

The word "session" specifies RTPProxy to search for the codec list in 
the initial Update command (when OpenSIPS calls rtpproxy_offer), so 
there is nothing wrong with it.


This is how RTPProxy works when it receives a Play command:
Assuming your initial codec list is "3,8,101", your command is 
interpreted like this:
it checks in the '/var/rtpproxy/prompts/' folder for the 'dmcaller.3' 
file (which should be a GSM encoded file). If found, it starts to play 
it to the uac/uas. Otherwise it checks further for the 'dmcaller.8' 
file, and so on until it finds a suitable media file. If it doesn't, it 
should log an error.
I guess your problem is that RTPProxy is unable to find a suitable file 
to open.


Regards,
Razvan

On 03/11/2011 03:23 PM, Flavio Goncalves wrote:

Hi,

Is there anyone with experience using rtpproxy_stream2uac command. I'm 
trying to use with OpenSIPS 1.6.4 with rtpproxy-1.2.1. I'm getting the 
following error:


ERROR:nathelper:rtpproxy_stream: required functionality is not 
supported by the version of the RTPproxy running on the selected node. 
 Please upgrade the RTPproxy and try again.


1. I tried to upgrade the RTPPROXY with the on in GIT, but it crashes 
as soon as I start OpenSIPS.


2. I tried to disabe the check in the rtpproxy_stream2uac, I don't get 
the error but still don't work. In this case:


OpenSIPS is sending to the RTPPROXY

DBUG:handle_command: received command "23907_6 P10 
f765564a-317a422b@192.168.1.175 
 
/var/rtpproxy/prompts/dmcaller session 5cc2dfbec1d32747o2;1 as1dc4b372;1"


The RTPPROXY is expecting according to the RTPPROXY protocol

P[args] callid play_name codecs from_tag to_tag

The mismatch seems to be in the word "session" in this place rtpproxy 
is expecting the codecs.


I tried everything I could.

Flavio E. Goncalves


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Re: [OpenSIPS-Users] RPID or FROM?

[Laszlo]

2011/3/11 Toyima Dias 

> so, the RPID will identify the originator of the call (in terms of the
> PSNT), right? what if i do not set any RPID or PAI, what will be presented
> as teh originator of the call on the pstn destiny?
>

Then it will be taken from the "From"

($fU)

>
> 2011/3/11 Jeff Pyle 
>
>>  Let me reword that…
>>
>> Most prefer at least an RPID header.
>>
>> Alternatively, some prefer a P-Asserted-Identity header.
>>
>> If you use a PAI header, and want to indicate caller privacy, you'd use
>> the Privacy header in conjunction with it.
>>
>>
>>
>> - Jeff
>>
>>  From: Jeff Pyle 
>>
>> Reply-To: OpenSIPS users mailling list 
>> Date: Fri, 11 Mar 2011 08:55:00 -0500
>>
>> To: OpenSIPS users mailling list 
>> Subject: Re: [OpenSIPS-Users] RPID or FROM?
>>
>>  Depends what the terminating carrier gateway wants to see.  At least
>> here in the US my experience is most prefer at least an RPID, some a
>> P-Asserted-Identity header with a Privacy header if you want to indicate
>> restricted caller ID.
>>
>>
>> - Jeff
>>
>>
>>  From: Toyima Dias 
>> Reply-To: OpenSIPS users mailling list 
>> Date: Fri, 11 Mar 2011 08:53:08 -0500
>> To: OpenSIPS users mailling list 
>> Subject: [OpenSIPS-Users] RPID or FROM?
>>
>> Hello,
>>
>> I have a question about connectivity to the public network, if a user
>> wants to make a call to the pstn, where does he define the originator of the
>> call? in the from header? or using a RPID? i'm quite confuse aboout this?
>>
>> Many thanks!
>>
>>
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>
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Re: [OpenSIPS-Users] RPID or FROM?

[Toyima Dias]

so, the RPID will identify the originator of the call (in terms of the
PSNT), right? what if i do not set any RPID or PAI, what will be presented
as teh originator of the call on the pstn destiny?

2011/3/11 Jeff Pyle 

>  Let me reword that…
>
> Most prefer at least an RPID header.
>
> Alternatively, some prefer a P-Asserted-Identity header.
>
> If you use a PAI header, and want to indicate caller privacy, you'd use the
> Privacy header in conjunction with it.
>
>
>
> - Jeff
>
> From: Jeff Pyle 
>
> Reply-To: OpenSIPS users mailling list 
> Date: Fri, 11 Mar 2011 08:55:00 -0500
>
> To: OpenSIPS users mailling list 
> Subject: Re: [OpenSIPS-Users] RPID or FROM?
>
>  Depends what the terminating carrier gateway wants to see.  At least here
> in the US my experience is most prefer at least an RPID, some a
> P-Asserted-Identity header with a Privacy header if you want to indicate
> restricted caller ID.
>
>
> - Jeff
>
>
> From: Toyima Dias 
> Reply-To: OpenSIPS users mailling list 
> Date: Fri, 11 Mar 2011 08:53:08 -0500
> To: OpenSIPS users mailling list 
> Subject: [OpenSIPS-Users] RPID or FROM?
>
> Hello,
>
> I have a question about connectivity to the public network, if a user wants
> to make a call to the pstn, where does he define the originator of the call?
> in the from header? or using a RPID? i'm quite confuse aboout this?
>
> Many thanks!
>
>
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> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
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[OpenSIPS-Users] changing the R-URI of an external domain

