Re: [OpenSIPS-Users] Fwd: 2.2.1 crashing

[Bogdan-Andrei Iancu]

Zupperrr...I remember I saw this backtrace before, but without the 
ability to reproduce it.


Do you have OpenSIPS locally compiled or installed from packages ?

Best regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 20.09.2016 20:22, Richard Robson wrote:

just made one for you.

http://pastebin.com/HJkwuTDc

I'd ramped this up to 800cps and the recieving asterisk was struggling 
crashed 5 times and autorestarted before i could ramp down the sipp





On 20/09/2016 08:51, Bogdan-Andrei Iancu wrote:

Hi Richard,

If not too much for you, I would really like to see the backtrace - 
even under OS stress conditions, OpenSIPS should NOT crash.


Best regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 16.09.2016 13:15, Richard Robson wrote:

Hi Bogdan,

It looks like it was the extra logging we were putting in using 
xlog. We'd added them while we were developing the script and left 
them in. this was causing systemd-journald to max out the box and 
would segfault the opensips, which restarted. journald was using 90% 
of the box and opensips 10%


I've taken out the xlogs ( there was about 20 per iteration of the 
script) and using sipp i've tested throughput to 250cps and 2000 
concurrent calls. (enought to kill the asterisk processing the calls 
the other side)


Unfortunately i deleted all the core files as the filled the disk 
up. If you still want one I'll be able to reproduce it as I've still 
got the script. I seem to remember though that it was the 
transaction module that was the culprit


Regards,

Richard



On 14/09/2016 15:28, Bogdan-Andrei Iancu wrote:

Hi Richard,

Have you managed to get a corefile and extract a backtrace ?

Best regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 06.09.2016 17:07, Richard Robson wrote:


If its any help, I can see packets coming in particularly BYEs 
that are not being processes at high call rates and are not 
getting to the point in the script where the rtpengine_delete is 
being triggered. this then causes the number of concurrent calls 
on the RTPengine to grow and fill its allocation of ports. This 
then stops calls being made. and the opensips then crashes.




 Forwarded Message 
Subject:2.2.1 crashing
Date:   Tue, 6 Sep 2016 13:45:11 +0100
From:   Richard Robson 
Organisation:   Greenlight Innovation
To: OpenSIPS users mailling list 



During testing we are ramping up sipp to give ~100 CPS between internal
servers and opensips crashes with a seg fault

sipp  -sn uac 192.168.36.141:5060 -s 441382250180  -r 1 -rp 1000
-recv_timeout 7500 -send_timeout 7500 -d 5000

the sipp calls are being routed to an asterisk server, which is playing
audio
Calls are going via an RTPengine on a different box
everything is on CENTOS 7.2
I'm using the latest git of 2.2.1 on a virtual host 8 cores and 8GB
its OK up to around 60 CPS (350 calls) but ramping up to 100 CPS (750
concurrent calls) causes the segfault.
we are seeing the systemd-journal being the heaviest CPU user, but not
sure if this is unrelated



here is the backtrace:http://pastebin.com/SjaSJx7w

--
Richard Robson
Greenlight Support
01382 843843
supp...@greenlightcrm.com



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--
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Greenlight Support
01382 843843
supp...@greenlightcrm.com





--
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Greenlight Support
01382 843843
supp...@greenlightcrm.com


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Re: [OpenSIPS-Users] drouting

[Bogdan-Andrei Iancu]

Hi Kirill,

The logs says that a rule was found, but now GW (destination) was 
selected to be used.


I see you checked the status of the GWs, but please check the status of 
the carriers also:

http://www.opensips.org/html/docs/modules/2.2.x/drouting.html#id296588

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 20.09.2016 15:44, Kirill Galinurov wrote:

Hello Bogdan. I try drouting() without any parameters.

2016-09-20 14:20 GMT+03:00 Bogdan-Andrei Iancu :

Hello Kirill,

I see the MI command succeeded, right? but the scrip do_routing() fails ?
what are the parameters you pass to do_routing() ?

Best regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com


On 19.09.2016 15:50, Kirill Galinurov wrote:

Hi all I try to use drouting module.

Sep 19 17:18:28 [37245] DBG:drouting:do_routing: using dr group 1,
rule_idx 0, username 79632780099
Sep 19 17:18:28 [37245] DBG:drouting:internal_check_rt: found rgid 1
(rule list 0x7fe60616b800)
Sep 19 17:18:28 [37245] INFO:drouting:do_routing: no destination for
dr group 1, rule_idx 0, username 79632780099
L_INFOdo_routing: No rules matching the URI
In dr_rules
   ruleid | groupid | prefix | timerec | priority | routeid | gwlist |
attrs |  description

+-++-+--+-++---+
   0 | 1   | 7  | |1 | | #cr1   |
 | outgoing calls

dr_carriers
   id | carrierid | gwlist | flags | state | attrs | description
+---++---+---+---+-
   1 | cr1   | gw1;   | 0 | 2 | 0x2   | OUT

dr_gateways
   id | gwid | type |  address  | strip | pri_prefix | attrs |
probe_mode | state | socket  | description

+--+--+---+---++---++---+-+-
   1 | gw1  |1 | sip:out_voip.ru | 0 ||   |
  0 | 0 | udp:XX.XX.XX.XX:5060 | out

opensipsctl fifo dr_gw_status
ID:: gw1 IP=XX.XXX.XXX.XX State=Active

[root@sbc-msk opensips]# opensipsctl fifo dr_number_routing 1 79182220222
Matched Prefix:: 7.
What is wrong?

