Re: [OpenSIPS-Users] presence xpidf

[jarrod]

Ben,

I had a similar issue using the older Polycom OS and BLF using the directory 
and buddy watch (xpidf+xml).

However, after upgrading the Polycom’s and using the newer 
attendant.resourceList method for BLF, I found that they use the standard 
dialoginfo supported by presence_dialoginfo module.

Jarrod

> On Jan 26, 2016, at 9:59 AM, Newlin, Ben  wrote:
> 
> I know I have used presence extensively with Polycom phones using pidf+xml. I 
> know they support it. Maybe there is some setting in your model specifying 
> the remote server type? If that is set to Microsoft Lync the Polycom may be 
> sending xpidf for compatibility.
> 
> But Polycom phones absolutely support pidf+xml. We implemented a presence 
> feature last year using only Polycom and SIP SIMPLE.
> 
> Ben Newlin
> 
> From:  > on behalf of Stas Kobzar 
> mailto:stas.kob...@modulis.ca>>
> Reply-To: OpenSIPS users mailling list  >
> Date: Monday, January 25, 2016 at 4:17 PM
> To: Bogdan-Andrei Iancu mailto:bog...@opensips.org>>
> Cc: OpenSIPS users mailling list  >
> Subject: Re: [OpenSIPS-Users] presence xpidf
> 
> Hi Bogdan,
> 
> It looks like this document. I have searched in Polycom documentations and 
> community forum and I can not find any mention of the document.
> So I posted the question on Polycom community forum hoping someone can give 
> an answer.
> 
> In Asterisk source code for chan_sip, they have a comment which says: "Early 
> pre-RFC 3863 format with MSN additions (Microsoft Messenger)" 
> (https://github.com/sipwise/asterisk/blob/master/channels/chan_sip.c#L308 
> )
> 
> 
> If Polycom follows Microsoft Lync then they probably stick to their 
> documentation:
> https://msdn.microsoft.com/en-us/library/cc246218.aspx 
> 
> 
> If I have any answer from Poycom forum I send it to you.
> 
> Thank you,
> Stas
> 
> On Mon, Jan 25, 2016 at 5:57 AM, Bogdan-Andrei Iancu  > wrote:
> Hi Stas,
> 
> Is this the actual draft :
> 
> http://www.cs.columbia.edu/sip/drafts/impp/draft-rosenberg-impp-pidf-00.txt 
> 
> 
> ?
> 
> Best regards,
>  Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com 
> On 23.01.2016 17:12, Stas Kobzar wrote:
>> Hi Bogdan,
>> 
>> I can confirm that Polycom still use xpidf with the latest firmware ver. 5. 
>> 
>> I think Digium phones also use xpidf. Can not say for other vendors. 
>> Looks like xpidf is supported by Asterisk, FreeSWITCH. 
>> Quick google search makes me believe that pjsip, reSIProcate and linphone 
>> (osip) have xpidf support.
>> 
>> You are right, the draft is quite old, and I can not find any update. 
>> But xpidf is still there.
>> 
>> Thank you,
>> Stas
>> 
>> 
>> On Fri, Jan 22, 2016 at 8:19 AM, Bogdan-Andrei Iancu < 
>> bog...@opensips.org 
>> > wrote:
>> Hi Stas,
>> 
>> While looking around for this xpidf I found this:
>> http://opensips.org/pipermail/users/2010-April/012336.html 
>> 
>> 
>> So, what is the story with this xpdif ? is it still in use ? was it replaced 
>> by pidf+xml ? as I see it died as draft.
>> 
>> Regards,
>>  Bogdan-Andrei Iancu
>> OpenSIPS Founder and Developer
>> http://www.opensips-solutions.com 
>> On 13.01.2016 18:47, Stas Kobzar wrote:
>>> Hi Bogdan,
>>> 
>>> I do not think the DOCTYPE is the problem here. What I see is that when I 
>>> use MI to publish this application/xpidf doc, OpenSIPS does not want to 
>>> parse the document, and if I understand correct, this is because this type 
>>> of document does not have  XML branch.
>>> 
>>> You are right, about end-to-end, and if I configure OpenSIPS just to relay 
>>> SUBSCRIBE/NOTIFY, it should work fine.
>>> But I want use OpenSIPS to be in the middle because I have a logic in my 
>>> application when it is me who change the status (for example with 
>>> web-interface)
>>> 
>>> So basically my question is, is it going to be supported by OpenSIPS 
>>> (application/xpidf)? Or as you mentioned, it is basically the work for UA 
>>> and it is not supposed to be in OpenSIPS?
>>> 
>>> Thank you,
>>> 
>>> 
>>> On Wed, Jan 13, 2016 at 7:56 AM, Bogdan-Andrei Iancu < 
>>> bog...@opensips.org 
>>> > wrote:
>>> Hi Stas,
>>> 
>>> You say you see the DOCTYPE line in NOTIFY packets and this is supported by 
>>> OpenSIPS ?
>>> 
>>> Now, on Polycom extension - if it is something end-2-end, it means it does 
>>> not require a presence server and everything should be between end points 
>>> by using SUBSCRIBE and NOTI

Re: [OpenSIPS-Users] presence xpidf

[Stas Kobzar]

Hello Ben,

Thank you for the hint with Lync. I did check our settings and we do not
have any Lync related configuration. It looks like default settings do not
have any special lync stuff.

The Polycom phones do support PIDF with BLF/call pick. The event is dialog.
Meanwhile for directory, when "buddy watch" is enabled, Polycom is sending
event "presence" with content type XPIDF.

Stas

On Tue, Jan 26, 2016 at 10:59 AM, Newlin, Ben  wrote:

> I know I have used presence extensively with Polycom phones using
> pidf+xml. I know they support it. Maybe there is some setting in your model
> specifying the remote server type? If that is set to Microsoft Lync the
> Polycom may be sending xpidf for compatibility.
>
> But Polycom phones absolutely support pidf+xml. We implemented a presence
> feature last year using only Polycom and SIP SIMPLE.
>
> Ben Newlin
>
> From:  on behalf of Stas Kobzar <
> stas.kob...@modulis.ca>
> Reply-To: OpenSIPS users mailling list 
> Date: Monday, January 25, 2016 at 4:17 PM
> To: Bogdan-Andrei Iancu 
> Cc: OpenSIPS users mailling list 
> Subject: Re: [OpenSIPS-Users] presence xpidf
>
> Hi Bogdan,
>
> It looks like this document. I have searched in Polycom documentations and
> community forum and I can not find any mention of the document.
> So I posted the question on Polycom community forum hoping someone can
> give an answer.
>
> In Asterisk source code for chan_sip, they have a comment which says: "Early
> pre-RFC 3863 format with MSN additions (Microsoft Messenger)" (
> https://github.com/sipwise/asterisk/blob/master/channels/chan_sip.c#L308)
>
> If Polycom follows Microsoft Lync then they probably stick to their
> documentation: https://msdn.microsoft.com/en-us/library/cc246218.aspx
>
> If I have any answer from Poycom forum I send it to you.
>
> Thank you,
> Stas
>
> On Mon, Jan 25, 2016 at 5:57 AM, Bogdan-Andrei Iancu 
> wrote:
>
>> Hi Stas,
>>
>> Is this the actual draft :
>>
>> http://www.cs.columbia.edu/sip/drafts/impp/draft-rosenberg-impp-pidf-00.txt
>>
>> ?
>>
>> Best regards,
>>
>> Bogdan-Andrei Iancu
>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>>
>> On 23.01.2016 17:12, Stas Kobzar wrote:
>>
>> Hi Bogdan,
>>
>> I can confirm that Polycom still use xpidf with the latest firmware ver.
>> 5.
>>
>> I think Digium phones also use xpidf. Can not say for other vendors.
>> Looks like xpidf is supported by Asterisk, FreeSWITCH.
>> Quick google search makes me believe that pjsip, reSIProcate and linphone
>> (osip) have xpidf support.
>>
>> You are right, the draft is quite old, and I can not find any update.
>> But xpidf is still there.
>>
>> Thank you,
>> Stas
>>
>>
>> On Fri, Jan 22, 2016 at 8:19 AM, Bogdan-Andrei Iancu <
>> bog...@opensips.org> wrote:
>>
>>> Hi Stas,
>>>
>>> While looking around for this xpidf I found this:
>>> http://opensips.org/pipermail/users/2010-April/012336.html
>>>
>>> So, what is the story with this xpdif ? is it still in use ? was it
>>> replaced by pidf+xml ? as I see it died as draft.
>>>
>>> Regards,
>>>
>>> Bogdan-Andrei Iancu
>>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>>>
>>> On 13.01.2016 18:47, Stas Kobzar wrote:
>>>
>>> Hi Bogdan,
>>>
>>> I do not think the DOCTYPE is the problem here. What I see is that when
>>> I use MI to publish this application/xpidf doc, OpenSIPS does not want to
>>> parse the document, and if I understand correct, this is because this type
>>> of document does not have  XML branch.
>>>
>>> You are right, about end-to-end, and if I configure OpenSIPS just to
>>> relay SUBSCRIBE/NOTIFY, it should work fine.
>>> But I want use OpenSIPS to be in the middle because I have a logic in my
>>> application when it is me who change the status (for example with
>>> web-interface)
>>>
>>> So basically my question is, is it going to be supported by OpenSIPS
>>> (application/xpidf)? Or as you mentioned, it is basically the work for UA
>>> and it is not supposed to be in OpenSIPS?
>>>
>>> Thank you,
>>>
>>>
>>> On Wed, Jan 13, 2016 at 7:56 AM, Bogdan-Andrei Iancu <
>>> bog...@opensips.org> wrote:
>>>
 Hi Stas,

 You say you see the DOCTYPE line in NOTIFY packets and this is
 supported by OpenSIPS ?

 Now, on Polycom extension - if it is something end-2-end, it means it
 does not require a presence server and everything should be between end
 points by using SUBSCRIBE and NOTIFY (no PUBLISH, as this is specific to
 the presence agent/server model). Am I wrong with this ?

 Best regards,

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

 On 12.01.2016 17:10, Stas Kobzar wrote:

 Hello Bogdan,

 Thank you for your response.
 The DOCTYPE within XML is seems to be Microsoft presence format:
 https://msdn.microsoft.com/en-ca/library/cc246193.aspx

 I am not sure if it can be used with PUBLISH though. For now I saw it
 only in NOTIFY packets.

 P

Re: [OpenSIPS-Users] potential memory leak warning - uac_auth:build_authorization_hdr

[Colin Martin]

Bogdan,

That’s brilliant. Thanks very much.

Colin

> On 26 Jan 2016, at 16:45, Bogdan-Andrei Iancu  wrote:
> 
> Hi Colin,
> 
> The issue should be fixed on GIT repo, on all maintained versions.
> 
> Thanks and Best regards,
> 
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
> 
> On 26.01.2016 11:41, Colin Martin wrote:
>> Hi Bogdan,
>> 
>> It's the uac_auth module, coupled with the uac module.
>> 
>> It's simply using the uac_auth() function to add the authentication response 
>> into the request, and that's the point where the warning occurs.
>> 
>> Thanks,
>> 
>> Colin
>> 
>> 
>>> On 26 Jan 2016, at 09:25, Bogdan-Andrei Iancu  wrote:
>>> 
>>> Hi Colin,
>>> 
>>> Which of the in which module do you use the uac authentication feature :
>>>uac_auth
>>>b2b_entities
>>>uac_registrant
>>> 
>>> ?
>>> 
>>> Regards,
>>> 
>>> Bogdan-Andrei Iancu
>>> OpenSIPS Founder and Developer
>>> http://www.opensips-solutions.com
>>> 
 On 26.01.2016 11:22, Colin Martin wrote:
 Just one extra bit of information, this doesn't happen on the first call 
 post-restart, but does occur for each subsequent call.
 