[Toyima Dias]

Hello,

If opensips receives a request with a R-URI the same of the proxy it will
make any changes on the R-URI as i want, right? but what about if the
domain of the R-URI is not the one of the opensips proxy? (its behavior
should be as stated on the section 16.12 of the rfc..) could i make changes
on the R-URI before sending it to the destination? like for example changing
the userid or just the domain, or maybe the hole r-uri (i think this should
not be possible because of the rfc 3261 16.12 section)

BEst Regards
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Re: [OpenSIPS-Users] RPID or FROM?

[Jeff Pyle]

Let me reword that…

Most prefer at least an RPID header.

Alternatively, some prefer a P-Asserted-Identity header.

If you use a PAI header, and want to indicate caller privacy, you'd use the 
Privacy header in conjunction with it.



- Jeff

From: Jeff Pyle mailto:jp...@fidelityvoice.com>>
Reply-To: OpenSIPS users mailling list 
mailto:users@lists.opensips.org>>
Date: Fri, 11 Mar 2011 08:55:00 -0500
To: OpenSIPS users mailling list 
mailto:users@lists.opensips.org>>
Subject: Re: [OpenSIPS-Users] RPID or FROM?

Depends what the terminating carrier gateway wants to see.  At least here in 
the US my experience is most prefer at least an RPID, some a 
P-Asserted-Identity header with a Privacy header if you want to indicate 
restricted caller ID.


- Jeff


From: Toyima Dias mailto:toyim...@gmail.com>>
Reply-To: OpenSIPS users mailling list 
mailto:users@lists.opensips.org>>
Date: Fri, 11 Mar 2011 08:53:08 -0500
To: OpenSIPS users mailling list 
mailto:users@lists.opensips.org>>
Subject: [OpenSIPS-Users] RPID or FROM?

Hello,

I have a question about connectivity to the public network, if a user wants to 
make a call to the pstn, where does he define the originator of the call? in 
the from header? or using a RPID? i'm quite confuse aboout this?

Many thanks!

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Re: [OpenSIPS-Users] RPID or FROM?

[Jeff Pyle]

Depends what the terminating carrier gateway wants to see.  At least here in 
the US my experience is most prefer at least an RPID, some a 
P-Asserted-Identity header with a Privacy header if you want to indicate 
restricted caller ID.


- Jeff


From: Toyima Dias mailto:toyim...@gmail.com>>
Reply-To: OpenSIPS users mailling list 
mailto:users@lists.opensips.org>>
Date: Fri, 11 Mar 2011 08:53:08 -0500
To: OpenSIPS users mailling list 
mailto:users@lists.opensips.org>>
Subject: [OpenSIPS-Users] RPID or FROM?

Hello,

I have a question about connectivity to the public network, if a user wants to 
make a call to the pstn, where does he define the originator of the call? in 
the from header? or using a RPID? i'm quite confuse aboout this?

Many thanks!

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[OpenSIPS-Users] RPID or FROM?

[Toyima Dias]

Hello,

I have a question about connectivity to the public network, if a user wants
to make a call to the pstn, where does he define the originator of the call?
in the from header? or using a RPID? i'm quite confuse aboout this?

Many thanks!
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[OpenSIPS-Users] rtp_proxy_stream2uac and opensips-1.6.4

[Flavio Goncalves]

Hi,

Is there anyone with experience using rtpproxy_stream2uac command. I'm
trying to use with OpenSIPS 1.6.4 with rtpproxy-1.2.1. I'm getting the
following error:

ERROR:nathelper:rtpproxy_stream: required functionality is not supported by
the version of the RTPproxy running on the selected node.  Please upgrade
the RTPproxy and try again.

1. I tried to upgrade the RTPPROXY with the on in GIT, but it crashes as
soon as I start OpenSIPS.

2. I tried to disabe the check in the rtpproxy_stream2uac, I don't get the
error but still don't work. In this case:

OpenSIPS is sending to the RTPPROXY

DBUG:handle_command: received command "23907_6 P10
f765564a-317a422b@192.168.1.175 /var/rtpproxy/prompts/dmcaller session
5cc2dfbec1d32747o2;1 as1dc4b372;1"

The RTPPROXY is expecting according to the RTPPROXY protocol

P[args] callid play_name codecs from_tag to_tag

The mismatch seems to be in the word "session" in this place rtpproxy is
expecting the codecs.

I tried everything I could.