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Re: [OpenSIPS-Users] Load testing in opensips with async in 2.1 version

[Ramachandran, Agalya (Contractor)]

Thanks for the clarification Bogdan.

Regards,
Agalya

From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu
Sent: Tuesday, September 20, 2016 7:06 AM
To: Ramachandran, Agalya (Contractor) ; Liviu 
Chircu ; OpenSIPS users mailling list 

Subject: Re: [OpenSIPS-Users] Load testing in opensips with async in 2.1 version

Hi Agalya,

With rest_client in async mode, you should see ~ cps rate as without having it 
(considering that the HTTP queries are finishing) - of course, this is valid 
for longer test intervals (much longer than the maximum time to reply for the 
HTTP queries).

In async, the time to query will translate in 0 extra load for OpenSIPS. Of 
course, the processing of a call will be affected by the HTTP time to query, 
but on longer periods of time, the CPS will not be affected (as opensips will 
be able to handle the load).

Regards,


Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
On 19.09.2016 18:10, Ramachandran, Agalya (Contractor) wrote:
Hi team,

Have you performed  load test with  OpenSIPS 2.1 version by placing HTTP 
request as async ?
If so can you share the throughput you have achieved? Because that would be 
bench mark for me as well to achieve the results.


1)  Without HTTP being used at all, I see the below throughput .
 150 calls per second 5000+ calls in full dialog


2)  With HTPP as synchronous mode, it is been blocked till the response is 
received.
As a workaround, I tried to increase the no of children 
parameter to handle many number of calls. Is this OK to do this way?


3)  I just want to know the throughput data you guys have achieved for a 
load test with HTTP being async mode.
   This will be very useful to scale up and compare with my data.

Regards,
Agalya

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Re: [OpenSIPS-Users] FW: Asynchronous operation for REST queries

[Ramachandran, Agalya (Contractor)]

Ok Liviu. I will update my code base and will do test. Thank you so much for 
the update.

Regards,
Agalya

From: Liviu Chircu [mailto:li...@opensips.org]
Sent: Tuesday, September 20, 2016 12:24 PM
To: Ramachandran, Agalya (Contractor) ; 
OpenSIPS users mailling list 
Subject: Re: FW:[OpenSIPS-Users] Asynchronous operation for REST queries


Hi Agalya,

2) has been pushed to master. As soon as we test it up a bit, I will have it 
backported, as it's the complete fix.

Liviu Chircu

OpenSIPS Developer

http://www.opensips-solutions.com
On 20.09.2016 16:57, Ramachandran, Agalya (Contractor) wrote:
Hi Liviu,

It would be great if we know tentative time for the below one's.


1)  Fix that you have committed in dev branch with regard to the 10sec 
delay to 2.1.4 or to any stable release?

2)  Feature addition for "connect_polling_interval" ?

Regards,
Agalya
From: Liviu Chircu [mailto:li...@opensips.org]
Sent: Thursday, September 15, 2016 3:53 AM
To: Ramachandran, Agalya (Contractor) 
; 
OpenSIPS users mailling list 

Subject: Re: FW:[OpenSIPS-Users] Asynchronous operation for REST queries


On 14.09.2016 21:33, Ramachandran, Agalya (Contractor) wrote:
Are you going to add one more parameter "connect_polling_interval" and what 
will be use case of it?
Is it replacement for 10% wait time logic?
Yes.



Also, I observed in the case of async call, it takes ~90 to 100ms to open the 
port itself. Is this expected behavior?
Yes.




Liviu Chircu

OpenSIPS Developer

http://www.opensips-solutions.com

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Re: [OpenSIPS-Users] Fwd: 2.2.1 crashing

[Richard Robson]

  
  
just made one for you.
  
  http://pastebin.com/HJkwuTDc
  
  I'd ramped this up to 800cps and the recieving asterisk was
  struggling crashed 5 times and autorestarted before i could ramp
  down the sipp
  
  
  
  
  On 20/09/2016 08:51, Bogdan-Andrei Iancu wrote:


  
  Hi Richard,

If not too much for you, I would really like to see the
backtrace - even under OS stress conditions, OpenSIPS should NOT
crash. 

Best regards,
  
  Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
  On 16.09.2016 13:15, Richard Robson
wrote:
  
  

Hi Bogdan,
  
  It looks like it was the extra logging we were putting in
  using xlog. We'd added them while we were developing the
  script and left them in. this was causing systemd-journald to
  max out the box and would segfault the opensips, which
  restarted. journald was using 90% of the box and opensips 10%
  
  I've taken out the xlogs ( there was about 20 per iteration of
  the script) and using sipp i've tested throughput to 250cps
  and 2000 concurrent calls. (enought to kill the asterisk
  processing the calls the other side)
  
  Unfortunately i deleted all the core files as the filled the
  disk up. If you still want one I'll be able to reproduce it as
  I've still got the script. I seem to remember though that it
  was the transaction module that was the culprit
  
  Regards,
  
  Richard
   
  
  
  On 14/09/2016 15:28, Bogdan-Andrei Iancu wrote:


  
  Hi Richard,

Have you managed to get a corefile and extract a backtrace ?

Best regards,
  
  Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
  On 06.09.2016 17:07, Richard
Robson wrote:
  
  

If its any help, I can see packets coming in particularly
  BYEs that are not being processes at high call rates and
  are not getting to the point in the script where the
  rtpengine_delete is being triggered. this then causes the
  number of concurrent calls on the RTPengine to grow and
  fill its allocation of ports. This then stops calls being
  made. and the opensips then crashes.