 I'm happy to ignore it if it won't affect stability, but I don't know how 
 to verify that!
 
 Colin
 
> On 25 Jan 2016, at 18:25, Colin Martin  wrote:
> 
> Hi,
> 
> I’m working with OpenSIPS 2.1.2 and have started to see the following 
> error message in the logs:
> 
> WARNING:uac_auth:build_authorization_hdr: potential memory leak at addr: 
> 0x7fc266b471f8
> 
> It seems to occur when using a TCP transport.
> 
> Is this a known bug? Is there anything I can do to capture more 
> information to help track it down?
> 
> Thanks,
> 
> Colin
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 ___
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Re: [OpenSIPS-Users] presence xpidf

[Stas Kobzar]

Hello Jarrod,

Thanks for pointing that. That's true what you are saying for
attendant.resourceList parameter. We also use it for BLF and call pickup.
XPIDF is used by Polycom directory presence.

It looks like XPIDF is the content type Polycom uses for P2P presence.
So, if I configure OpenSIPS to simply relay SUBSCRIBE of one Polycom to
another, it works great. Then I just need to take care that NOTIFY packets
from one Polycom can reach another Polycom
and I can set with phone status like "On lunch". Nice colorful icons are
changed on the phone's screen.

What I want, is to change it on the Polycom phone with OpenSIPS MI. And
this is where I had some trouble.

I do not insist that OpenSIPS must support it as I want it. I am just
looking around if this is possible.

Stas


On Tue, Jan 26, 2016 at 11:12 AM,  wrote:

> I had a similar issue using the older Polycom OS and BLF using the
> directory and buddy watch (xpidf+xml) that Freeswitch supports really well.
>
> However, after upgrading the Polycom’s and using the
> newer attendant.resourceList method for BLF, I found that they use the
> standard dialoginfo supported by OpenSIPS’ presence_dialoginfo module.
>
> Jarrod
>
> On Jan 26, 2016, at 9:59 AM, Newlin, Ben  wrote:
>
> I know I have used presence extensively with Polycom phones using
> pidf+xml. I know they support it. Maybe there is some setting in your model
> specifying the remote server type? If that is set to Microsoft Lync the
> Polycom may be sending xpidf for compatibility.
>
> But Polycom phones absolutely support pidf+xml. We implemented a presence
> feature last year using only Polycom and SIP SIMPLE.
>
> Ben Newlin
>
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>


-- 

Stas Kobzar

Developeur VoIP / VoIP Developer


Modulis­.ca Inc.

# Bureau / Office: 514-284-2020 x 246

Email: s tas.kob...@modulis.ca

https://www.modulis.com
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[OpenSIPS-Users] retransmissions after 1xx response

[Юрий Насида]

Hi there!

I use opensip 1.7.2 and noted retransmissions of INVITEs in 450ms despite
the fact I got 100 Trying from my external carrier.
FYI:
1) it's is from opensips exactly (not from UA).
2) Opensips sends absolutely same INVITEs (same cseg, tags, etc).

I can be wrong but RCF says:
"After receiving a 1xx response, any retransmissions cease altogether, and
the client waits for further responses."

I also a bit wondering why retransmissions happens after 450ms (not 500ms
according defult T1_timer). Should I use modparam("tm", "own_timer_proc",
1) ?

Its also strange that Opensips ignore 18X messages which come to
retransmissions INVITE. It happens till 6 sec (when fr_timer exceeded)

Please advice.
Thanks.
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[OpenSIPS-Users] OpenSIPS at FOSDEM'16

[Răzvan Crainea]

Hello, everybody!

This year we'll be again present at the biggest open-source conference, 
FOSDEM'16[1].


This year, me and Liviu we'll be showing you how to enhance your VoIP 
capabilities by using it as an Edge Proxy/SBC[2]. Looking forward to 
seeing you there!


[1] https://fosdem.org/2016/
[2] https://fosdem.org/2016/schedule/event/opensips/

Best regards,

--
Răzvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com

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[OpenSIPS-Users] 1.x LTS upgrade path

[Jock McKechnie]

Greetings all;

We have several hundred 1.8.8 OpenSIPS proxies across our corporate
networks which are running Debian Wheezy. I apparently haven't built a
new one in a while as sometime between then and now the 1.8 LTS has
become deprecated (although I can't find any eMails or notices on the
website actually announcing  it... I knew it would be coming
eventually).

I was planning on moving up to 1.11 but I've struck an issue - 1.11's
Debian Wheezy packages were built with Wheezy/testing and all require
libc6 >= 2.14. Wheezy/stable's libc6 version is 2.13-38. I really,
really don't want to have to move 200+ VMs to Wheezy/testing, let
alone the "running 'testing' in a corporate environment" question.

So I guess I've got two questions:
For now, are the 1.8.8 Debian packages hiding somewhere that I can
pull and store on a local repo so I can use them until I can get the
upgrade path figured out? (Possibly a Jessie upgrade, but that doesn't
excite me any better)

And is there any chance that 1.11 will be rebuilt on the _correct_
Wheezy release? Or is there actually a real dependancy that requires
libc6 2.14+ baked into 1.11 somehow?

Any (reasonable) suggestions are welcome, and as always, thank you very much.

 - Jock

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Re: [OpenSIPS-Users] potential memory leak warning - uac_auth:build_authorization_hdr

[Bogdan-Andrei Iancu]

Hi Colin,

The issue should be fixed on GIT repo, on all maintained versions.

Thanks and Best regards,

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

On 26.01.2016 11:41, Colin Martin wrote:

Hi Bogdan,

It's the uac_auth module, coupled with the uac module.

It's simply using the uac_auth() function to add the authentication response 
into the request, and that's the point where the warning occurs.

Thanks,

Colin



On 26 Jan 2016, at 09:25, Bogdan-Andrei Iancu  wrote:

Hi Colin,

Which of the in which module do you use the uac authentication feature :
uac_auth
b2b_entities
uac_registrant

?

Regards,

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


On 26.01.2016 11:22, Colin Martin wrote:
Just one extra bit of information, this doesn't happen on the first call 
post-restart, but does occur for each subsequent call.

I'm happy to ignore it if it won't affect stability, but I don't know how to 
verify that!

Colin


On 25 Jan 2016, at 18:25, Colin Martin  wrote:

Hi,

I’m working with OpenSIPS 2.1.2 and have started to see the following error 
message in the logs:

WARNING:uac_auth:build_authorization_hdr: potential memory leak at addr: 
0x7fc266b471f8

It seems to occur when using a TCP transport.

Is this a known bug? Is there anything I can do to capture more information to 
help track it down?

Thanks,

Colin
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Re: [OpenSIPS-Users] presence xpidf

[jarrod]

I had a similar issue using the older Polycom OS and BLF using the directory 
and buddy watch (xpidf+xml) that Freeswitch supports really well.

However, after upgrading the Polycom’s and using the newer 
attendant.resourceList method for BLF, I found that they use the standard 
dialoginfo supported by OpenSIPS’ presence_dialoginfo module.

Jarrod

> On Jan 26, 2016, at 9:59 AM, Newlin, Ben  wrote:
> 
> I know I have used presence extensively with Polycom phones using pidf+xml. I 
> know they support it. Maybe there is some setting in your model specifying 
> the remote server type? If that is set to Microsoft Lync the Polycom may be 
> sending xpidf for compatibility.
> 
> But Polycom phones absolutely support pidf+xml. We implemented a presence 
> feature last year using only Polycom and SIP SIMPLE.
> 
> Ben Newlin
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Re: [OpenSIPS-Users] presence xpidf

[Newlin, Ben]

I know I have used presence extensively with Polycom phones using pidf+xml. I 
know they support it. Maybe there is some setting in your model specifying the 
remote server type? If that is set to Microsoft Lync the Polycom may be sending 
xpidf for compatibility.

But Polycom phones absolutely support pidf+xml. We implemented a presence 
feature last year using only Polycom and SIP SIMPLE.

Ben Newlin

From: 
mailto:users-boun...@lists.opensips.org>> on 
behalf of Stas Kobzar mailto:stas.kob...@modulis.ca>>
Reply-To: OpenSIPS users mailling list 
mailto:users@lists.opensips.org>>
Date: Monday, January 25, 2016 at 4:17 PM
To: Bogdan-Andrei Iancu mailto:bog...@opensips.org>>
Cc: OpenSIPS users mailling list 
mailto:users@lists.opensips.org>>
Subject: Re: [OpenSIPS-Users] presence xpidf

Hi Bogdan,

It looks like this document. I have searched in Polycom documentations and 
community forum and I can not find any mention of the document.
So I posted the question on Polycom community forum hoping someone can give an 
answer.

In Asterisk source code for chan_sip, they have a comment which says: "Early 
pre-RFC 3863 format with MSN additions (Microsoft Messenger)" 
(https://github.com/sipwise/asterisk/blob/master/channels/chan_sip.c#L308)

If Polycom follows Microsoft Lync then they probably stick to their 
documentation: https://msdn.microsoft.com/en-us/library/cc246218.aspx

If I have any answer from Poycom forum I send it to you.

Thank you,
Stas

On Mon, Jan 25, 2016 at 5:57 AM, Bogdan-Andrei Iancu 
mailto:bog...@opensips.org>> wrote:
Hi Stas,

Is this the actual draft :
http://www.cs.columbia.edu/sip/drafts/impp/draft-rosenberg-impp-pidf-00.txt

?

Best regards,

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

On 23.01.2016 17:12, Stas Kobzar wrote:
Hi Bogdan,

I can confirm that Polycom still use xpidf with the latest firmware ver. 5.

I think Digium phones also use xpidf. Can not say for other vendors.
Looks like xpidf is supported by Asterisk, FreeSWITCH.
Quick google search makes me believe that pjsip, reSIProcate and linphone 
(osip) have xpidf support.

You are right, the draft is quite old, and I can not find any update.
But xpidf is still there.

Thank you,
Stas


On Fri, Jan 22, 2016 at 8:19 AM, Bogdan-Andrei Iancu 
<bog...@opensips.org> 
wrote:
Hi Stas,

While looking around for this xpidf I found this:
http://opensips.org/pipermail/users/2010-April/012336.html

So, what is the story with this xpdif ? is it still in use ? was it replaced by 
pidf+xml ? as I see it died as draft.

Regards,

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

On 13.01.2016 18:47, Stas Kobzar wrote:
Hi Bogdan,

I do not think the DOCTYPE is the problem here. What I see is that when I use 
MI to publish this application/xpidf doc, OpenSIPS does not want to parse the 
document, and if I understand correct, this is because this type of document 
does not have  XML branch.

You are right, about end-to-end, and if I configure OpenSIPS just to relay 
SUBSCRIBE/NOTIFY, it should work fine.
But I want use OpenSIPS to be in the middle because I have a logic in my 
application when it is me who change the status (for example with web-interface)

So basically my question is, is it going to be supported by OpenSIPS 
(application/xpidf)? Or as you mentioned, it is basically the work for UA and 
it is not supposed to be in OpenSIPS?

Thank you,


On Wed, Jan 13, 2016 at 7:56 AM, Bogdan-Andrei Iancu 
<bog...@opensips.org> 
wrote:
Hi Stas,

You say you see the DOCTYPE line in NOTIFY packets and this is supported by 
OpenSIPS ?

Now, on Polycom extension - if it is something end-2-end, it means it does not 
require a presence server and everything should be between end points by using 
SUBSCRIBE and NOTIFY (no PUBLISH, as this is specific to the presence 
agent/server model). Am I wrong with this ?

Best regards,

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

On 12.01.2016 17:10, Stas Kobzar wrote:
Hello Bogdan,

Thank you for your response.
The DOCTYPE within XML is seems to be Microsoft presence format:
https://msdn.microsoft.com/en-ca/library/cc246193.aspx

I am not sure if it can be used with PUBLISH though. For now I saw it only in 
NOTIFY packets.