Flavio E. Goncalves
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Re: [OpenSIPS-Users] opensips 1.6.4 out of memory

[Iulian Macare]

I put the latest dialog module from svn; I will see if it will still happen.
Thank you

On Fri, Mar 11, 2011 at 1:04 PM, Anca Vamanu  wrote:

>  Hi Iulian,
>
> I suggest to update the dialog module from svn, branch 1.6. There was a
> memory leak discovered after the 1.6.2 release and it was fixed in January.
> It might also be showing up in your configuration.
>
> Regards,
> 
> Anca Vamanu
> OpenSIPS Developer
>
> On 03/11/2011 11:39 AM, Iulian Macare wrote:
>
> That's the version I am using. 1.6.4-2 ; I tried to double the SHM
> parameter and recompile but the same problem happened.
>
> opensips-1.6.4-2-notls_src.tar.gz
>
> On Fri, Mar 11, 2011 at 11:37 AM, Denis Putyato  wrote:
>
>>  Hello
>>
>>
>>
>> I had the same problem on 1.6.4, you should use 1.6.4-2 version
>>
>>
>>
>> *From:* users-boun...@lists.opensips.org [mailto:
>> users-boun...@lists.opensips.org] *On Behalf Of *Iulian Macare
>> *Sent:* Friday, March 11, 2011 12:24 PM
>> *To:* users@lists.opensips.org
>> *Subject:* [OpenSIPS-Users] opensips 1.6.4 out of memory
>>
>>
>>
>> 2-3 times per day my opensips configuration with 300 channels and a load
>> balancer fails ; It's opensips 1.6.4 on centos 5.5. 32 bit
>>
>> The erors I get are :
>>
>> Any ideas?
>>
>> Mar 11 11:15:08 opensipsh /usr/local/sbin/opensips[1718]: ERROR:tm:new_t:
>> out of mem
>> Mar 11 11:15:08 opensipsh /usr/local/sbin/opensips[1701]:
>> ERROR:tm:sip_msg_cloner: no more share memory
>> Mar 11 11:15:08 opensipsh /usr/local/sbin/opensips[1716]:
>> ERROR:dialog:dlg_create_dialog: could not add further info to the dialog
>> Mar 11 11:15:08 opensipsh /usr/local/sbin/opensips[1706]:
>> ERROR:dialog:dlg_create_dialog: could not add further info to the dialog
>> Mar 11 11:15:08 opensipsh /usr/local/sbin/opensips[1727]:
>> ERROR:tm:t_newtran: new_t failed
>> Mar 11 11:15:08 opensipsh /usr/local/sbin/opensips[1704]:
>> ERROR:tm:t_newtran: new_t failed
>> Mar 11 11:15:08 opensipsh /usr/local/sbin/opensips[1700]:
>> ERROR:load_balancer:do_load_balance: failed to create dialog
>> Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1698]:
>> ERROR:tm:t_newtran: new_t failed
>> Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1723]:
>> ERROR:tm:t_newtran: new_t failed
>> Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1725]:
>> ERROR:core:print_rr_body: too many RR
>> Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1708]:
>> ERROR:dialog:dlg_add_leg_info: Failed to resize legs array
>> Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1712]:
>> ERROR:dialog:dlg_add_leg_info: Failed to resize legs array
>> Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1714]:
>> ERROR:dialog:dlg_create_dialog: could not add further info to the dialog
>> Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1720]:
>> ERROR:tm:t_newtran: new_t failed
>> Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1721]: ERROR:tm:new_t:
>> out of mem
>> Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1710]:
>> ERROR:tm:sip_msg_cloner: no more share memory
>> Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1718]:
>> ERROR:tm:sip_msg_cloner: no more share memory
>> Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1701]:
>> ERROR:tm:t_newtran: new_t failed
>> Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1716]:
>> ERROR:dialog:init_leg_info: dlg_add_leg_info failed
>> Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1706]:
>> ERROR:dialog:init_leg_info: dlg_add_leg_info failed
>> Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1727]: ERROR:tm:new_t:
>> out of mem
>> Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1704]: ERROR:tm:new_t:
>> out of mem
>> Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1700]:
>> ERROR:dialog:dlg_create_dialog: could not add further info to the dialog
>> Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1698]: ERROR:tm:new_t:
>> out of mem
>> Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1723]: ERROR:tm:new_t:
>> out of mem
>> Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1708]:
>> ERROR:tm:t_newtran: new_t failed
>> Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1712]:
>> ERROR:tm:t_newtran: new_t failed
>> Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1701]:
>> ERROR:tm:t_newtran: new_t failed
>> Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1716]: ERROR:tm:new_t:
>> out of mem
>> Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1706]: ERROR:tm:new_t:
>> out of mem
>> Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1727]:
>> ERROR:dialog:get_routing_info: failed to print route records
>> Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1725]:
>> ERROR:dialog:dlg_add_leg_info: Failed to resize legs array
>> Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1704]:
>> ERROR:dialog:get_routing_info: failed to print route records
>> Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1700]:
>> ERROR:dialog:get_routing_info: failed to print route records
>> Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1698]:
>> ERROR:dialog

Re: [OpenSIPS-Users] opensips 1.6.4 out of memory

[Vlad Paiu]

Hello Iulian,

I took a look at the error output, and something seems very wrong. The 
errors

ERROR:core:print_rr_body: too many RR
suggest the fact the your SIP messages have more than 64 Record-Route 
headers, which is huge. Are you sure you are not having a traffic loop 
problem ? Please post a traffic capture of SIP messages that trigger the 
out of mem problems.