  
   Forwarded Message 
  

  
Subject:

2.2.1 crashing
  
  
Date:

Tue, 6 Sep 2016 13:45:11 +0100
  
  
From:

Richard Robson 
  
  
Organisation:

Greenlight Innovation
  
  
To:

OpenSIPS users mailling list 
  

  
  
  
  During testing we are ramping up sipp to give ~100 CPS between internal 
servers and opensips crashes with a seg fault

sipp  -sn uac 192.168.36.141:5060 -s 441382250180  -r 1 -rp 1000 
-recv_timeout 7500 -send_timeout 7500 -d 5000

the sipp calls are being routed to an asterisk server, which is playing 
audio
Calls are going via an RTPengine on a different box
everything is on CENTOS 7.2
I'm using the latest git of 2.2.1 on a virtual host 8 cores and 8GB
its OK up to around 60 CPS (350 calls) but ramping up to 100 CPS (750 
concurrent calls) causes the segfault.
we are seeing the systemd-journal being the heaviest CPU user, but not 
sure if this is unrelated



here is the backtrace: http://pastebin.com/SjaSJx7w

-- 
Richard Robson
Greenlight Support
01382 843843
supp...@greenlightcrm.com






___
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-- 
Richard Robson
Greenlight Support
01382 843843
supp...@greenlightcrm.com
  
  




-- 
Richard Robson
Greenlight Support
01382 843843
supp...@greenlightcrm.com
  


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Re: [OpenSIPS-Users] Fwd: 2.2.1 crashing

[Richard Robson]

  
  
Hi Bogdan,
  
  I'll try and get that for you tomorrow.
  
  Regards,
  
  Richard
  
  
  On 20/09/2016 08:51, Bogdan-Andrei Iancu wrote:


  
  Hi Richard,

If not too much for you, I would really like to see the
backtrace - even under OS stress conditions, OpenSIPS should NOT
crash. 

Best regards,
  
  Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
  On 16.09.2016 13:15, Richard Robson
wrote:
  
  

Hi Bogdan,
  
  It looks like it was the extra logging we were putting in
  using xlog. We'd added them while we were developing the
  script and left them in. this was causing systemd-journald to
  max out the box and would segfault the opensips, which
  restarted. journald was using 90% of the box and opensips 10%
  
  I've taken out the xlogs ( there was about 20 per iteration of
  the script) and using sipp i've tested throughput to 250cps
  and 2000 concurrent calls. (enought to kill the asterisk
  processing the calls the other side)
  
  Unfortunately i deleted all the core files as the filled the
  disk up. If you still want one I'll be able to reproduce it as
  I've still got the script. I seem to remember though that it
  was the transaction module that was the culprit
  
  Regards,
  
  Richard
   
  
  
  On 14/09/2016 15:28, Bogdan-Andrei Iancu wrote:


  
  Hi Richard,

Have you managed to get a corefile and extract a backtrace ?

Best regards,
  
  Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
  On 06.09.2016 17:07, Richard
Robson wrote:
  
  

If its any help, I can see packets coming in particularly
  BYEs that are not being processes at high call rates and
  are not getting to the point in the script where the
  rtpengine_delete is being triggered. this then causes the
  number of concurrent calls on the RTPengine to grow and
  fill its allocation of ports. This then stops calls being
  made. and the opensips then crashes.


  
   Forwarded Message 
  

  
Subject:

2.2.1 crashing
  
  
Date:

Tue, 6 Sep 2016 13:45:11 +0100
  
  
From:

Richard Robson 
  
  
Organisation:

Greenlight Innovation
  
  
To:

OpenSIPS users mailling list 
  

  
  
  
  During testing we are ramping up sipp to give ~100 CPS between internal 
servers and opensips crashes with a seg fault

sipp  -sn uac 192.168.36.141:5060 -s 441382250180  -r 1 -rp 1000 
-recv_timeout 7500 -send_timeout 7500 -d 5000

the sipp calls are being routed to an asterisk server, which is playing 
audio
Calls are going via an RTPengine on a different box
everything is on CENTOS 7.2
I'm using the latest git of 2.2.1 on a virtual host 8 cores and 8GB
its OK up to around 60 CPS (350 calls) but ramping up to 100 CPS (750 
concurrent calls) causes the segfault.
we are seeing the systemd-journal being the heaviest CPU user, but not 
sure if this is unrelated



here is the backtrace: http://pastebin.com/SjaSJx7w

-- 
Richard Robson
Greenlight Support
01382 843843
supp...@greenlightcrm.com






___
Users mailing list
Users@lists.opensips.org
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-- 
Richard Robson
Greenlight Support
01382 843843
supp...@greenlightcrm.com
  
  




-- 
Richard Robson
Greenlight Support
01382 843843
supp...@greenlightcrm.com
  


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Re: [OpenSIPS-Users] FW: Asynchronous operation for REST queries

[Liviu Chircu]

Hi Agalya,

2) has been pushed to master. Assoon as we test it up a bit, I will have 
it backported, as it's the complete fix.


Liviu Chircu
OpenSIPS Developer
http://www.opensips-solutions.com

On 20.09.2016 16:57, Ramachandran, Agalya (Contractor) wrote:


Hi Liviu,

It would be great if we know tentative time for the below one’s.

1)Fix that you have committed in dev branch with regard to the 10sec 
delay to 2.1.4 or to any stable release?


2)Feature addition for “connect_polling_interval” ?

Regards,

Agalya

*From:*Liviu Chircu [mailto:li...@opensips.org]
*Sent:* Thursday, September 15, 2016 3:53 AM
*To:* Ramachandran, Agalya (Contractor) 
; OpenSIPS users mailling list 


*Subject:* Re: FW:[OpenSIPS-Users] Asynchronous operation for REST queries

On 14.09.2016 21:33, Ramachandran, Agalya (Contractor) wrote:

Are you going to add one more parameter “connect_polling_interval”
and what will be use case of it?