Polycom UA is using this type of presence for end-to-end presence between 
phones.
I would like to publish this with MI to change presence status on Polycom 
phones.

Thank you,
Stas







On Mon, Jan 11, 2016 at 4:59 AM, Bogdan-Andrei Iancu 
<bog...@opensips.org> 
wrote:
Hi Stas,

I checked with couple of SIP UACs and I found none using the "DOCTYPE" line the 
published presence XML. So, I guess you should simply drop such a line in your 
testing.

The "tuple" node is replacing your "atom" node (at

Re: [OpenSIPS-Users] tomcat external app listening to OpenSIPS events

[Bogdan-Andrei Iancu]

Hi Julian,

Could you test please this fix made by John:
https://github.com/OpenSIPS/opensips/commit/be43fcdb7696c6ee53673b351380fe701c209a44

If it works ok, we will do the backport to the stable releases too.

Best regards,

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

On 26.01.2016 01:45, Julian Kay wrote:

thanks, I appreciate that!!!

Juls

-Original Message-
From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu
Sent: Tuesday, January 19, 2016 6:47 AM
To: OpenSIPS users mailling list; juli...@vazycomm.com
Subject: Re: [OpenSIPS-Users] tomcat external app listening to OpenSIPS events

Hi Julian,

I see your problem and I agree with you - the path should be be missing in the 
HTTP URL. I will open a bug report and have it fixed asap.

Thanks and regards,

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

On 14.01.2016 00:28, Julian Kay wrote:

HI,

I really appreciate you taking the time!!

yes OpenSIPS is connecting to Tomcat, the problem I see (I think) is when the 
event is being raised OpenSIPS is NOT sending the complete URI. Tomcat server 
returns an error of 404.
the Tomcat logs seem to indicate  OpenSIPS is only be sending
192.168.3.132:8080 -> when it should be sending
192.168.3.132:8080/ccurbiz/xmlrpc

Thanks!!
Juls

-Original Message-
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei
Iancu
Sent: Wednesday, January 13, 2016 5:44 AM
To: OpenSIPS users mailling list; juli...@vazycomm.com
Subject: Re: [OpenSIPS-Users] tomcat external app listening to
OpenSIPS events

Hi Julian,

So, you say OpenSIPS is actually connecting via HTTP to tomcat in order to 
deliver the event, right ?

Regards,

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

On 13.01.2016 03:02, Julian Kay wrote:

Hi;

Thanks for the help!

the event subscription seems to be successful as  show from the
output of a test xmlrpc test client
Event:: E_SIP_MESSAGE id=11
	Subscriber::
socket=xmlrpc:192.168.3.132:8080:ccradius/xmlrpc/OpenSIPSInterface.ms
g tests expire=never


the parameters I'm using for the for subscribe_event:
subscribe_event("E_SIP_MESSAGE","xmlrpc:192.168.3.132:8080:ccradius/x
m
lrpc/OpenSIPSInterface.msgtests");

for testing purposes I raise the event with ->
raise_event("E_SIP_MESSAGE")

and it seems OpenSIPs attempts to deliver the event because this is
what I see in the Tomcat logs:  192.168.3.167 - -
[06/Jan/2016:23:21:19 -0500] "POST /RPC2 HTTP/1.1" 404 959

I'm able to successfully call the Java class from a test xmlrpc test client.

I would really appreciate ideas how I can resolve this as this is critical for 
me. I'm even willing to consider some other methodology as long as I can 
interface with Tomcat.

Thanks for your input!!
Juls

-Original Message-
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei
Iancu
Sent: Tuesday, January 12, 2016 4:14 AM
To: OpenSIPS users mailling list; juli...@vazycomm.com
Subject: Re: [OpenSIPS-Users] tomcat external app listening to
OpenSIPS events

Hi Julian,

First check if your event subscription was successful (and still valid).
Use the "subscriber_list" MI command :
http://www.opensips.org/Documentation/Interface-CoreMI-1-11#toc18

After that, when events happens, check at network level if there is any attempt 
from opensips side to deliver the event via XMLRPC to the indicated URL.

Regards,

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

On 12.01.2016 05:15, Julian Kay wrote:

Hi;

Has anyone successfully interfaced OpenSIPS event_xmlrpc with a Tombat xmlrpc 
server servlet? If you yes can you share any tips, because I've been trying it 
for a while without any success.

Thanks for any help!!

Juls

-Original Message-
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Julian Kay
Sent: Thursday, January 07, 2016 9:33 AM
To: 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] external app listening to OpenSIPS/SIP
events

Thanks for your help!!

yes the event is reaching the xmlrpc server I'm including 2 entries in the 
Tomcat log file:

call using xmlrpc-test-tool:192.168.3.167 - - [06/Jan/2016:16:26:11 -0500] "POST 
/ccurbiz/xmlrpc/RPC2 HTTP/1.0" 200 156

call using event_xmlrpc:192.168.3.167 - - [06/Jan/2016:23:21:19 -0500] "POST 
/RPC2 HTTP/1.1" 404 959

(192.168.3.167 is the OpenSIPs server)

I've tried a few variations for the host name parameter including:
subscribe_event("E_SIP_MESSAGE","xmlrpc:192.168.3.132:8080:ccurbiz.x
m
l
rpc.OpenSIPSInterface.msgtests");

subscribe_event("E_SIP_MESSAGE","xmlrpc:192.168.3.132:8080/ccurbiz/xmlrpc:OpenSIPSInterface.msgtests");
 -> this is the most similar to other xmlrpc clients I've tested.

Best regards

Re: [OpenSIPS-Users] external app listening to OpenSIPS/SIP events

[Ionut Ionita]

Hi,

Fixed it (at least tried to) with this commit 
https://github.com/OpenSIPS/opensips/commit/be43fcdb7696c6ee53673b351380fe701c209a44. 
Can you please test it an tell me if everything works
correctly? If no path provided '/RPC2' shall be used as before, but if 
you do provide a path
(like '/ccurbiz/xmlrpc' in 'xmlrpc:192.168.3.132:8080/ccurbiz/xmlrpc') 
the one you provided shall be used.


Regards,
Ionut Ionita

On 01/07/2016 04:32 PM, Julian Kay wrote:

Thanks for your help!!

yes the event is reaching the xmlrpc server I'm including 2 entries in the 
Tomcat log file:

call using xmlrpc-test-tool:192.168.3.167 - - [06/Jan/2016:16:26:11 -0500] "POST 
/ccurbiz/xmlrpc/RPC2 HTTP/1.0" 200 156

call using event_xmlrpc:192.168.3.167 - - [06/Jan/2016:23:21:19 -0500] "POST 
/RPC2 HTTP/1.1" 404 959

(192.168.3.167 is the OpenSIPs server)

I've tried a few variations for the host name parameter including:
subscribe_event("E_SIP_MESSAGE","xmlrpc:192.168.3.132:8080:ccurbiz.xmlrpc.OpenSIPSInterface.msgtests");

subscribe_event("E_SIP_MESSAGE","xmlrpc:192.168.3.132:8080/ccurbiz/xmlrpc:OpenSIPSInterface.msgtests");
 -> this is the most similar to other xmlrpc clients I've tested.

Best regards!
Juls


-Original Message-
From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Razvan Crainea
Sent: Thursday, January 07, 2016 4:34 AM
To: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] external app listening to OpenSIPS/SIP events

Hi, Julian!

First of all, is the event reaching the xmlrpc server? If not sure, try to make 
a tcpdump to capture the communication.
If it does reach the server, what are the errors you are seeing in the Tomcat 
server?

Best regards,

Răzvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com

On 01/07/2016 12:32 AM, Julian Kay wrote:

THANKS!!

_raising the event in the script got rid of the error._

But OpenSIPS is not calling the method I'm testing on a Tomcat XMLRPC
serverlet. This is my actual event subscription code I'm using:

subscribe_event("E_SIP_MESSAGE","xmlrpc:192.168.3.132:8080:ccurbiz.xml
rpc.OpenSIPSInterface.msgtests");

I've tried several (many) syntax variations.

On the Tomcat server it always points to the root RPC2 and returns the
error code 404.

ccurbiz -> is my  java project name

xmlrpc -> is the servelet mapping to the Java class OpenSIPSInterface

msgtest -> is the method I'm trying to call

I wanted to make sure the xmlrpc server was working, I'm able to
successfully call the method with xmlrpc-test-tool

Any suggestion to be able to make this work with Tomcat server are
greatly appreciated!!

Thanks!

Juls

*From:*users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] *On Behalf Of *Razvan
Crainea
*Sent:* Wednesday, December 30, 2015 10:40 AM
*To:* users@lists.opensips.org
*Subject:* Re: [OpenSIPS-Users] external app listening to OpenSIPS/SIP
events

Hi, Julian!

Make sure you are raising that event from your script (i.e.
raise_event("E_SIP_MESSAGE")).

http://www.opensips.org/Documentation/Tutorials-EventInterface#toc9

Best regards,
Răzvan

On 12/24/2015 03:39 AM, Julian Kay wrote:

 Thanks for the input, but for now I need to work with XMLRPC.

 is it possible to create a custom event? I've been trying to use in
 the startup_route: subscribe_event("E_SIP_MESSAGE",
 "xmlrpc:http://192.168.3.201:/msgtests";)

 I get the error: ERROR: core:evi_event_subscribe: invalid event name
 

 Any help much appreciated, thx!!!

 Juls

 *From:*users-boun...@lists.opensips.org
 
 [mailto:users-boun...@lists.opensips.org] *On Behalf Of *Tito Cumpen
 *Sent:* Wednesday, December 16, 2015 6:19 PM
 *To:* OpenSIPS users mailling list
 *Subject:* Re: [OpenSIPS-Users] external app listening to
 OpenSIPS/SIP events

 Julian,

 Look into using
 http://www.opensips.org/html/docs/modules/devel/event_rabbitmq This
 module allows you to spin up an event based at any moment in the
 script and pass variables in the process(Meaning it is a
 publisher.). Your app can subscribe as reader of the queue and do
 whatever task needs to be done. You can go as far as using this CDR
 as well.

 Goodluck,

 Tito

 On Wed, Dec 16, 2015 at 6:14 PM, Julian Kay mailto:juli...@vazycomm.com>> wrote:

 Hi;

 I'm looking for some guidance or confirmation that I'm on the right
 path. If I want an external app to be able to listen to SIP events
 like SIP event 180 (ringing), is registering a custom event with
 OpenSIPS MI and then use "raise_event" to fire a custom event a good
 way to get my external app to listen to OpenSIPS events?

 Thanks!!

 Juls


 ___
 Users mailing list
 Users@lists.opensips.org 
 http://lists.opensips.org/cgi-b

Re: [OpenSIPS-Users] WARNING:core:utimer_ticker: utimer task already schedualed

[Bogdan-Andrei Iancu]

Hi Dragomir,

What RADIUS problem do you still have in 2.1 ?

Regards,

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

On 26.01.2016 10:17, Dragomir Haralambiev wrote:

Thanks Bogdan,

Why this problem exists only in Opensips 2.1?

Best regards,
Dragomir

2016-01-26 9:59 GMT+02:00 Bogdan-Andrei Iancu >:


So after all, the problem was so slow/blocking communication with
the Radius server. For the future, to debug such issue you can use
the exec_msg_threshold to see what are the slow parts of your script:
http://www.opensips.org/Documentation/Script-CoreParameters-2-1#toc57

Regards,

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

On 26.01.2016 00:18, Aqs Younas wrote:

Resolved the issue by down grading to radius 2.2. But stuck in
another problem going to post another thread.

Thank you All.