Regards,

--
Vlad Paiu
OpenSIPS Developer



On 03/11/2011 11:23 AM, Iulian Macare wrote:
2-3 times per day my opensips configuration with 300 channels and a 
load balancer fails ; It's opensips 1.6.4 on centos 5.5. 32 bit


The erors I get are :

Any ideas?

Mar 11 11:15:08 opensipsh /usr/local/sbin/opensips[1718]: 
ERROR:tm:new_t: out of mem
Mar 11 11:15:08 opensipsh /usr/local/sbin/opensips[1701]: 
ERROR:tm:sip_msg_cloner: no more share memory
Mar 11 11:15:08 opensipsh /usr/local/sbin/opensips[1716]: 
ERROR:dialog:dlg_create_dialog: could not add further info to the dialog
Mar 11 11:15:08 opensipsh /usr/local/sbin/opensips[1706]: 
ERROR:dialog:dlg_create_dialog: could not add further info to the dialog
Mar 11 11:15:08 opensipsh /usr/local/sbin/opensips[1727]: 
ERROR:tm:t_newtran: new_t failed
Mar 11 11:15:08 opensipsh /usr/local/sbin/opensips[1704]: 
ERROR:tm:t_newtran: new_t failed
Mar 11 11:15:08 opensipsh /usr/local/sbin/opensips[1700]: 
ERROR:load_balancer:do_load_balance: failed to create dialog
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1698]: 
ERROR:tm:t_newtran: new_t failed
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1723]: 
ERROR:tm:t_newtran: new_t failed
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1725]: 
ERROR:core:print_rr_body: too many RR
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1708]: 
ERROR:dialog:dlg_add_leg_info: Failed to resize legs array
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1712]: 
ERROR:dialog:dlg_add_leg_info: Failed to resize legs array
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1714]: 
ERROR:dialog:dlg_create_dialog: could not add further info to the dialog
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1720]: 
ERROR:tm:t_newtran: new_t failed
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1721]: 
ERROR:tm:new_t: out of mem
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1710]: 
ERROR:tm:sip_msg_cloner: no more share memory
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1718]: 
ERROR:tm:sip_msg_cloner: no more share memory
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1701]: 
ERROR:tm:t_newtran: new_t failed
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1716]: 
ERROR:dialog:init_leg_info: dlg_add_leg_info failed
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1706]: 
ERROR:dialog:init_leg_info: dlg_add_leg_info failed
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1727]: 
ERROR:tm:new_t: out of mem
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1704]: 
ERROR:tm:new_t: out of mem
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1700]: 
ERROR:dialog:dlg_create_dialog: could not add further info to the dialog
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1698]: 
ERROR:tm:new_t: out of mem
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1723]: 
ERROR:tm:new_t: out of mem
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1708]: 
ERROR:tm:t_newtran: new_t failed
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1712]: 
ERROR:tm:t_newtran: new_t failed
Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1701]: 
ERROR:tm:t_newtran: new_t failed
Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1716]: 
ERROR:tm:new_t: out of mem
Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1706]: 
ERROR:tm:new_t: out of mem
Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1727]: 
ERROR:dialog:get_routing_info: failed to print route records
Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1725]: 
ERROR:dialog:dlg_add_leg_info: Failed to resize legs array
Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1704]: 
ERROR:dialog:get_routing_info: failed to print route records
Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1700]: 
ERROR:dialog:get_routing_info: failed to print route records
Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1698]: 
ERROR:dialog:get_routing_info: failed to print route records
Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1723]: 
ERROR:dialog:dlg_add_leg_info: no more shm mem
Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1708]: 
ERROR:dialog:dlg_add_leg_info: Failed to resize legs array
Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1712]: 
ERROR:dialog:dlg_add_leg_info: Failed to resize legs array
Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1714]: 
ERROR:dialog:build_new_dlg: no more shm mem (202)
Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1720]: 
ERROR:tm:shm_clone_proxy: no more shm memory
Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1721]: 
ERROR:tm:sip_msg_cloner: no more share memory
Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1710]: 
ERROR:dial

Re: [OpenSIPS-Users] opensips 1.6.4 out of memory

[Anca Vamanu]

Hi Iulian,

I suggest to update the dialog module from svn, branch 1.6. There was a 
memory leak discovered after the 1.6.2 release and it was fixed in 
January. It might also be showing up in your configuration.