Is it replacement for 10% wait time logic?

Yes.

Also, I observed in the case of async call, it takes ~90 to 100ms
to open the port itself. Is this expected behavior?

Yes.


Liviu Chircu
OpenSIPS Developer
http://www.opensips-solutions.com


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Re: [OpenSIPS-Users] drouting

[Jeff Wilkie]

Sorry ...do_routing(1)

Thanks

Jeff Wilkie
USIP Communications

On Sep 20, 2016 8:45 AM, "Kirill Galinurov"  wrote:

> Hello Bogdan. I try drouting() without any parameters.
>
> 2016-09-20 14:20 GMT+03:00 Bogdan-Andrei Iancu :
> > Hello Kirill,
> >
> > I see the MI command succeeded, right? but the scrip do_routing() fails ?
> > what are the parameters you pass to do_routing() ?
> >
> > Best regards,
> >
> > Bogdan-Andrei Iancu
> > OpenSIPS Founder and Developer
> > http://www.opensips-solutions.com
> >
> >
> > On 19.09.2016 15:50, Kirill Galinurov wrote:
> >>
> >> Hi all I try to use drouting module.
> >>
> >> Sep 19 17:18:28 [37245] DBG:drouting:do_routing: using dr group 1,
> >> rule_idx 0, username 79632780099
> >> Sep 19 17:18:28 [37245] DBG:drouting:internal_check_rt: found rgid 1
> >> (rule list 0x7fe60616b800)
> >> Sep 19 17:18:28 [37245] INFO:drouting:do_routing: no destination for
> >> dr group 1, rule_idx 0, username 79632780099
> >> L_INFOdo_routing: No rules matching the URI
> >> In dr_rules
> >>   ruleid | groupid | prefix | timerec | priority | routeid | gwlist |
> >> attrs |  description
> >>
> >> +-++-+--+-+-
> ---+---+
> >>   0 | 1   | 7  | |1 | | #cr1   |
> >> | outgoing calls
> >>
> >> dr_carriers
> >>   id | carrierid | gwlist | flags | state | attrs | description
> >> +---++---+---+---+-
> >>   1 | cr1   | gw1;   | 0 | 2 | 0x2   | OUT
> >>
> >> dr_gateways
> >>   id | gwid | type |  address  | strip | pri_prefix | attrs |
> >> probe_mode | state | socket  | description
> >>
> >> +--+--+---+---++
> ---++---+-+-
> >>   1 | gw1  |1 | sip:out_voip.ru | 0 ||   |
> >>  0 | 0 | udp:XX.XX.XX.XX:5060 | out
> >>
> >> opensipsctl fifo dr_gw_status
> >> ID:: gw1 IP=XX.XXX.XXX.XX State=Active
> >>
> >> [root@sbc-msk opensips]# opensipsctl fifo dr_number_routing 1
> 79182220222
> >> Matched Prefix:: 7.
> >> What is wrong?
> >>
> >> ___
> >> Users mailing list
> >> Users@lists.opensips.org
> >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >>
> >
>
> ___
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>
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Re: [OpenSIPS-Users] drouting

[Jeff Wilkie]

Use the group 1 you created in the command drouting(1) to send out using
group 1.

Thanks

Jeff Wilkie
USIP Communications

On Sep 20, 2016 8:45 AM, "Kirill Galinurov"  wrote:

> Hello Bogdan. I try drouting() without any parameters.
>
> 2016-09-20 14:20 GMT+03:00 Bogdan-Andrei Iancu :
> > Hello Kirill,
> >
> > I see the MI command succeeded, right? but the scrip do_routing() fails ?
> > what are the parameters you pass to do_routing() ?
> >
> > Best regards,
> >
> > Bogdan-Andrei Iancu
> > OpenSIPS Founder and Developer
> > http://www.opensips-solutions.com
> >
> >
> > On 19.09.2016 15:50, Kirill Galinurov wrote:
> >>
> >> Hi all I try to use drouting module.
> >>
> >> Sep 19 17:18:28 [37245] DBG:drouting:do_routing: using dr group 1,
> >> rule_idx 0, username 79632780099
> >> Sep 19 17:18:28 [37245] DBG:drouting:internal_check_rt: found rgid 1
> >> (rule list 0x7fe60616b800)
> >> Sep 19 17:18:28 [37245] INFO:drouting:do_routing: no destination for
> >> dr group 1, rule_idx 0, username 79632780099
> >> L_INFOdo_routing: No rules matching the URI
> >> In dr_rules
> >>   ruleid | groupid | prefix | timerec | priority | routeid | gwlist |
> >> attrs |  description
> >>
> >> +-++-+--+-+-
> ---+---+
> >>   0 | 1   | 7  | |1 | | #cr1   |
> >> | outgoing calls
> >>
> >> dr_carriers
> >>   id | carrierid | gwlist | flags | state | attrs | description
> >> +---++---+---+---+-
> >>   1 | cr1   | gw1;   | 0 | 2 | 0x2   | OUT
> >>
> >> dr_gateways
> >>   id | gwid | type |  address  | strip | pri_prefix | attrs |
> >> probe_mode | state | socket  | description
> >>
> >> +--+--+---+---++
> ---++---+-+-
> >>   1 | gw1  |1 | sip:out_voip.ru | 0 ||   |
> >>  0 | 0 | udp:XX.XX.XX.XX:5060 | out
> >>
> >> opensipsctl fifo dr_gw_status
> >> ID:: gw1 IP=XX.XXX.XXX.XX State=Active
> >>
> >> [root@sbc-msk opensips]# opensipsctl fifo dr_number_routing 1
> 79182220222
> >> Matched Prefix:: 7.
> >> What is wrong?
> >>
> >> ___
> >> Users mailing list
> >> Users@lists.opensips.org
> >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >>
> >
>
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[OpenSIPS-Users] Dispatcher issues.