On 25 January 2016 at 19:55, Aqs Younas mailto:aqsyou...@gmail.com>> wrote:

We are still looking where things went wrong. Actually code
is not changed a bit , same radius configuration in
production server works perfect. We just copied the setup to
new server and facing the problem. I see a lot of people
posted this issue before but none have shared the solutions.

Could you help where we need to look for radius connections.

Much thanks for your pointers

On 25 January 2016 at 19:18, Bogdan-Andrei Iancu
mailto:bog...@opensips.org>> wrote:

Hi Aqs,

I assume after fixing your RADIUS issue the timer
warnings disappeared   ?

Regards,

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

On 22.01.2016 23:09, Aqs Younas wrote:

Sorry, I must search archive before query here.

The problem is related to radius accounting. I am too
using radius for AAA and this problem appears after
calls began to terminate.
Below threads are purely related.

http://comments.gmane.org/gmane.comp.voip.opensips.user/31415
https://www.mail-archive.com/users@lists.opensips.org/msg30675.html

On 22 January 2016 at 22:37, Aqs Younas
mailto:aqsyou...@gmail.com>> wrote:

On the start of test. Using top


[...]




I see opensips stuck(flood with warnings) durings
calls termination process which leads me to manually
kill opensisp to prevent opensips from eating all my
server resources.

Test is performed using sipp with 30 cps and 2000
concurrent calls of during 6 minutes.

:lscpu

Architecture: x86_64
CPU op-mode(s): 32-bit, 64-bit
Byte Order: Little Endian
CPU(s): 32
On-line CPU(s) list:   0-31
Thread(s) per core:2
Core(s) per socket:8
Socket(s): 2
NUMA node(s): 2
Vendor ID: GenuineIntel
CPU family: 6
Model: 45
Stepping: 7
CPU MHz: 1199.531
BogoMIPS: 4001.49
Virtualization: VT-x
L1d cache: 32K
L1i cache: 32K
L2 cache: 256K
L3 cache: 20480K
NUMA node0 CPU(s): 0-7,16-23
NUMA node1 CPU(s): 8-15,24-31

egrep --color 'Mem|Cache|Swap' /proc/meminfo

MemTotal: 49530560 kB
MemFree: 29677688 kB
MemAvailable: 45436220 kB



Let me know if you need anything else.

Thanks.

On 22 January 2016 at 21:18, Bogdan-Andrei Iancu
mailto:bog...@opensips.org>>
wrote:

Yes, the "children" option - 10 should be more
than ok.

What is the CPU usage from opensips during the
test ?

Regards,

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

On 22.01.2016 18:08, Aqs Younas wrote:

Hi, Bogdan

You mean children? First I thought it is due to children(10 
default) which I increased to 500 but no avail.
This I have in my configuration file.

debug=3
log_stderror=no
log_facility=LOG_LOCAL3

fork=yes
children=500
open_files_limit=9

Thanks for replying.

On

Re: [OpenSIPS-Users] presence xpidf

[Stas Kobzar]

Hi Bogdan,

It looks like this document. I have searched in Polycom documentations and
community forum and I can not find any mention of the document.
So I posted the question on Polycom community forum hoping someone can give
an answer.

In Asterisk source code for chan_sip, they have a comment which says: "Early
pre-RFC 3863 format with MSN additions (Microsoft Messenger)" (
https://github.com/sipwise/asterisk/blob/master/channels/chan_sip.c#L308)

If Polycom follows Microsoft Lync then they probably stick to their
documentation: https://msdn.microsoft.com/en-us/library/cc246218.aspx

If I have any answer from Poycom forum I send it to you.

Thank you,
Stas

On Mon, Jan 25, 2016 at 5:57 AM, Bogdan-Andrei Iancu 
wrote:

> Hi Stas,
>
> Is this the actual draft :
>
> http://www.cs.columbia.edu/sip/drafts/impp/draft-rosenberg-impp-pidf-00.txt
>
> ?
>
> Best regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 23.01.2016 17:12, Stas Kobzar wrote:
>
> Hi Bogdan,
>
> I can confirm that Polycom still use xpidf with the latest firmware ver.
> 5.
>
> I think Digium phones also use xpidf. Can not say for other vendors.
> Looks like xpidf is supported by Asterisk, FreeSWITCH.
> Quick google search makes me believe that pjsip, reSIProcate and linphone
> (osip) have xpidf support.
>
> You are right, the draft is quite old, and I can not find any update.
> But xpidf is still there.
>
> Thank you,
> Stas
>
>
> On Fri, Jan 22, 2016 at 8:19 AM, Bogdan-Andrei Iancu <
> bog...@opensips.org> wrote:
>
>> Hi Stas,
>>
>> While looking around for this xpidf I found this:
>> http://opensips.org/pipermail/users/2010-April/012336.html
>>
>> So, what is the story with this xpdif ? is it still in use ? was it
>> replaced by pidf+xml ? as I see it died as draft.
>>
>> Regards,
>>
>> Bogdan-Andrei Iancu
>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>>
>> On 13.01.2016 18:47, Stas Kobzar wrote:
>>
>> Hi Bogdan,
>>
>> I do not think the DOCTYPE is the problem here. What I see is that when I
>> use MI to publish this application/xpidf doc, OpenSIPS does not want to
>> parse the document, and if I understand correct, this is because this type
>> of document does not have  XML branch.
>>
>> You are right, about end-to-end, and if I configure OpenSIPS just to
>> relay SUBSCRIBE/NOTIFY, it should work fine.
>> But I want use OpenSIPS to be in the middle because I have a logic in my
>> application when it is me who change the status (for example with
>> web-interface)
>>
>> So basically my question is, is it going to be supported by OpenSIPS
>> (application/xpidf)? Or as you mentioned, it is basically the work for UA
>> and it is not supposed to be in OpenSIPS?
>>
>> Thank you,
>>
>>
>> On Wed, Jan 13, 2016 at 7:56 AM, Bogdan-Andrei Iancu <
>> bog...@opensips.org> wrote:
>>
>>> Hi Stas,
>>>
>>> You say you see the DOCTYPE line in NOTIFY packets and this is supported
>>> by OpenSIPS ?
>>>
>>> Now, on Polycom extension - if it is something end-2-end, it means it
>>> does not require a presence server and everything should be between end
>>> points by using SUBSCRIBE and NOTIFY (no PUBLISH, as this is specific to
>>> the presence agent/server model). Am I wrong with this ?
>>>
>>> Best regards,
>>>
>>> Bogdan-Andrei Iancu
>>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>>>
>>> On 12.01.2016 17:10, Stas Kobzar wrote:
>>>
>>> Hello Bogdan,
>>>
>>> Thank you for your response.
>>> The DOCTYPE within XML is seems to be Microsoft presence format:
>>> https://msdn.microsoft.com/en-ca/library/cc246193.aspx
>>>
>>> I am not sure if it can be used with PUBLISH though. For now I saw it
>>> only in NOTIFY packets.
>>>
>>> Polycom UA is using this type of presence for end-to-end presence
>>> between phones.
>>> I would like to publish this with MI to change presence status on
>>> Polycom phones.
>>>
>>> Thank you,
>>> Stas
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> On Mon, Jan 11, 2016 at 4:59 AM, Bogdan-Andrei Iancu <
>>> bog...@opensips.org> wrote:
>>>
 Hi Stas,

 I checked with couple of SIP UACs and I found none using the "DOCTYPE"
 line the published presence XML. So, I guess you should simply drop such a
 line in your testing.

 The "tuple" node is replacing your "atom" node (at least this is what I
 noticed while trying other UACs). Here is an example of a PUBLISH xml
 generated by Zoiper:

 
 >>> 
 "sip:bog...@opensips.org;transport=UDP"
 >
 
 open
 
 Busy
 
 

 In regards to the crash, even if the XML is not properly formated, it
 should not crash - can you send me the actual MI command + content to try
 to reproduce the crash and have it fixed ?

 Best regards,

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

 On 04.01.2016 18:49, Stas Kobzar w

Re: [OpenSIPS-Users] rabbit event not passing params 2.2

[Tito Cumpen]

Hey Razvan,

This is still an issue with the latest dev build. The event is entirely
empty when it is transmitted to the queue. I've tried
modparam("event_rabbitmq", "sync_mode", 1) but it does not make a
difference.

THanks,
Tito

On Mon, Jan 25, 2016 at 3:39 PM, Tito Cumpen  wrote:

> Hey Razvan,
>
> This is still an issue with the latest dev build. The event is entirely
> empty when it is transmitted to the queue. I've tried
> modparam("event_rabbitmq", "sync_mode", 1) but it does not make a
> difference.
>
> THanks,
> Tito
>
> On Fri, Oct 30, 2015 at 1:20 PM, Răzvan Crainea 
> wrote:
>
>> Hi, Tito!
>>
>> Apologies for getting back so late. The only addition to 2.2 was the
>> async support. Have you tried setting the sync_mode to 1
>> modparam("event_rabbitmq", "sync_mode", 1)
>>
>> Best regards,
>>
>> Răzvan Crainea
>> OpenSIPS Solutionswww.opensips-solutions.com
>>
>> On 10/20/2015 08:15 PM, Tito Cumpen wrote:
>>
>> Razvan,
>>
>>
>> I went back to version: opensips 2.1.1 and found that the events are
>> being raised with params as expected. Meaning all is working. Any
>> possibility it broke sometime in 2.2-dev?
>>
>>
>> Thanks,
>> Tito
>>
>> On Tue, Oct 13, 2015 at 5:56 PM, Tito Cumpen  wrote:
>>
>>> Razvan,
>>>
>>>
>>> Yes, There is no body in the event. The event name is as well. Would you
>>> like me to send you a trace ?
>>>
>>> Thanks,
>>> Tito
>>>
>>> On Tue, Oct 13, 2015 at 5:54 PM, Răzvan Crainea < 
>>> raz...@opensips.org> wrote:
>>>
 Hi, Tito!

 So you do get the event, but not the parameters? Is there any body in
 the event?

 Best regards,

 Răzvan Crainea
 OpenSIPS Core Developer
 http://www.opensips-solutions.com

 On 10/13/2015 07:52 PM, Tito Cumpen wrote:

> Răzvan,
>
> Are you referring to
> http://www.opensips.org/html/docs/modules/2.2.x/event_rabbitmq
> section 1.3?
>  From section 2.1 it looks like the subscription syntax looks the same.
>
> Mine is as I sent it before
>
>  subscribe_event("UL_AOR_INSERT",
> "rabbitmq:myrabbitserver/sip1dev");
>
>
> I think the issue lies when appending params as everything except the
> params are sent to the queue. I am appending params in a wrapper like
> this
>
>
> raise_event("UL_AOR_DELETE", $avp(param));
>
>
>
>
>
> On Mon, Oct 12, 2015 at 8:27 PM, Răzvan Crainea  > wrote:
>
> Hi, Tito!
>
> So you're saying that the exact logic works in 1.11, but not in
> 2.2?
> Starting from 2.1 the socket syntax was changed a bit, to be able
> to
> specify both the routing key and the exchange used. This was not
> possible in 1.11. Are you sure you are specifying both?
>
> Best regards,
>
> Răzvan Crainea
> OpenSIPS Core Developer
> http://www.opensips-solutions.com
>
> On 10/12/2015 10:09 PM, Tito Cumpen wrote:
>
> Any idea what has broken in the current dev version ?
>
>
> Thanks,
> Tito
>
> On Thu, Oct 1, 2015 at 9:17 PM, Tito Cumpen  
> >> wrote:
>
>  Razvan,
>
>
>  The connection looks fine check this output from the logs
>
>
>  Oct  2 01:06:34 cloud-server-06 /sbin/opensips[15024]:
>  DBG:core:evi_param_set: adding string param
>
>  Oct  2 01:06:34 cloud-server-06 /sbin/opensips[15024]:
>  DBG:core:evi_param_set: adding string param
>
>  Oct  2 01:06:34 cloud-server-06 /sbin/opensips[15024]:
>  DBG:core:evi_param_set: adding string param
>
>  Oct  2 01:06:34 cloud-server-06 /sbin/opensips[15024]:
>  DBG:core:evi_param_set: adding string param
>
>  Oct  2 01:06:34 cloud-server-06 /sbin/opensips[15024]:
>  DBG:core:evi_param_set: adding int param
>
>  Oct  2 01:06:34 cloud-server-06 /sbin/opensips[15024]:
>  DBG:core:destroy_avp_list: destroying list (nil)
>
>  Oct  2 01:06:34 cloud-server-06 /sbin/opensips[15024]:
>  DBG:core:evi_param_set: adding string param
>
>  Oct  2 01:06:34 cloud-server-06 /sbin/opensips[15024]:
>  DBG:core:evi_param_set: adding string param
>
>  Oct  2 01:06:34 cloud-server-06 /sbin/opensips[15024]:
>  DBG:core:evi_param_set: adding string param
>
>  Oct  2 01:06:34 cloud-server-06 /sbin/opensips[15024]:
>  DBG:core:evi_param_set: adding string param
>
>  Oct  2 01:06:34 cloud-server-06 /sbin/opensips[15024]:
>  DBG:core:evi

Re: [OpenSIPS-Users] rabbit event not passing params 2.2

[Tito Cumpen]

Hey Razvan,

This is still an issue with the latest dev build. The event is entirely
empty when it is transmitted to the queue. I've tried
modparam("event_rabbitmq", "sync_mode", 1) but it does not make a
difference.