Regards,

Anca Vamanu
OpenSIPS Developer

On 03/11/2011 11:39 AM, Iulian Macare wrote:
That's the version I am using. 1.6.4-2 ; I tried to double the SHM 
parameter and recompile but the same problem happened.


opensips-1.6.4-2-notls_src.tar.gz

On Fri, Mar 11, 2011 at 11:37 AM, Denis Putyato > wrote:


Hello

I had the same problem on 1.6.4, you should use 1.6.4-2 version

*From:*users-boun...@lists.opensips.org

[mailto:users-boun...@lists.opensips.org
] *On Behalf Of *Iulian
Macare
*Sent:* Friday, March 11, 2011 12:24 PM
*To:* users@lists.opensips.org 
*Subject:* [OpenSIPS-Users] opensips 1.6.4 out of memory

2-3 times per day my opensips configuration with 300 channels and
a load balancer fails ; It's opensips 1.6.4 on centos 5.5. 32 bit

The erors I get are :

Any ideas?

Mar 11 11:15:08 opensipsh /usr/local/sbin/opensips[1718]:
ERROR:tm:new_t: out of mem
Mar 11 11:15:08 opensipsh /usr/local/sbin/opensips[1701]:
ERROR:tm:sip_msg_cloner: no more share memory
Mar 11 11:15:08 opensipsh /usr/local/sbin/opensips[1716]:
ERROR:dialog:dlg_create_dialog: could not add further info to the
dialog
Mar 11 11:15:08 opensipsh /usr/local/sbin/opensips[1706]:
ERROR:dialog:dlg_create_dialog: could not add further info to the
dialog
Mar 11 11:15:08 opensipsh /usr/local/sbin/opensips[1727]:
ERROR:tm:t_newtran: new_t failed
Mar 11 11:15:08 opensipsh /usr/local/sbin/opensips[1704]:
ERROR:tm:t_newtran: new_t failed
Mar 11 11:15:08 opensipsh /usr/local/sbin/opensips[1700]:
ERROR:load_balancer:do_load_balance: failed to create dialog
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1698]:
ERROR:tm:t_newtran: new_t failed
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1723]:
ERROR:tm:t_newtran: new_t failed
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1725]:
ERROR:core:print_rr_body: too many RR
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1708]:
ERROR:dialog:dlg_add_leg_info: Failed to resize legs array
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1712]:
ERROR:dialog:dlg_add_leg_info: Failed to resize legs array
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1714]:
ERROR:dialog:dlg_create_dialog: could not add further info to the
dialog
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1720]:
ERROR:tm:t_newtran: new_t failed
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1721]:
ERROR:tm:new_t: out of mem
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1710]:
ERROR:tm:sip_msg_cloner: no more share memory
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1718]:
ERROR:tm:sip_msg_cloner: no more share memory
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1701]:
ERROR:tm:t_newtran: new_t failed
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1716]:
ERROR:dialog:init_leg_info: dlg_add_leg_info failed
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1706]:
ERROR:dialog:init_leg_info: dlg_add_leg_info failed
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1727]:
ERROR:tm:new_t: out of mem
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1704]:
ERROR:tm:new_t: out of mem
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1700]:
ERROR:dialog:dlg_create_dialog: could not add further info to the
dialog
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1698]:
ERROR:tm:new_t: out of mem
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1723]:
ERROR:tm:new_t: out of mem
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1708]:
ERROR:tm:t_newtran: new_t failed
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1712]:
ERROR:tm:t_newtran: new_t failed
Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1701]:
ERROR:tm:t_newtran: new_t failed
Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1716]:
ERROR:tm:new_t: out of mem
Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1706]:
ERROR:tm:new_t: out of mem
Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1727]:
ERROR:dialog:get_routing_info: failed to print route records
Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1725]:
ERROR:dialog:dlg_add_leg_info: Failed to resize legs array
Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1704]:
ERROR:dialog:get_routing_info: failed to print route records
Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1700]:
ERROR:dialog:get_routing_info: failed to print route records
Mar 11 11:15:04 opensip

Re: [OpenSIPS-Users] opensips 1.6.4 out of memory

[Iulian Macare]

That's the version I am using. 1.6.4-2 ; I tried to double the SHM parameter
and recompile but the same problem happened.

opensips-1.6.4-2-notls_src.tar.gz

On Fri, Mar 11, 2011 at 11:37 AM, Denis Putyato  wrote:

> Hello
>
>
>
> I had the same problem on 1.6.4, you should use 1.6.4-2 version
>
>
>
> *From:* users-boun...@lists.opensips.org [mailto:
> users-boun...@lists.opensips.org] *On Behalf Of *Iulian Macare
> *Sent:* Friday, March 11, 2011 12:24 PM
> *To:* users@lists.opensips.org
> *Subject:* [OpenSIPS-Users] opensips 1.6.4 out of memory
>
>
>
> 2-3 times per day my opensips configuration with 300 channels and a load
> balancer fails ; It's opensips 1.6.4 on centos 5.5. 32 bit
>
> The erors I get are :
>
> Any ideas?
>
> Mar 11 11:15:08 opensipsh /usr/local/sbin/opensips[1718]: ERROR:tm:new_t:
> out of mem
> Mar 11 11:15:08 opensipsh /usr/local/sbin/opensips[1701]:
> ERROR:tm:sip_msg_cloner: no more share memory
> Mar 11 11:15:08 opensipsh /usr/local/sbin/opensips[1716]:
> ERROR:dialog:dlg_create_dialog: could not add further info to the dialog
> Mar 11 11:15:08 opensipsh /usr/local/sbin/opensips[1706]:
> ERROR:dialog:dlg_create_dialog: could not add further info to the dialog
> Mar 11 11:15:08 opensipsh /usr/local/sbin/opensips[1727]:
> ERROR:tm:t_newtran: new_t failed
> Mar 11 11:15:08 opensipsh /usr/local/sbin/opensips[1704]:
> ERROR:tm:t_newtran: new_t failed
> Mar 11 11:15:08 opensipsh /usr/local/sbin/opensips[1700]:
> ERROR:load_balancer:do_load_balance: failed to create dialog
> Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1698]:
> ERROR:tm:t_newtran: new_t failed
> Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1723]:
> ERROR:tm:t_newtran: new_t failed
> Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1725]:
> ERROR:core:print_rr_body: too many RR
> Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1708]:
> ERROR:dialog:dlg_add_leg_info: Failed to resize legs array
> Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1712]:
> ERROR:dialog:dlg_add_leg_info: Failed to resize legs array
> Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1714]:
> ERROR:dialog:dlg_create_dialog: could not add further info to the dialog
> Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1720]:
> ERROR:tm:t_newtran: new_t failed
> Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1721]: ERROR:tm:new_t:
> out of mem
> Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1710]:
> ERROR:tm:sip_msg_cloner: no more share memory
> Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1718]:
> ERROR:tm:sip_msg_cloner: no more share memory
> Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1701]:
> ERROR:tm:t_newtran: new_t failed
> Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1716]:
> ERROR:dialog:init_leg_info: dlg_add_leg_info failed
> Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1706]:
> ERROR:dialog:init_leg_info: dlg_add_leg_info failed
> Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1727]: ERROR:tm:new_t:
> out of mem
> Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1704]: ERROR:tm:new_t:
> out of mem
> Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1700]:
> ERROR:dialog:dlg_create_dialog: could not add further info to the dialog
> Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1698]: ERROR:tm:new_t:
> out of mem
> Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1723]: ERROR:tm:new_t:
> out of mem
> Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1708]:
> ERROR:tm:t_newtran: new_t failed
> Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1712]:
> ERROR:tm:t_newtran: new_t failed
> Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1701]:
> ERROR:tm:t_newtran: new_t failed
> Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1716]: ERROR:tm:new_t:
> out of mem
> Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1706]: ERROR:tm:new_t:
> out of mem
> Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1727]:
> ERROR:dialog:get_routing_info: failed to print route records
> Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1725]:
> ERROR:dialog:dlg_add_leg_info: Failed to resize legs array
> Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1704]:
> ERROR:dialog:get_routing_info: failed to print route records
> Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1700]:
> ERROR:dialog:get_routing_info: failed to print route records
> Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1698]:
> ERROR:dialog:get_routing_info: failed to print route records
> Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1723]:
> ERROR:dialog:dlg_add_leg_info: no more shm mem
> Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1708]:
> ERROR:dialog:dlg_add_leg_info: Failed to resize legs array
> Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1712]:
> ERROR:dialog:dlg_add_leg_info: Failed to resize legs array
> Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1714]:
> ERROR:dialog:build_new_dlg: no more shm mem (202)
> Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1720]:
> 

Re: [OpenSIPS-Users] opensips 1.6.4 out of memory

[Denis Putyato]

Hello

 

I had the same problem on 1.6.4, you should use 1.6.4-2 version

 

From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Iulian Macare
Sent: Friday, March 11, 2011 12:24 PM
To: users@lists.opensips.org
Subject: [OpenSIPS-Users] opensips 1.6.4 out of memory

 

2-3 times per day my opensips configuration with 300 channels and a load 
balancer fails ; It's opensips 1.6.4 on centos 5.5. 32 bit 

The erors I get are :

Any ideas?