[Ahmed Munir]

Hi,

I've currently migrated from opensips 1.6.3 to 1.8.9 and upgraded the
routing script and DB attributes. The issue currently I'm facing on 1.8.9
is the dispatcher module i.e. showing most of the nodes in passive mode
rather than active. However, the nodes are showing passive mode in 1.8.9
were actually showing active state in 1.6.3. My question is, is there any
changes in dispatcher module in 1.8.9? As wen through the documentation of
1.8 opensips, don't see any difference in change.


Below are the module parameter I've set for dispatcher both in 1.8.9 and
1.6.3;

modparam("dispatcher|avpops","db_url","mysql://opensips:opensipsrw@localhost
/opensips")
modparam("dispatcher", "ds_ping_method", "OPTIONS")
modparam("dispatcher", "ds_ping_interval", 10)
modparam("dispatcher", "ds_probing_threshhold", 3)
modparam("dispatcher", "ds_probing_mode", 1)
modparam("dispatcher", "options_reply_codes", "501, 403")
modparam("dispatcher", "ds_ping_from", "sip:pr...@proxy.com")

Setting 'Flag' value to 0  and 'weight' to 1 for each node in dispatcher
table.

Please advise, if I missed out any config.


-- 
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Ahmed Munir Chohan
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Re: [OpenSIPS-Users] FW: Asynchronous operation for REST queries

[Ramachandran, Agalya (Contractor)]

Hi Liviu,

It would be great if we know tentative time for the below one's.


1)  Fix that you have committed in dev branch with regard to the 10sec 
delay to 2.1.4 or to any stable release?

2)  Feature addition for "connect_polling_interval" ?

Regards,
Agalya
From: Liviu Chircu [mailto:li...@opensips.org]
Sent: Thursday, September 15, 2016 3:53 AM
To: Ramachandran, Agalya (Contractor) ; 
OpenSIPS users mailling list 
Subject: Re: FW:[OpenSIPS-Users] Asynchronous operation for REST queries


On 14.09.2016 21:33, Ramachandran, Agalya (Contractor) wrote:
Are you going to add one more parameter "connect_polling_interval" and what 
will be use case of it?
Is it replacement for 10% wait time logic?
Yes.


Also, I observed in the case of async call, it takes ~90 to 100ms to open the 
port itself. Is this expected behavior?
Yes.



Liviu Chircu

OpenSIPS Developer

http://www.opensips-solutions.com
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Re: [OpenSIPS-Users] drouting

[Kirill Galinurov]

Hello Bogdan. I try drouting() without any parameters.

2016-09-20 14:20 GMT+03:00 Bogdan-Andrei Iancu :
> Hello Kirill,
>
> I see the MI command succeeded, right? but the scrip do_routing() fails ?
> what are the parameters you pass to do_routing() ?
>
> Best regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
>
>
> On 19.09.2016 15:50, Kirill Galinurov wrote:
>>
>> Hi all I try to use drouting module.
>>
>> Sep 19 17:18:28 [37245] DBG:drouting:do_routing: using dr group 1,
>> rule_idx 0, username 79632780099
>> Sep 19 17:18:28 [37245] DBG:drouting:internal_check_rt: found rgid 1
>> (rule list 0x7fe60616b800)
>> Sep 19 17:18:28 [37245] INFO:drouting:do_routing: no destination for
>> dr group 1, rule_idx 0, username 79632780099
>> L_INFOdo_routing: No rules matching the URI
>> In dr_rules
>>   ruleid | groupid | prefix | timerec | priority | routeid | gwlist |
>> attrs |  description
>>
>> +-++-+--+-++---+
>>   0 | 1   | 7  | |1 | | #cr1   |
>> | outgoing calls
>>
>> dr_carriers
>>   id | carrierid | gwlist | flags | state | attrs | description
>> +---++---+---+---+-
>>   1 | cr1   | gw1;   | 0 | 2 | 0x2   | OUT
>>
>> dr_gateways
>>   id | gwid | type |  address  | strip | pri_prefix | attrs |
>> probe_mode | state | socket  | description
>>
>> +--+--+---+---++---++---+-+-
>>   1 | gw1  |1 | sip:out_voip.ru | 0 ||   |
>>  0 | 0 | udp:XX.XX.XX.XX:5060 | out
>>
>> opensipsctl fifo dr_gw_status
>> ID:: gw1 IP=XX.XXX.XXX.XX State=Active
>>
>> [root@sbc-msk opensips]# opensipsctl fifo dr_number_routing 1 79182220222
>> Matched Prefix:: 7.
>> What is wrong?
>>
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Re: [OpenSIPS-Users] Routing in opensips

[Bogdan-Andrei Iancu]

He Eric,

In the script (generated by menuconfig), you have an IF block with :
append_hf("P-hint: outbound\r\n");

That block is executed if one of your users is dialing an external SIP 
domain. So, that is the place where the calls will break out to public 
internet.


As your OpenSIPS has a private IP (not routable in internet), you need 
to configure it to advertise the public IP of your NAT via the 
set_advertise_address() - see:

http://www.opensips.org/Documentation/Script-CoreFunctions-2-2#toc46

Put there the public IP of your NAT. Also, be sure you create in your 
router (running the NAT) a port forwarding from the public IP port 5060 
UDP to the private IP of your OpenSIPS port 5060 UDP.