THanks,
Tito

On Fri, Oct 30, 2015 at 1:20 PM, Răzvan Crainea  wrote:

> Hi, Tito!
>
> Apologies for getting back so late. The only addition to 2.2 was the async
> support. Have you tried setting the sync_mode to 1
> modparam("event_rabbitmq", "sync_mode", 1)
>
> Best regards,
>
> Răzvan Crainea
> OpenSIPS Solutionswww.opensips-solutions.com
>
> On 10/20/2015 08:15 PM, Tito Cumpen wrote:
>
> Razvan,
>
>
> I went back to version: opensips 2.1.1 and found that the events are being
> raised with params as expected. Meaning all is working. Any possibility it
> broke sometime in 2.2-dev?
>
>
> Thanks,
> Tito
>
> On Tue, Oct 13, 2015 at 5:56 PM, Tito Cumpen  wrote:
>
>> Razvan,
>>
>>
>> Yes, There is no body in the event. The event name is as well. Would you
>> like me to send you a trace ?
>>
>> Thanks,
>> Tito
>>
>> On Tue, Oct 13, 2015 at 5:54 PM, Răzvan Crainea < 
>> raz...@opensips.org> wrote:
>>
>>> Hi, Tito!
>>>
>>> So you do get the event, but not the parameters? Is there any body in
>>> the event?
>>>
>>> Best regards,
>>>
>>> Răzvan Crainea
>>> OpenSIPS Core Developer
>>> http://www.opensips-solutions.com
>>>
>>> On 10/13/2015 07:52 PM, Tito Cumpen wrote:
>>>
 Răzvan,

 Are you referring to
 http://www.opensips.org/html/docs/modules/2.2.x/event_rabbitmq
 section 1.3?
  From section 2.1 it looks like the subscription syntax looks the same.

 Mine is as I sent it before

  subscribe_event("UL_AOR_INSERT",
 "rabbitmq:myrabbitserver/sip1dev");


 I think the issue lies when appending params as everything except the
 params are sent to the queue. I am appending params in a wrapper like
 this


 raise_event("UL_AOR_DELETE", $avp(param));





 On Mon, Oct 12, 2015 at 8:27 PM, Răzvan Crainea >>> > wrote:

 Hi, Tito!

 So you're saying that the exact logic works in 1.11, but not in 2.2?
 Starting from 2.1 the socket syntax was changed a bit, to be able to
 specify both the routing key and the exchange used. This was not
 possible in 1.11. Are you sure you are specifying both?

 Best regards,

 Răzvan Crainea
 OpenSIPS Core Developer
 http://www.opensips-solutions.com

 On 10/12/2015 10:09 PM, Tito Cumpen wrote:

 Any idea what has broken in the current dev version ?


 Thanks,
 Tito

 On Thu, Oct 1, 2015 at 9:17 PM, Tito Cumpen >>> 
 >> wrote:

  Razvan,


  The connection looks fine check this output from the logs


  Oct  2 01:06:34 cloud-server-06 /sbin/opensips[15024]:
  DBG:core:evi_param_set: adding string param

  Oct  2 01:06:34 cloud-server-06 /sbin/opensips[15024]:
  DBG:core:evi_param_set: adding string param

  Oct  2 01:06:34 cloud-server-06 /sbin/opensips[15024]:
  DBG:core:evi_param_set: adding string param

  Oct  2 01:06:34 cloud-server-06 /sbin/opensips[15024]:
  DBG:core:evi_param_set: adding string param

  Oct  2 01:06:34 cloud-server-06 /sbin/opensips[15024]:
  DBG:core:evi_param_set: adding int param

  Oct  2 01:06:34 cloud-server-06 /sbin/opensips[15024]:
  DBG:core:destroy_avp_list: destroying list (nil)

  Oct  2 01:06:34 cloud-server-06 /sbin/opensips[15024]:
  DBG:core:evi_param_set: adding string param

  Oct  2 01:06:34 cloud-server-06 /sbin/opensips[15024]:
  DBG:core:evi_param_set: adding string param

  Oct  2 01:06:34 cloud-server-06 /sbin/opensips[15024]:
  DBG:core:evi_param_set: adding string param

  Oct  2 01:06:34 cloud-server-06 /sbin/opensips[15024]:
  DBG:core:evi_param_set: adding string param

  Oct  2 01:06:34 cloud-server-06 /sbin/opensips[15024]:
  DBG:core:evi_param_set: adding int param

  Oct  2 01:06:34 cloud-server-06 /sbin/opensips[15024]:
  DBG:core:destroy_avp_list: destroying list (nil)

  Oct  2 01:06:34 cloud-server-06 /sbin/opensips[15024]:
  DBG:core:evi_param_set: adding string param

  Oct  2 01:06:34 cloud-server-06 /sbin/opensips[15024]:
  DBG:core:evi_param_set: adding string param

>>

Re: [OpenSIPS-Users] add_diversion counter variable

[Bogdan-Andrei Iancu]

Hi Edwin,

Yes, the add_diversion() does not support variables in parameters.

Still, you can simply add the header via append_hf() (or insert_hf()), 
by building the whole header by hand - these functions do support 
variables. And the syntax of the Diversion hdr is trivial.


You can also open a feature request on the github tracker.

Regards,

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

On 26.01.2016 13:26, Edwin wrote:

I'm using the latest opensips release 2.1.2 on Debian and have a problem with
the add_diversion header.

For every callforward I have to add a diversion header at top level with the
counter increased by one. I can put the counter number in manual, but not
use a variable:

add_diversion("unknown;privacy=off;screen=yes","sip:+1234...@sip.provider.org;user=phone","1");
[OK]

add_diversion("unknown;privacy=off;screen=yes","sip:+1234...@sip.provider.org;user=phone","$var(dic)");
[FAIL]

This is my chunck of code:

if($di)
{
   $var(dic) = $(hdr(Diversion){param.value,counter}{s.int});
}
else {
   $var(dic) = 0;
}

add_diversion("unknown;privacy=off;screen=yes","sip:+1234...@sip.provider.org;user=phone","$var(dic)");

Jan 26 opensips[]: ERROR:core:fixup_uint: bad number <$var(dic)>
Jan 26 opensips[]: ERROR:core:fix_actions: fixing failed (code=-6) at
/etc/opensips/opensips.cfg:842
Jan 26 opensips[]: ERROR:core:main: failed to fix configuration with err
code -6

I also tried other options (with and without quotes etc. It would be nice to
use a variable for the counter or have a function that automaticly increases
the counter with one.



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[OpenSIPS-Users] add_diversion counter variable

[Edwin]

I'm using the latest opensips release 2.1.2 on Debian and have a problem with
the add_diversion header.

For every callforward I have to add a diversion header at top level with the
counter increased by one. I can put the counter number in manual, but not
use a variable:

add_diversion("unknown;privacy=off;screen=yes","sip:+1234...@sip.provider.org;user=phone","1");
[OK]

add_diversion("unknown;privacy=off;screen=yes","sip:+1234...@sip.provider.org;user=phone","$var(dic)");
[FAIL]

This is my chunck of code:

if($di)
{
  $var(dic) = $(hdr(Diversion){param.value,counter}{s.int});
}
else {
  $var(dic) = 0;
}

add_diversion("unknown;privacy=off;screen=yes","sip:+1234...@sip.provider.org;user=phone","$var(dic)");

Jan 26 opensips[]: ERROR:core:fixup_uint: bad number <$var(dic)>
Jan 26 opensips[]: ERROR:core:fix_actions: fixing failed (code=-6) at
/etc/opensips/opensips.cfg:842
Jan 26 opensips[]: ERROR:core:main: failed to fix configuration with err
code -6

I also tried other options (with and without quotes etc. It would be nice to
use a variable for the counter or have a function that automaticly increases
the counter with one.



--
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http://opensips-open-sip-server.1449251.n2.nabble.com/add-diversion-counter-variable-tp7601042.html
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[OpenSIPS-Users] [OpenSIPS Public Meeting] Scripting variables in OpenSIPS 3.x

[Liviu Chircu]

Hello all,

The upcoming public meeting will be held on IRC (#opensips on FreeNode), 
Wednesday, 27th of January 2016, at 15:00 [1].


The discussions will be based on finding solutions / making improvements 
to the way different data types are both stored in and retrieved from 
scripting variables.


Here is a list of the current issues (as we see them):
- AVPs cannot hold NULL values
- confusion and zero flexibility with regards to variable scoping ($var 
is script-scoped; $avp is transaction-scoped; $dlg_val is dialog-scoped; 
$shv is global-scoped) - this often leads to all sort of hacks / 
auxiliary variables / workarounds in order to have the data processed 
the way it's meant to be

- confusion between NULL, int and string values a variable may hold
- $json variables cannot be passed as route parameters

Along with our own initial proposal of improving on these issues below 
[2] comes your valuable feedback and suggestions on this matter, so do 
not hesitate to join the meeting!


[1]: 
http://www.timeanddate.com/worldclock/fixedtime.html?msg=OpenSIPS+Public+Meeting&iso=20160127T15&p1=1440&ah=1

[2]: http://www.opensips.org/Community/IRCmeeting20160127

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


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Re: [OpenSIPS-Users] potential memory leak warning - uac_auth:build_authorization_hdr

[Colin Martin]

Hi Bogdan,

It's the uac_auth module, coupled with the uac module.

It's simply using the uac_auth() function to add the authentication response 
into the request, and that's the point where the warning occurs.

Thanks,

Colin


> On 26 Jan 2016, at 09:25, Bogdan-Andrei Iancu  wrote:
> 
> Hi Colin,
> 
> Which of the in which module do you use the uac authentication feature :
>uac_auth
>b2b_entities
>uac_registrant
> 
> ?
> 
> Regards,
> 
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
> 
>> On 26.01.2016 11:22, Colin Martin wrote:
>> Just one extra bit of information, this doesn't happen on the first call 
>> post-restart, but does occur for each subsequent call.
>> 
>> I'm happy to ignore it if it won't affect stability, but I don't know how to 
>> verify that!
>> 
>> Colin
>> 
>>> On 25 Jan 2016, at 18:25, Colin Martin  wrote:
>>> 
>>> Hi,
>>> 
>>> I’m working with OpenSIPS 2.1.2 and have started to see the following error 
>>> message in the logs:
>>> 
>>> WARNING:uac_auth:build_authorization_hdr: potential memory leak at addr: 
>>> 0x7fc266b471f8
>>> 
>>> It seems to occur when using a TCP transport.
>>> 
>>> Is this a known bug? Is there anything I can do to capture more information 
>>> to help track it down?
>>> 
>>> Thanks,
>>> 
>>> Colin
>>> ___
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>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> 

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Re: [OpenSIPS-Users] OpenSIPS on Amazon EC2

[Bogdan-Andrei Iancu]

:O.the classical problem strikes again !!!