Mar 11 11:15:08 opensipsh /usr/local/sbin/opensips[1718]: ERROR:tm:new_t: out 
of mem
Mar 11 11:15:08 opensipsh /usr/local/sbin/opensips[1701]: 
ERROR:tm:sip_msg_cloner: no more share memory
Mar 11 11:15:08 opensipsh /usr/local/sbin/opensips[1716]: 
ERROR:dialog:dlg_create_dialog: could not add further info to the dialog
Mar 11 11:15:08 opensipsh /usr/local/sbin/opensips[1706]: 
ERROR:dialog:dlg_create_dialog: could not add further info to the dialog
Mar 11 11:15:08 opensipsh /usr/local/sbin/opensips[1727]: ERROR:tm:t_newtran: 
new_t failed
Mar 11 11:15:08 opensipsh /usr/local/sbin/opensips[1704]: ERROR:tm:t_newtran: 
new_t failed
Mar 11 11:15:08 opensipsh /usr/local/sbin/opensips[1700]: 
ERROR:load_balancer:do_load_balance: failed to create dialog
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1698]: ERROR:tm:t_newtran: 
new_t failed
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1723]: ERROR:tm:t_newtran: 
new_t failed
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1725]: 
ERROR:core:print_rr_body: too many RR
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1708]: 
ERROR:dialog:dlg_add_leg_info: Failed to resize legs array
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1712]: 
ERROR:dialog:dlg_add_leg_info: Failed to resize legs array
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1714]: 
ERROR:dialog:dlg_create_dialog: could not add further info to the dialog
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1720]: ERROR:tm:t_newtran: 
new_t failed
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1721]: ERROR:tm:new_t: out 
of mem
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1710]: 
ERROR:tm:sip_msg_cloner: no more share memory
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1718]: 
ERROR:tm:sip_msg_cloner: no more share memory
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1701]: ERROR:tm:t_newtran: 
new_t failed
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1716]: 
ERROR:dialog:init_leg_info: dlg_add_leg_info failed
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1706]: 
ERROR:dialog:init_leg_info: dlg_add_leg_info failed
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1727]: ERROR:tm:new_t: out 
of mem
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1704]: ERROR:tm:new_t: out 
of mem
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1700]: 
ERROR:dialog:dlg_create_dialog: could not add further info to the dialog
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1698]: ERROR:tm:new_t: out 
of mem
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1723]: ERROR:tm:new_t: out 
of mem
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1708]: ERROR:tm:t_newtran: 
new_t failed
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1712]: ERROR:tm:t_newtran: 
new_t failed
Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1701]: ERROR:tm:t_newtran: 
new_t failed
Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1716]: ERROR:tm:new_t: out 
of mem
Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1706]: ERROR:tm:new_t: out 
of mem
Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1727]: 
ERROR:dialog:get_routing_info: failed to print route records
Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1725]: 
ERROR:dialog:dlg_add_leg_info: Failed to resize legs array
Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1704]: 
ERROR:dialog:get_routing_info: failed to print route records
Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1700]: 
ERROR:dialog:get_routing_info: failed to print route records
Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1698]: 
ERROR:dialog:get_routing_info: failed to print route records
Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1723]: 
ERROR:dialog:dlg_add_leg_info: no more shm mem
Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1708]: 
ERROR:dialog:dlg_add_leg_info: Failed to resize legs array
Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1712]: 
ERROR:dialog:dlg_add_leg_info: Failed to resize legs array
Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1714]: 
ERROR:dialog:build_new_dlg: no more shm mem (202)
Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1720]: 
ERROR:tm:shm_clone_proxy: no more shm memory
Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1721]: 
ERROR:tm:sip_msg_cloner: no more share memory
Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1710]: 
ERROR:dialog:dlg_add_leg_info: Failed to resize legs array
Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1718]: 
ERROR:core:build_req_buf_from_sip_req: out of p

[OpenSIPS-Users] opensips 1.6.4 out of memory

[Iulian Macare]

2-3 times per day my opensips configuration with 300 channels and a load
balancer fails ; It's opensips 1.6.4 on centos 5.5. 32 bit

The erors I get are :

Any ideas?

Mar 11 11:15:08 opensipsh /usr/local/sbin/opensips[1718]: ERROR:tm:new_t:
out of mem
Mar 11 11:15:08 opensipsh /usr/local/sbin/opensips[1701]:
ERROR:tm:sip_msg_cloner: no more share memory
Mar 11 11:15:08 opensipsh /usr/local/sbin/opensips[1716]:
ERROR:dialog:dlg_create_dialog: could not add further info to the dialog
Mar 11 11:15:08 opensipsh /usr/local/sbin/opensips[1706]:
ERROR:dialog:dlg_create_dialog: could not add further info to the dialog
Mar 11 11:15:08 opensipsh /usr/local/sbin/opensips[1727]:
ERROR:tm:t_newtran: new_t failed
Mar 11 11:15:08 opensipsh /usr/local/sbin/opensips[1704]:
ERROR:tm:t_newtran: new_t failed
Mar 11 11:15:08 opensipsh /usr/local/sbin/opensips[1700]:
ERROR:load_balancer:do_load_balance: failed to create dialog
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1698]:
ERROR:tm:t_newtran: new_t failed
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1723]:
ERROR:tm:t_newtran: new_t failed
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1725]:
ERROR:core:print_rr_body: too many RR
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1708]:
ERROR:dialog:dlg_add_leg_info: Failed to resize legs array
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1712]:
ERROR:dialog:dlg_add_leg_info: Failed to resize legs array
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1714]:
ERROR:dialog:dlg_create_dialog: could not add further info to the dialog
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1720]:
ERROR:tm:t_newtran: new_t failed
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1721]: ERROR:tm:new_t:
out of mem
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1710]:
ERROR:tm:sip_msg_cloner: no more share memory
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1718]:
ERROR:tm:sip_msg_cloner: no more share memory
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1701]:
ERROR:tm:t_newtran: new_t failed
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1716]:
ERROR:dialog:init_leg_info: dlg_add_leg_info failed
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1706]:
ERROR:dialog:init_leg_info: dlg_add_leg_info failed
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1727]: ERROR:tm:new_t:
out of mem
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1704]: ERROR:tm:new_t:
out of mem
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1700]:
ERROR:dialog:dlg_create_dialog: could not add further info to the dialog
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1698]: ERROR:tm:new_t:
out of mem
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1723]: ERROR:tm:new_t:
out of mem
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1708]:
ERROR:tm:t_newtran: new_t failed
Mar 11 11:15:07 opensipsh /usr/local/sbin/opensips[1712]:
ERROR:tm:t_newtran: new_t failed
Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1701]:
ERROR:tm:t_newtran: new_t failed
Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1716]: ERROR:tm:new_t:
out of mem
Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1706]: ERROR:tm:new_t:
out of mem
Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1727]:
ERROR:dialog:get_routing_info: failed to print route records
Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1725]:
ERROR:dialog:dlg_add_leg_info: Failed to resize legs array
Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1704]:
ERROR:dialog:get_routing_info: failed to print route records
Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1700]:
ERROR:dialog:get_routing_info: failed to print route records
Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1698]:
ERROR:dialog:get_routing_info: failed to print route records
Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1723]:
ERROR:dialog:dlg_add_leg_info: no more shm mem
Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1708]:
ERROR:dialog:dlg_add_leg_info: Failed to resize legs array
Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1712]:
ERROR:dialog:dlg_add_leg_info: Failed to resize legs array
Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1714]:
ERROR:dialog:build_new_dlg: no more shm mem (202)
Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1720]:
ERROR:tm:shm_clone_proxy: no more shm memory
Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1721]:
ERROR:tm:sip_msg_cloner: no more share memory
Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1710]:
ERROR:dialog:dlg_add_leg_info: Failed to resize legs array
Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1718]:
ERROR:core:build_req_buf_from_sip_req: out of pkg memory
Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1701]: ERROR:tm:new_t:
out of mem
Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1716]:
ERROR:tm:sip_msg_cloner: no more share memory
Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1706]:
ERROR:tm:sip_msg_cloner: no more share memory
Mar 11 11:15:04 opensipsh /usr/local/sbin/opensips[1727