Best regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 15.09.2016 13:10, Eric Freeman wrote:

Yes.

Get Outlook for iOS 


*From:* Bogdan-Andrei Iancu 
*Sent:* Thursday, September 15, 2016 4:57:45 AM
*To:* Eric Freeman; OpenSIPS users mailling list
*Subject:* Re: [OpenSIPS-Users] Routing in opensips
Hello Eric,

So, you OpenSIPS is running behind a NAT ?

Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 14.09.2016 18:43, Eric Freeman wrote:


Thank you. I am using the residential script and my device is 
registered. When I register two devices I can call between them. But, 
when I want to send the traffic out to the Internet I receive these 
types of errors. My device is a 10 address trying to call public IPs 
on the Internet. It appears I need to add a default route in 
opensips? It seems like it can't send the traffic to the Internet or 
a default route. The opensips server itself can get onto the Internet.



Aug 22 15:12:55 cd-ubuntu-opensips /usr/local/sbin/opensips[11973]: 
DBG:core:get_hdr_field


: found end of header

Aug 22 15:12:55 cd-ubuntu-opensips /usr/local/sbin/opensips[11973]: 
DBG:rr:find_first_rout


e: No Route headers found

Aug 22 15:12:55 cd-ubuntu-opensips /usr/local/sbin/opensips[11973]: 
DBG:rr:loose_route: Th


ere is no Route HF

Aug 22 15:12:55 cd-ubuntu-opensips /usr/local/sbin/opensips[11973]: 
DBG:dialog:w_create_di


alog2: bye on timeout activated


Thank you,


Eric Freeman



*From:* Bogdan-Andrei Iancu 
*Sent:* Wednesday, September 14, 2016 10:47:23 AM
*To:* OpenSIPS users mailling list; Eric Freeman
*Subject:* Re: [OpenSIPS-Users] Routing in opensips
Hi Eric,

A good starting point is the residential script that can be generated 
via the menuconfig tool. See:

http://www.opensips.org/Documentation/Generating-Configs-2-2

Then have your unit registered with your OpenSIPS.

Best regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 12.09.2016 18:15, Eric Freeman wrote:


I am very new to opensips and I want to use it for one application. 
I have installed version: opensips 2.2.1 (x86_64/linux)


on Ubuntu 14.04.2 LTS

Release: 14.04


I have a video conference unit and I want to be able make SIP calls 
to other video conference


units over the Internet.


I believe I need to set up some type of routing to route calls from 
this device out


to the Internet. I believe I need to add some commands to the route 
section of the configuration file. Can someone please let me know 
what commands I need to the config to allow the unit to call other 
units via sip over the Internet. I am using the “default routing” 
from the residential script.



I have registered the video conference unit to the opensips server


Contact:: sip:LifeSize@10.89.71.12;transport=UDP Q=


from config

listen=udp:10.88.23.13:5060   # CUSTOMIZE ME


listen=tcp:10.88.23.13:5060   # CUSTOMIZE ME


Please let me know if you need more information.


Thank you,

Eric Freeman



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Re: [OpenSIPS-Users] How to save avp-values to singal attribute name in cache_store?

[Bogdan-Andrei Iancu]

Hello,

The avp_db_query() stores the DB result (multiple rows with multiple 
columns) into a set of AVPs (kind of an array). See

http://www.opensips.org/html/docs/modules/2.2.x/avpops.html#id293973

Using a while loop (see 
http://www.opensips.org/Documentation/Script-Statements-2-2#toc3), you 
can iterate throw all the rows, by checking the if 
$(avp(xxx)[$var(idx)]) is NULL or not (where $var(idx) is the variable 
in the while loop).


Inside the loop, simply store the row (all the AVPs with that index) 
into cache DB .


Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 18.09.2016 12:15, xiaofeng wrote:

Hi there,

I want to save several avp_db_query-values to one attribute use 
cache_store at startup, how to do that?


Thanks.

--
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Re: [OpenSIPS-Users] drouting

[Bogdan-Andrei Iancu]

Hello Kirill,

I see the MI command succeeded, right? but the scrip do_routing() fails 
? what are the parameters you pass to do_routing() ?


Best regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 19.09.2016 15:50, Kirill Galinurov wrote:

Hi all I try to use drouting module.

Sep 19 17:18:28 [37245] DBG:drouting:do_routing: using dr group 1,
rule_idx 0, username 79632780099
Sep 19 17:18:28 [37245] DBG:drouting:internal_check_rt: found rgid 1
(rule list 0x7fe60616b800)
Sep 19 17:18:28 [37245] INFO:drouting:do_routing: no destination for
dr group 1, rule_idx 0, username 79632780099
L_INFOdo_routing: No rules matching the URI
In dr_rules
  ruleid | groupid | prefix | timerec | priority | routeid | gwlist |
attrs |  description
+-++-+--+-++---+
  0 | 1   | 7  | |1 | | #cr1   |
| outgoing calls

dr_carriers
  id | carrierid | gwlist | flags | state | attrs | description
+---++---+---+---+-
  1 | cr1   | gw1;   | 0 | 2 | 0x2   | OUT

dr_gateways
  id | gwid | type |  address  | strip | pri_prefix | attrs |
probe_mode | state | socket  | description
+--+--+---+---++---++---+-+-
  1 | gw1  |1 | sip:out_voip.ru | 0 ||   |
 0 | 0 | udp:XX.XX.XX.XX:5060 | out

opensipsctl fifo dr_gw_status
ID:: gw1 IP=XX.XXX.XXX.XX State=Active

[root@sbc-msk opensips]# opensipsctl fifo dr_number_routing 1 79182220222
Matched Prefix:: 7.
What is wrong?