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

On 26.01.2016 11:28, Jason Bedward wrote:


Thanks. Somehow iptables had turned itself back on! All working now.

On 26 Jan 2016 09:23, "Bogdan-Andrei Iancu" > wrote:


Maybe some blocking iptables rules on the INPUT chain ??

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

On 26.01.2016 11:18, Jason Bedward wrote:


No shows 0

On 26 Jan 2016 09:16, "Bogdan-Andrei Iancu" mailto:bog...@opensips.org>> wrote:

And do you see any data pending on the opensips listening
socket (via netstat in the Recv-Q column) ?

Regards,

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

On 26.01.2016 11:08, Jason Bedward wrote:


Yes as I said ngrep shows traffic but no action by opensips.
Very annoying as I have had it working on Amazon in the past.

Netstat shows opensips listening on my local host address on
port 5060 UDP.

Opensips logs not showing any traffic.

I have exact same config on digital ocean and it works. But
I want all servers in the same place. As for my config it's
the same that's on the freeswitch site wiki.

Regards

On 26 Jan 2016 08:57, "Bogdan-Andrei Iancu"
mailto:bog...@opensips.org>> wrote:

Do you see the incoming traffic on the opensips machine
(check with ngrep/tcpdump). Have you checked if OpenSIPS
is listening on the right port (where the traffic is
sent to) via "netstat -ulnp| greop opensips" ? Do you
see any pending data on the socket ?

Regards,

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

On 26.01.2016 10:46, Jason Bedward wrote:


When traiffic comes in for example options, it does not
respond or seem to process. There are no logs in
opensips for this traffic coming through.

On 26 Jan 2016 07:57, "Bogdan-Andrei Iancu"
mailto:bog...@opensips.org>> wrote:

In this case can you elaborate on "opensips does
not process traffic" ? do you see waiting data on
the listening sockets ?

Regards,

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

On 26.01.2016 00:32, Jason Bedward wrote:

I have the below already set.:

alias=42.48.159.140
listen=udp:eth0:5060
advertised_address=42.48.159.140


On Mon, Jan 25, 2016 at 1:37 PM, Jason Bedward
mailto:mase2...@gmail.com>>
wrote:

 Hi

I have opensips listening to all. If I type
Opensips it displays local and advertised ip
addresses.

Not sure why it's not working as it should.

Regards

Hi Jason,

If running in Amazon, what you have to do is:
1) set as listener into opensips the
local/private IP of the machine
2) set as "advertise_address" and as "alias"
the elastic IP/public you have.

Regards,

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

On 23.01.2016 20:08, Jason Bedward wrote:

Hi,

I have previously had this working without
issue. But for some reason now I can not get
it to work. I have set advertised IP, and
Opensips is showing that its waiting for
traffic on that IP.

But when traiffic comes in OpenSIPS does not
process it. I have used ngrep to monitor.
OpenSIPS itself seems to be working as I have
a keep alive which was talking to a
freeswitch server that was working.

I have taken the same config and put it on
Digital Ocean and it worked first time.

Also I have tried version 1.71, 1.8, 2 all
with same result. Anyone having success that
could give advice?

Thanks


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Re: [OpenSIPS-Users] potential memory leak warning - uac_auth:build_authorization_hdr

[Bogdan-Andrei Iancu]

Hi Colin,

Which of the in which module do you use the uac authentication feature :
uac_auth
b2b_entities
uac_registrant

?

Regards,

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

On 26.01.2016 11:22, Colin Martin wrote:

Just one extra bit of information, this doesn't happen on the first call 
post-restart, but does occur for each subsequent call.

I'm happy to ignore it if it won't affect stability, but I don't know how to 
verify that!

Colin


On 25 Jan 2016, at 18:25, Colin Martin  wrote:

Hi,

I’m working with OpenSIPS 2.1.2 and have started to see the following error 
message in the logs:

WARNING:uac_auth:build_authorization_hdr: potential memory leak at addr: 
0x7fc266b471f8

It seems to occur when using a TCP transport.

Is this a known bug? Is there anything I can do to capture more information to 
help track it down?

Thanks,

Colin
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Re: [OpenSIPS-Users] OpenSIPS on Amazon EC2

[Bogdan-Andrei Iancu]

Maybe some blocking iptables rules on the INPUT chain ??

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

On 26.01.2016 11:18, Jason Bedward wrote:


No shows 0

On 26 Jan 2016 09:16, "Bogdan-Andrei Iancu" > wrote:


And do you see any data pending on the opensips listening socket
(via netstat in the Recv-Q column) ?

Regards,

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

On 26.01.2016 11:08, Jason Bedward wrote:


Yes as I said ngrep shows traffic but no action by opensips. Very
annoying as I have had it working on Amazon in the past.

Netstat shows opensips listening on my local host address on port
5060 UDP.

Opensips logs not showing any traffic.

I have exact same config on digital ocean and it works. But I
want all servers in the same place. As for my config it's the
same that's on the freeswitch site wiki.

Regards

On 26 Jan 2016 08:57, "Bogdan-Andrei Iancu" mailto:bog...@opensips.org>> wrote:

Do you see the incoming traffic on the opensips machine
(check with ngrep/tcpdump). Have you checked if OpenSIPS is
listening on the right port (where the traffic is sent to)
via "netstat -ulnp| greop opensips" ? Do you see any pending
data on the socket ?

Regards,

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

On 26.01.2016 10:46, Jason Bedward wrote:


When traiffic comes in for example options, it does not
respond or seem to process. There are no logs in opensips
for this traffic coming through.

On 26 Jan 2016 07:57, "Bogdan-Andrei Iancu"
mailto:bog...@opensips.org>> wrote:

In this case can you elaborate on "opensips does not
process traffic" ? do you see waiting data on the
listening sockets ?

Regards,

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

On 26.01.2016 00:32, Jason Bedward wrote:

I have the below already set.:

alias=42.48.159.140
listen=udp:eth0:5060
advertised_address=42.48.159.140


On Mon, Jan 25, 2016 at 1:37 PM, Jason Bedward
mailto:mase2...@gmail.com>> wrote:

 Hi

I have opensips listening to all. If I type
Opensips it displays local and advertised ip
addresses.

Not sure why it's not working as it should.

Regards

Hi Jason,

If running in Amazon, what you have to do is:
1) set as listener into opensips the local/private
IP of the machine
2) set as "advertise_address" and as "alias" the
elastic IP/public you have.

Regards,

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

On 23.01.2016 20:08, Jason Bedward wrote:

Hi,

I have previously had this working without issue.
But for some reason now I can not get it to work.
I have set advertised IP, and Opensips is showing
that its waiting for traffic on that IP.

But when traiffic comes in OpenSIPS does not
process it. I have used ngrep to monitor. OpenSIPS
itself seems to be working as I have a keep alive
which was talking to a freeswitch server that was
working.

I have taken the same config and put it on Digital
Ocean and it worked first time.

Also I have tried version 1.71, 1.8, 2 all with
same result. Anyone having success that could give
advice?

Thanks


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Re: [OpenSIPS-Users] potential memory leak warning - uac_auth:build_authorization_hdr

[Colin Martin]

Just one extra bit of information, this doesn't happen on the first call 
post-restart, but does occur for each subsequent call.

I'm happy to ignore it if it won't affect stability, but I don't know how to 
verify that!

Colin 

> On 25 Jan 2016, at 18:25, Colin Martin  wrote:
> 
> Hi,
> 
> I’m working with OpenSIPS 2.1.2 and have started to see the following error 
> message in the logs:
> 
> WARNING:uac_auth:build_authorization_hdr: potential memory leak at addr: 
> 0x7fc266b471f8
> 
> It seems to occur when using a TCP transport.
> 
> Is this a known bug? Is there anything I can do to capture more information 
> to help track it down?
> 
> Thanks,
> 
> Colin
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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Re: [OpenSIPS-Users] OpenSIPS on Amazon EC2

[Bogdan-Andrei Iancu]

And do you see any data pending on the opensips listening socket (via 
netstat in the Recv-Q column) ?


Regards,

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

On 26.01.2016 11:08, Jason Bedward wrote:


Yes as I said ngrep shows traffic but no action by opensips. Very 
annoying as I have had it working on Amazon in the past.


Netstat shows opensips listening on my local host address on port 5060 
UDP.


Opensips logs not showing any traffic.

I have exact same config on digital ocean and it works. But I want all 
servers in the same place. As for my config it's the same that's on 
the freeswitch site wiki.


Regards

On 26 Jan 2016 08:57, "Bogdan-Andrei Iancu" > wrote:


Do you see the incoming traffic on the opensips machine (check
with ngrep/tcpdump). Have you checked if OpenSIPS is listening on
the right port (where the traffic is sent to) via "netstat -ulnp|
greop opensips" ? Do you see any pending data on the socket ?

Regards,

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

On 26.01.2016 10:46, Jason Bedward wrote:


When traiffic comes in for example options, it does not respond
or seem to process. There are no logs in opensips for this
traffic coming through.

On 26 Jan 2016 07:57, "Bogdan-Andrei Iancu" mailto:bog...@opensips.org>> wrote:

In this case can you elaborate on "opensips does not process
traffic" ? do you see waiting data on the listening sockets ?

Regards,

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

On 26.01.2016 00:32, Jason Bedward wrote:

I have the below already set.:

alias=42.48.159.140
listen=udp:eth0:5060
advertised_address=42.48.159.140


On Mon, Jan 25, 2016 at 1:37 PM, Jason Bedward
mailto:mase2...@gmail.com>> wrote:

 Hi

I have opensips listening to all. If I type Opensips it
displays local and advertised ip addresses.

Not sure why it's not working as it should.

Regards

Hi Jason,

If running in Amazon, what you have to do is:
1) set as listener into opensips the local/private IP of
the machine
2) set as "advertise_address" and as "alias" the elastic
IP/public you have.

Regards,

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

On 23.01.2016 20:08, Jason Bedward wrote:

Hi,

I have previously had this working without issue. But
for some reason now I can not get it to work. I have
set advertised IP, and Opensips is showing that its
waiting for traffic on that IP.

But when traiffic comes in OpenSIPS does not process
it. I have used ngrep to monitor. OpenSIPS itself seems
to be working as I have a keep alive which was talking
to a freeswitch server that was working.

I have taken the same config and put it on Digital
Ocean and it worked first time.

Also I have tried version 1.71, 1.8, 2 all with same
result. Anyone having success that could give advice?

Thanks


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Re: [OpenSIPS-Users] OpenSIPS on Amazon EC2

[Jason Bedward]

Yes as I said ngrep shows traffic but no action by opensips. Very annoying
as I have had it working on Amazon in the past.

Netstat shows opensips listening on my local host address on port 5060 UDP.

Opensips logs not showing any traffic.

I have exact same config on digital ocean and it works. But I want all
servers in the same place. As for my config it's the same that's on the
freeswitch site wiki.