[OpenSIPS-Users] Opensips+RTPProxy+two sub NetWork

[aksai.china]

http://opensips-open-sip-server.1449251.n2.nabble.com/file/n6081772/Screenshot.png
 

1. How can I passed the voice from the UA to another by configuration the
opensips.cfg and using the rtpproxy command line ?



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[OpenSIPS-Users] opensips 1.6.4 load balancing performance

[Iulian Macare]

Hello


I have installed OpenSips 1.6.4 on CentOS 5.5 32bit with load balancing &
mysql support ; I want to balance calls to 2 asterisk servers . I am sending
traffic to opensips from 1 x  gnudialer & 1 x vicidial ( so from predictive
dialers ). Situation is like this:


++--+--+---++-+
| id | group_id | dst_uri  | resources | probe_mode |
description |
++--+--+---++-+
|  1 |1 | sip:192.168.254.241:5060 | pstn=300  |  0
| |
|  2 |1 | sip:192.168.254.242:5060 | pstn=300  |  0
| |
++--+--+---++-+

600 channels in total , and I send around 500 channels ; OpenSips drops a
lot of calls; By drop I mean a call that is not sent to one of those 2
asterisk servers that I have.

The code for balancing in this situation is:

if (uri=~"^sip:0[1-9][0-9]*@") {
load_balance("1","pstn");
xlog("sending call $ru to $du\n");
t_relay();
exit;
}

! An important thing to say is that in /var/log/messages I see the specific
number that is sent to 192.168.254.241 for example ; So the parameter
xlog("sending call $ru to $du\n"); works ; The problem is that in logs on
192.168.254.241 that number never arrives in asterisk logs ; In logs of
vicidial & gnudialer I see it like congestion.

If I do something like this:

++--+-+---++-+
| id | group_id | dst_uri | resources | probe_mode | description
|
++--+-+---++-+
|  1 |1 | sip:192.168.254.241| pstn=150   |  0
| |
|  2 |2 | sip:192.168.254.241| pstn=150   |  0
| |
|  3 |1 | sip:192.168.254.242| pstn=150   |  0
| |
|  4 |2 | sip:192.168.254.242| pstn=150   |  0
| |

And I split opensips balancing in 2


if(src_ip==192.168.3.10 )
{
load_balance("1","pstn");
   xlog("sending call to $du\n");
t_relay();
exit;
};

if(src_ip==192.168.3.11 )
{
load_balance("2","pstn");
   xlog("sending call to $du\n");
t_relay();
exit;
};


and by doing this I get the same numbers of channels on opensips ( around
500 channels ) but I am splitting in 2 groups of load balancing; It can
process all the calls.

Another question that I saw is that when I make a single call to opensips
and I involve load balancing in /var/log/messages I get 2 times the same
message .. just like it send 2 time to asterisk server the call .. but on
asterisk I receive only one time.

Mar 10 14:58:47 opensips /usr/local/sbin/opensips[27611]: sending call to
sip:192.168.254.241
Mar 10 14:58:47 opensips /usr/local/sbin/opensips[27611]: sending call to
sip:192.168.254.241


Isn't load balancing fast enough the process the calls made by predictive
dialers, when over 300 channels is sent .. ?  Or I have some mistakes made .
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