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Re: [OpenSIPS-Users] Load testing in opensips with async in 2.1 version

[Bogdan-Andrei Iancu]

Hi Agalya,

With rest_client in async mode, you should see ~ cps rate as without 
having it (considering that the HTTP queries are finishing) - of course, 
this is valid for longer test intervals (much longer than the maximum 
time to reply for the HTTP queries).


In async, the time to query will translate in 0 extra load for OpenSIPS. 
Of course, the processing of a call will be affected by the HTTP time to 
query, but on longer periods of time, the CPS will not be affected (as 
opensips will be able to handle the load).


Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 19.09.2016 18:10, Ramachandran, Agalya (Contractor) wrote:


Hi team,

Have you performed  load test with  OpenSIPS 2.1 version by placing 
HTTP request as async ?


If so can you share the throughput you have achieved? Because that 
would be bench mark for me as well to achieve the results.


1)Without HTTP being used at all, I see the below throughput .

 150 calls per second 5000+ calls in full dialog

2)With HTPP as synchronous mode, it is been blocked till the response 
is received.


As a workaround, I tried to increase the no of 
children parameter to handle many number of calls. Is this OK to do 
this way?


3)I just want to know the throughput data you guys have achieved for a 
load test with HTTP being async mode.


   This will be very useful to scale up and compare with 
my data.


Regards,
Agalya



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Re: [OpenSIPS-Users] Fwd: 2.2.1 crashing

[Bogdan-Andrei Iancu]

Hi Richard,

If not too much for you, I would really like to see the backtrace - even 
under OS stress conditions, OpenSIPS should NOT crash.


Best regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 16.09.2016 13:15, Richard Robson wrote:

Hi Bogdan,

It looks like it was the extra logging we were putting in using xlog. 
We'd added them while we were developing the script and left them in. 
this was causing systemd-journald to max out the box and would 
segfault the opensips, which restarted. journald was using 90% of the 
box and opensips 10%


I've taken out the xlogs ( there was about 20 per iteration of the 
script) and using sipp i've tested throughput to 250cps and 2000 
concurrent calls. (enought to kill the asterisk processing the calls 
the other side)


Unfortunately i deleted all the core files as the filled the disk up. 
If you still want one I'll be able to reproduce it as I've still got 
the script. I seem to remember though that it was the transaction 
module that was the culprit


Regards,

Richard



On 14/09/2016 15:28, Bogdan-Andrei Iancu wrote:

Hi Richard,

Have you managed to get a corefile and extract a backtrace ?

Best regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 06.09.2016 17:07, Richard Robson wrote:


If its any help, I can see packets coming in particularly BYEs that 
are not being processes at high call rates and are not getting to 
the point in the script where the rtpengine_delete is being 
triggered. this then causes the number of concurrent calls on the 
RTPengine to grow and fill its allocation of ports. This then stops 
calls being made. and the opensips then crashes.




 Forwarded Message 
Subject:2.2.1 crashing
Date:   Tue, 6 Sep 2016 13:45:11 +0100
From:   Richard Robson 
Organisation:   Greenlight Innovation
To: OpenSIPS users mailling list 



During testing we are ramping up sipp to give ~100 CPS between internal
servers and opensips crashes with a seg fault

sipp  -sn uac 192.168.36.141:5060 -s 441382250180  -r 1 -rp 1000
-recv_timeout 7500 -send_timeout 7500 -d 5000

the sipp calls are being routed to an asterisk server, which is playing
audio
Calls are going via an RTPengine on a different box
everything is on CENTOS 7.2
I'm using the latest git of 2.2.1 on a virtual host 8 cores and 8GB
its OK up to around 60 CPS (350 calls) but ramping up to 100 CPS (750
concurrent calls) causes the segfault.
we are seeing the systemd-journal being the heaviest CPU user, but not
sure if this is unrelated



here is the backtrace:http://pastebin.com/SjaSJx7w

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Re: [OpenSIPS-Users] Oopensips 2.2.1 wss Crash

[Răzvan Crainea]

Hi, Kirill!

My assumption is that this is something related to this bug[1], reported 
by Tito Cumpen. Could you please update your sources to the latest 2.2 
version and let us know if this is still happening?


[1] 
https://github.com/OpenSIPS/opensips/commit/39e40f93fe5bc76c58c5b64f8b32b327b975f62e


Best regards,

Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com

On 09/20/2016 07:35 AM, Kirill Galinurov wrote:

Hi,All I testing wss in Opensips 2.2.1. Sometimes it's crashed

  19 17:46:00 [32693] INFO:core:init_sock_keepalive: TCP keepalive
enabled on socket 24
Sep 19 17:46:00 [32693] DBG:core:print_ip: tcpconn_new: new tcp
connection to: 194.186.5.238
Sep 19 17:46:00 [32693] DBG:core:tcpconn_new: on port 35888, proto 6
Sep 19 17:46:00 [32693] DBG:proto_wss:tls_conn_init: entered: Creating
a whole new ssl connection
Sep 19 17:46:00 [32693] DBG:proto_wss:tls_conn_init: looking up socket
based TLS server domain [92.53.102.226:8443]
Sep 19 17:46:00 [32693] DBG:tls_mgm:tls_find_server_domain: virtual
TLS server domain not found, Using default TLS server domain settin
gs
Sep 19 17:46:00 [32693] DBG:proto_wss:tls_conn_init: found socket
based TLS server domain [0.0.0.0:0]
Sep 19 17:46:00 [32693] DBG:proto_wss:tls_conn_init: Setting in ACCEPT
mode (server)
Sep 19 17:46:00 [32693] DBG:core:tcpconn_add: hashes: 754, 98
Sep 19 17:46:00 [32693] DBG:core:handle_new_connect: new connection:
0x7f4b8ab6cab0 24 flags: 0006
Sep 19 17:46:00 [32693] DBG:core:send2child: no free tcp receiver,
connection passed to the least busy one (proc #1, 1 con)
Sep 19 17:46:00 [32693] DBG:core:send2child: to tcp child 1 0(32692),
0x7f4b8ab6cab0 rw 1
Sep 19 17:46:00 [32692] DBG:core:handle_io: We have received conn
0x7f4b8ab6cab0 with rw 1 on fd 25
Sep 19 17:46:00 [32692] DBG:core:io_watch_add: [TCP_worker]
io_watch_add op (25 on 5) (0x7e2fc0, 25, 19, 0x7f4b8ab6cab0,1),
fd_no=3/102
4
Sep 19 17:46:00 [32692] DBG:proto_wss:tls_update_fd: New fd is 25
Sep 19 17:46:00 [32693] DBG:core:io_wait_loop_epoll: [TCP_main]
EPOLLHUP on IN ->connection closed by the remote peer!
Sep 19 17:46:00 [32693] DBG:core:handle_tcp_worker: dead tcp worker 1
(pid 32692, no 0) (shutting down?)
Sep 19 17:46:00 [32693] DBG:core:io_watch_del: [TCP_main] io_watch_del
op on index 10 18 (0x7e2fc0, 18, 10, 0x0,0x1) fd_no=11 called
Sep 19 17:46:00 [32693] DBG:core:io_wait_loop_epoll: [TCP_main]
EPOLLHUP on IN ->connection closed by the remote peer!
Sep 19 17:46:00 [32693] CRITICAL:core:receive_fd: EOF on 21
Sep 19 17:46:00 [32693] DBG:core:handle_worker: dead child 6, pid
32692 (shutting down?)
Sep 19 17:46:00 [32693] DBG:core:io_watch_del: [TCP_main] io_watch_del
op on index 8 21 (0x7e2fc0, 21, 8, 0x0,0x1) fd_no=10 called
Sep 19 17:46:00 [32686] DBG:core:handle_sigs: status = 139
Sep 19 17:46:00 [32686] INFO:core:handle_sigs: child process 32692
exited by a signal 11
Sep 19 17:46:00 [32686] INFO:core:handle_sigs: core was generated
Sep 19 17:46:00 [32686] INFO:core:handle_sigs: terminating due to SIGCHLD
Sep 19 17:46:00 [32688] INFO:core:sig_usr: signal 15 received
Sep 19 17:46:00 [32687] INFO:core:sig_usr: signal 15 received
Sep 19 17:46:00 [32690] INFO:core:sig_usr: signal 15 received
Sep 19 17:46:00 [32691] INFO:core:sig_usr: signal 15 received
Sep 19 17:46:00 [32689] INFO:core:sig_usr: signal 15 received
Sep 19 17:46:00 [32693] INFO:core:sig_usr: signal 15 received
Sep 19 17:46:00 [32686] INFO:core:cleanup: cleanup
Sep 19 17:46:00 [32686] DBG:uac_auth:mod_destroy: done
Sep 19 17:46:00 [32686] DBG:core:pool_remove: removing connection from the pool
Sep 19 17:46:00 [32686] DBG:db_postgres:db_postgres_free_connection:
PQfinish(0xb8df30)
Sep 19 17:46:00 [32686] DBG:db_postgres:db_postgres_free_connection:
pkg_free(0x7f4b8c805150)
Sep 19 17:46:00 [32686] DBG:tls_mgm:mod_destroy: entered
Sep 19 17:46:00 [32686] DBG:tm:tm_shutdown: tm_shutdown : start
Sep 19 17:46:00 [32686] DBG:tm:unlink_timer_lists: emptying DELETE
list for set 0
Sep 19 17:46:00 [32686] DBG:tm:tm_shutdown: emptying hash table
Sep 19 17:46:00 [32686] DBG:tm:tm_shutdown: releasing timers
Sep 19 17:46:00 [32686] DBG:tm:tm_shutdown: removing semaphores
Sep 19 17:46:00 [32686] DBG:tm:tm_shutdown: destroying callback lists
Sep 19 17:46:00 [32686] DBG:tm:tm_shutdown: tm_shutdown : done
Sep 19 17:46:00 [32686] DBG:proto_wss:tls_update_fd: New fd is 24
Sep 19 17:46:00 [32686] ERROR:proto_wss:tls_write: TLS connection to
62.183.124.129:32224 write failed
Sep 19 17:46:00 [32686] ERROR:proto_wss:tls_write: TLS write error:
Sep 19 17:46:00 [32686] ERROR:proto_wss:tls_blocking_write: TLS failed
to send data
Sep 19 17:46:00 [32686] DBG:proto_wss:tls_conn_clean: entered
Sep 19 17:46:00 [32686] DBG:proto_wss:tls_update_fd: New fd is 22
Sep 19 17:47:00 [32686] CRITICAL:core:sig_alarm_abort: BUG - shutdown
timeout triggered, dying...Aborted (core dumped)


(gdb) bt full
#0  0x7f4b88c28669 in ?? ()
No symbol table info available.
#1  0x0001 in ?? ()
No symbol table info available.
#2