Regards
On 26 Jan 2016 08:57, "Bogdan-Andrei Iancu"  wrote:

> Do you see the incoming traffic on the opensips machine (check with
> ngrep/tcpdump). Have you checked if OpenSIPS is listening on the right port
> (where the traffic is sent to) via "netstat -ulnp| greop opensips" ? Do you
> see any pending data on the socket ?
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 26.01.2016 10:46, Jason Bedward wrote:
>
> When traiffic comes in for example options, it does not respond or seem to
> process. There are no logs in opensips for this traffic coming through.
> On 26 Jan 2016 07:57, "Bogdan-Andrei Iancu"  wrote:
>
>> In this case can you elaborate on "opensips does not process traffic" ?
>> do you see waiting data on the listening sockets ?
>>
>> Regards,
>>
>> Bogdan-Andrei Iancu
>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>>
>> On 26.01.2016 00:32, Jason Bedward wrote:
>>
>> I have the below already set.:
>>
>> alias=42.48.159.140
>> listen=udp:eth0:5060
>> advertised_address=42.48.159.140
>>
>>
>> On Mon, Jan 25, 2016 at 1:37 PM, Jason Bedward 
>> wrote:
>>
>>>  Hi
>>>
>>> I have opensips listening to all. If I type Opensips it displays local
>>> and advertised ip addresses.
>>>
>>> Not sure why it's not working as it should.
>>>
>>> Regards
>>> Hi Jason,
>>>
>>> If running in Amazon, what you have to do is:
>>> 1) set as listener into opensips the local/private IP of the machine
>>> 2) set as "advertise_address" and as "alias" the elastic IP/public you
>>> have.
>>>
>>> Regards,
>>>
>>> Bogdan-Andrei Iancu
>>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>>>
>>> On 23.01.2016 20:08, Jason Bedward wrote:
>>>
>>> Hi,
>>>
>>> I have previously had this working without issue. But for some reason
>>> now I can not get it to work. I have set advertised IP, and Opensips is
>>> showing that its waiting for traffic on that IP.
>>>
>>> But when traiffic comes in OpenSIPS does not process it. I have used
>>> ngrep to monitor. OpenSIPS itself seems to be working as I have a keep
>>> alive which was talking to a freeswitch server that was working.
>>>
>>> I have taken the same config and put it on Digital Ocean and it worked
>>> first time.
>>>
>>> Also I have tried version 1.71, 1.8, 2 all with same result. Anyone
>>> having success that could give advice?
>>>
>>> Thanks
>>>
>>>
>>> ___
>>> Users mailing 
>>> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>>>
>>
>>
>
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Re: [OpenSIPS-Users] OpenSIPS on Amazon EC2

[Bogdan-Andrei Iancu]

Do you see the incoming traffic on the opensips machine (check with 
ngrep/tcpdump). Have you checked if OpenSIPS is listening on the right 
port (where the traffic is sent to) via "netstat -ulnp| greop opensips" 
? Do you see any pending data on the socket ?


Regards,

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

On 26.01.2016 10:46, Jason Bedward wrote:


When traiffic comes in for example options, it does not respond or 
seem to process. There are no logs in opensips for this traffic coming 
through.


On 26 Jan 2016 07:57, "Bogdan-Andrei Iancu" > wrote:


In this case can you elaborate on "opensips does not process
traffic" ? do you see waiting data on the listening sockets ?

Regards,

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

On 26.01.2016 00:32, Jason Bedward wrote:

I have the below already set.:

alias=42.48.159.140
listen=udp:eth0:5060
advertised_address=42.48.159.140


On Mon, Jan 25, 2016 at 1:37 PM, Jason Bedward
mailto:mase2...@gmail.com>> wrote:

 Hi

I have opensips listening to all. If I type Opensips it
displays local and advertised ip addresses.

Not sure why it's not working as it should.

Regards

Hi Jason,

If running in Amazon, what you have to do is:
1) set as listener into opensips the local/private IP of the
machine
2) set as "advertise_address" and as "alias" the elastic
IP/public you have.

Regards,

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

On 23.01.2016 20:08, Jason Bedward wrote:

Hi,

I have previously had this working without issue. But for
some reason now I can not get it to work. I have set
advertised IP, and Opensips is showing that its waiting for
traffic on that IP.

But when traiffic comes in OpenSIPS does not process it. I
have used ngrep to monitor. OpenSIPS itself seems to be
working as I have a keep alive which was talking to a
freeswitch server that was working.

I have taken the same config and put it on Digital Ocean and
it worked first time.

Also I have tried version 1.71, 1.8, 2 all with same result.
Anyone having success that could give advice?

Thanks


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Re: [OpenSIPS-Users] opensips transparent technology

[MichaelLeung]

can uac_replace_from read real phone number from databases?

On 01/25/2016 01:03 PM, MichaelLeung wrote:

thanks for reply
no , it is just a asking , i don't have real phone number database, or 
should i have one ?

can you tell me what is the name of this technology ?

On 01/24/2016 07:33 PM, Stefano Pisani wrote:

Where is their real phone number?
Do you have it in a database?
You can change the From header to show the real phone number.



Il 24/01/2016 12.22, MichaelLeung ha scritto:

Hi all

i was trying to make my opensips users to sent their real phone 
number when they call .


what is the name of this technology ? transmit transparently ?

i search google find nothing, and where can i read document of this 
technology ?


thanks.


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Re: [OpenSIPS-Users] rabbit event not passing params 2.2

[Răzvan Crainea]

Hi, Tito!

Can you send me a trace?

Thanks,
Răzvan

On 01/25/2016 10:41 PM, Tito Cumpen wrote:

Hey Razvan,

This is still an issue with the latest dev build. The event is 
entirely empty when it is transmitted to the queue. I've tried 
modparam("event_rabbitmq", "sync_mode", 1) but it does not make a 
difference.


THanks,
Tito


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OpenSIPS Core Developer
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Re: [OpenSIPS-Users] Tables Missing Access Permissions

[Bogdan-Andrei Iancu]

Hi Nathaniel,

Thanks for confirming the fix. Indeed, I found out the answer to my own 
question after sending you my first reply...after digging more into the 
code of opensipsdbctl.


Anyhow, thanks for report !

Regards,

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

On 26.01.2016 10:11, Nathaniel L. Keeling III wrote:

Bogdan,

To answer the first question, I created all of the tables including 
the extras. I did another git pull and recreated the tables. All of 
the permissions were in place except for the table route_tree_id_seq.


Thanks

Nathaniel Keeling

On 1/25/16 6:07 AM, Bogdan-Andrei Iancu wrote:

Hi (again),

I pushed a fix on GIT, please update and try to re-create your DB to 
see if all the grants are now fine.


Thanks and regards,

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

On 25.01.2016 13:48, Bogdan-Andrei Iancu wrote:

Hi Nathaniel,

When you created the tables, have you created all the "extra" tables 
? I see you have the b2b related tables (part of extra), but no kind 
of missing privileges on other extra tables like "siptrace".


Regards,

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

On 25.01.2016 05:38, Nathaniel L. Keeling III wrote:
When starting Opensips 2.1, I am getting permissions errors on 
dr_carriers and dr_carriers_id_seq tables in Postgresql and 
Opensips errors. After checking the database, these tables were 
missing any access permissions.



  Access privileges for database 
"opensips21"
 Schema |Name|   Type 
|  Access privileges
++--+-- 


 public | b2b_entities   | table|
 public | b2b_entities_id_seq| sequence |
 public | b2b_logic  | table|
 public | b2b_logic_id_seq   | sequence |
 public | dr_carriers| table|
 public | dr_carriers_id_seq | sequence |
 public | dr_partitions  | table|
 public | dr_partitions_id_seq   | sequence |
 public | route_tree_id_seq  | sequence |


Thanks

Nathaniel Keeling

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Re: [OpenSIPS-Users] WARNING:core:utimer_ticker: utimer task already schedualed

[Dragomir Haralambiev]

Thanks Bogdan,

Why this problem exists only in Opensips 2.1?

Best regards,
Dragomir

2016-01-26 9:59 GMT+02:00 Bogdan-Andrei Iancu :

> So after all, the problem was so slow/blocking communication with the
> Radius server. For the future, to debug such issue you can use the
> exec_msg_threshold to see what are the slow parts of your script:
> http://www.opensips.org/Documentation/Script-CoreParameters-2-1#toc57
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 26.01.2016 00:18, Aqs Younas wrote:
>
> Resolved the issue by down grading to radius 2.2. But stuck in another
> problem going to post another thread.
>
> Thank you All.
>
> On 25 January 2016 at 19:55, Aqs Younas  wrote:
>
>> We are still looking where things went wrong. Actually code is not
>> changed a bit , same radius configuration in production server works
>> perfect. We just copied the setup to new server and facing the problem. I
>> see a lot of people posted this issue before but none have shared the
>> solutions.
>>
>> Could you help where we need to look for radius connections.
>>
>> Much thanks for your pointers
>>
>> On 25 January 2016 at 19:18, Bogdan-Andrei Iancu < 
>> bog...@opensips.org> wrote:
>>
>>> Hi Aqs,
>>>
>>> I assume after fixing your RADIUS issue the timer warnings disappeared
>>> ?
>>>
>>> Regards,
>>>
>>> Bogdan-Andrei Iancu
>>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>>>
>>> On 22.01.2016 23:09, Aqs Younas wrote:
>>>
>>> Sorry, I must search archive before query here.
>>>
>>> The problem is related to radius accounting. I am too using radius for
>>> AAA and this problem appears after calls began to terminate.
>>> Below threads are purely related.
>>>
>>> http://comments.gmane.org/gmane.comp.voip.opensips.user/31415
>>> https://www.mail-archive.com/users@lists.opensips.org/msg30675.html
>>>
>>> On 22 January 2016 at 22:37, Aqs Younas < 
>>> aqsyou...@gmail.com> wrote:
>>>
 On the start of test. Using top

>>> [...]
>>>
>>>

 I see opensips stuck(flood with warnings) durings calls termination
 process which leads me to manually kill opensisp to prevent opensips from
 eating all my server resources.

 Test is performed using sipp with 30 cps and 2000 concurrent calls of
 during 6 minutes.

 :lscpu

 Architecture:  x86_64
 CPU op-mode(s):32-bit, 64-bit
 Byte Order:Little Endian
 CPU(s):32
 On-line CPU(s) list:   0-31
 Thread(s) per core:2
 Core(s) per socket:8
 Socket(s): 2
 NUMA node(s):  2
 Vendor ID: GenuineIntel
 CPU family:6
 Model: 45
 Stepping:  7
 CPU MHz:   1199.531
 BogoMIPS:  4001.49
 Virtualization:VT-x
 L1d cache: 32K
 L1i cache: 32K
 L2 cache:  256K
 L3 cache:  20480K
 NUMA node0 CPU(s): 0-7,16-23
 NUMA node1 CPU(s): 8-15,24-31

 egrep --color 'Mem|Cache|Swap' /proc/meminfo

 MemTotal:   49530560 kB
 MemFree:29677688 kB
 MemAvailable:   45436220 kB



 Let me know if you need anything else.

 Thanks.

 On 22 January 2016 at 21:18, Bogdan-Andrei Iancu <
 bog...@opensips.org> wrote:

> Yes, the "children" option - 10 should be more than ok.
>
> What is the CPU usage from opensips during the test ?
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 22.01.2016 18:08, Aqs Younas wrote:
>
> Hi, Bogdan
>
>
> You mean children? First I thought it is due to children(10 default) 
> which I increased to 500 but no avail.
>
> This I have in my configuration file.
>
> debug=3
> log_stderror=no
> log_facility=LOG_LOCAL3
>
> fork=yes
> children=500
> open_files_limit=9
>
>
> Thanks for replying.
>
>
> On 22 January 2016 at 20:49, Bogdan-Andrei Iancu <
> bog...@opensips.org> wrote:
>
>> How many workers have you configured into opensips ?
>>
>> Regards,
>>
>> Bogdan-Andrei Iancu
>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>>
>> On 22.01.2016 17:43, Aqs Younas wrote:
>>
>> I see this warning when i am sending calls with more than 10 cps. On
>> 5 to 10 cps everything seems ok.
>>
>> On 22 January 2016 at 19:38, Aqs Younas < 
>> aqsyou...@gmail.com> wrote:
>>
>>> Hi,
>>>
>>> I am using sipp to load test my opensips (version: opensips 2.1.2
>>> (x86_64/linux)) after calls get terminated i see my opensips being flood
>>> with below warnings.
>>>
>>> Jan 22 14:31:38 66-226-76-150
>>> /usr/local/origination/opensips/sbin

Re: [OpenSIPS-Users] CRITICAL:db_mysql:db_mysql_do_prepared_query: too many mysql server reconnection failures

[Bogdan-Andrei Iancu]

Hi Aqs,

It looks like you have some sql connectivity problems. Are you sure your 
opensips can reach the mysql server ? all your errors (about long query 
time and about the failure to reconnect) do point to a connectivity 
problem to mysql server.


Regards,

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

On 26.01.2016 00:27, Aqs Younas wrote:

Hi.

I am load testing my opensips(2.1.2) with sipp on. As soon as I send 
calls with 50 cps i see this in logs.


Jan 25 21:00:05 66-226-76-150 ./opensips[16398]: 
INFO:db_mysql:switch_state_to_disconnected: disconnect event for 
0x7f17b530f338
Jan 25 21:00:05 66-226-76-150 ./opensips[16398]: 
INFO:db_mysql:reset_all_statements: reseting all statements on 
connection: (0x7f17b530f1a0) 0x7f17b530f338
Jan 25 21:00:05 66-226-76-150 ./opensips[16400]: 
WARNING:db_mysql:log_expiry: threshold exceeded : mysql prep stmt took 
too long - 1999302 us.Source : delete from dialo

g where dlg_id=?
Jan 25 21:00:05 66-226-76-150 ./opensips[16400]: 
INFO:db_mysql:switch_state_to_disconnected: disconnect event for 
0x7f17b530f338
Jan 25 21:00:05 66-226-76-150 ./opensips[16400]: 
INFO:db_mysql:reset_all_statements: reseting all statements on 
connection: (0x7f17b530f1a0) 0x7f17b530f338
Jan 25 21:00:05 66-226-76-150 ./opensips[16398]: 
INFO:db_mysql:connect_with_retry: re-connected successful for 
0x7f17b530f338
Jan 25 21:00:05 66-226-76-150 ./opensips[16400]: 
INFO:db_mysql:connect_with_retry: re-connected successful for 
0x7f17b530f338
Jan 25 21:00:05 66-226-76-150 ./opensips[16398]: 
INFO:db_mysql:db_mysql_do_prepared_query: reconnected to mysql server 
-> re-init the statement
Jan 25 21:00:05 66-226-76-150 ./opensips[16400]: 
INFO:db_mysql:db_mysql_do_prepared_query: reconnected to mysql server 
-> re-init the statement
Jan 25 21:00:05 66-226-76-150 ./opensips[16398]: 
CRITICAL:db_mysql:db_mysql_do_prepared_query: too many mysql server 
reconnection failures
Jan 25 21:00:05 66-226-76-150 ./opensips[16398]: 
ERROR:dialog:update_dialog_dbinfo: could not update database info
Jan 25 21:00:05 66-226-76-150 ./opensips[16400]: 
CRITICAL:db_mysql:db_mysql_do_prepared_query: too many mysql server 
reconnection failures
Jan 25 21:00:05 66-226-76-150 ./opensips[16400]: 
ERROR:dialog:update_dialog_dbinfo: could not add another dialog to db
Jan 25 21:00:05 66-226-76-150 ./opensips[16398]: 
WARNING:core:handle_timer_job: timer job  has a 231 us 
delay in execution
Jan 25 21:00:05 66-226-76-150 ./opensips[16398]: 
WARNING:core:handle_timer_job: timer job  has a 231 
us delay in execution
Jan 25 21:00:05 66-226-76-150 ./opensips[16396]: 
WARNING:core:handle_timer_job: utimer job  has a 8 us 
delay in execution
Jan 25 21:00:05 66-226-76-150 ./opensips[16400]: 
WARNING:core:handle_timer_job: utimer job  has a 8 us 
delay in execution
Jan 25 21:00:05 66-226-76-150 ./opensips[16390]: 
WARNING:core:utimer_ticker: utimer task  already schedualed 
for 1436630 ms (now 1436730 ms), it may overlap.

.
Jan 25 21:00:05 66-226-76-150 ./opensips[16390]: 
WARNING:core:utimer_ticker: utimer task  already schedualed 
for 1436730 ms (now 1436830 ms), it may overlap.

.
Jan 25 21:00:05 66-226-76-150 ./opensips[16395]: 
WARNING:db_mysql:log_expiry: threshold exceeded : mysql prep stmt took 
too long - 1999067 us.Source : delete from dialo

g where dlg_id=?
Jan 25 21:00:05 66-226-76-150 ./opensips[16395]: 
INFO:db_mysql:switch_state_to_disconnected: disconnect event for 
0x7f17b530f338
Jan 25 21:00:05 66-226-76-150 ./opensips[16395]: 
INFO:db_mysql:reset_all_statements: reseting all statements on 
connection: (0x7f17b530f1a0) 0x7f17b530f338
Jan 25 21:00:05 66-226-76-150 ./opensips[16395]: 
INFO:db_mysql:connect_with_retry: re-connected successful for 
0x7f17b530f338
Jan 25 21:00:05 66-226-76-150 ./opensips[16395]: 
INFO:db_mysql:db_mysql_do_prepared_query: reconnected to mysql server 
-> re-init the statement
Jan 25 21:00:05 66-226-76-150 ./opensips[16395]: 
CRITICAL:db_mysql:db_mysql_do_prepared_query: too many mysql server 
reconnection failures
Jan 25 21:00:05 66-226-76-150 ./opensips[16395]: 
ERROR:dialog:update_dialog_dbinfo: could not add another dialog to db
Jan 25 21:00:05 66-226-76-150 ./opensips[16395]: 
WARNING:core:handle_timer_job: utimer job  has a 5 us 
delay in execution


mysql server is on localhost and server is of good specs.

Could someone help me how i resolve or why there are so many these lines.

Thanks.
 
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Re: [OpenSIPS-Users] AVP_DB_Querry transformation question

[Bogdan-Andrei Iancu]

Travis,

The documentation on formating the variables is available under:
http://www.opensips.org/Documentation/Script-CoreVar-2-1

Regards,

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

On 26.01.2016 00:16, Travis Manson-Drake wrote:


That did it!

Thank you!

*From:*users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] *On Behalf Of *jar...@unixc.org

*Sent:* Monday, January 25, 2016 3:11 PM
*To:* OpenSIPS users mailling list 
*Subject:* Re: [OpenSIPS-Users] AVP_DB_Querry transformation question

try $(fU{s.escape.common})

On Jan 25, 2016, at 4:09 PM, Travis Manson-Drake
mailto:trav...@simplybits.com>> wrote:

Hello everyone,

I can’t seem to find any documentation on how to format my Query
while using the {s.escape.common} transformation.

I tried setting it up as follows:

select MAC FROM `phone_registrations` where auth_userid =
'$fU{s.escape.common}'))", "$avp(PBX)")

however the query just returns the value PV $fU with
{s.escape.common} in the return, for example 7117test{s.escape.common}

where should I apply the transformation? Can someone give me an
example?

Thank you for your time.

Travis Manson-Drake
Voice Systems Analyst

Simply Bits, LLC
T:520.545.0311 F:520.545.7252
E:em...@simplybits.com 
5225 N. Sabino Canyon Road
Tucson, AZ 85750
Support Hotline: 520.545.0333

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Re: [OpenSIPS-Users] Tables Missing Access Permissions

[Nathaniel L. Keeling III]

Bogdan,

To answer the first question, I created all of the tables including the 
extras. I did another git pull and recreated the tables. All of the 
permissions were in place except for the table route_tree_id_seq.


Thanks

Nathaniel Keeling

On 1/25/16 6:07 AM, Bogdan-Andrei Iancu wrote:

Hi (again),

I pushed a fix on GIT, please update and try to re-create your DB to 
see if all the grants are now fine.


Thanks and regards,

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

On 25.01.2016 13:48, Bogdan-Andrei Iancu wrote:

Hi Nathaniel,

When you created the tables, have you created all the "extra" tables 
? I see you have the b2b related tables (part of extra), but no kind 
of missing privileges on other extra tables like "siptrace".


Regards,

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

On 25.01.2016 05:38, Nathaniel L. Keeling III wrote:
When starting Opensips 2.1, I am getting permissions errors on 
dr_carriers and dr_carriers_id_seq tables in Postgresql and Opensips 
errors. After checking the database, these tables were missing any 
access permissions.



  Access privileges for database 
"opensips21"
 Schema |Name|   Type |  
Access privileges
++--+-- 


 public | b2b_entities   | table|
 public | b2b_entities_id_seq| sequence |
 public | b2b_logic  | table|
 public | b2b_logic_id_seq   | sequence |
 public | dr_carriers| table|
 public | dr_carriers_id_seq | sequence |
 public | dr_partitions  | table|
 public | dr_partitions_id_seq   | sequence |
 public | route_tree_id_seq  | sequence |


Thanks

Nathaniel Keeling

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Re: [OpenSIPS-Users] WARNING:core:utimer_ticker: utimer task already schedualed

[Bogdan-Andrei Iancu]

So after all, the problem was so slow/blocking communication with the 
Radius server. For the future, to debug such issue you can use the 
exec_msg_threshold to see what are the slow parts of your script:

http://www.opensips.org/Documentation/Script-CoreParameters-2-1#toc57

Regards,

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

On 26.01.2016 00:18, Aqs Younas wrote:
Resolved the issue by down grading to radius 2.2. But stuck in another 
problem going to post another thread.


Thank you All.

On 25 January 2016 at 19:55, Aqs Younas > wrote:


We are still looking where things went wrong. Actually code is not
changed a bit , same radius configuration in production server
works perfect. We just copied the setup to new server and facing
the problem. I see a lot of people posted this issue before but
none have shared the solutions.

Could you help where we need to look for radius connections.

Much thanks for your pointers

On 25 January 2016 at 19:18, Bogdan-Andrei Iancu
mailto:bog...@opensips.org>> wrote:

Hi Aqs,

I assume after fixing your RADIUS issue the timer warnings
disappeared   ?

Regards,

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

On 22.01.2016 23:09, Aqs Younas wrote:

Sorry, I must search archive before query here.

The problem is related to radius accounting. I am too using
radius for AAA and this problem appears after calls began to
terminate.
Below threads are purely related.

http://comments.gmane.org/gmane.comp.voip.opensips.user/31415
https://www.mail-archive.com/users@lists.opensips.org/msg30675.html

On 22 January 2016 at 22:37, Aqs Younas mailto:aqsyou...@gmail.com>> wrote:

On the start of test. Using top


[...]




I see opensips stuck(flood with warnings) durings calls
termination process which leads me to manually kill
opensisp to prevent opensips from eating all my server
resources.

Test is performed using sipp with 30 cps and 2000
concurrent calls of during 6 minutes.

:lscpu

Architecture:  x86_64
CPU op-mode(s):32-bit, 64-bit
Byte Order:Little Endian
CPU(s):32
On-line CPU(s) list:   0-31
Thread(s) per core:2
Core(s) per socket:8
Socket(s): 2
NUMA node(s):  2
Vendor ID: GenuineIntel
CPU family:6
Model: 45
Stepping:  7
CPU MHz: 1199.531
BogoMIPS:  4001.49
Virtualization:VT-x
L1d cache: 32K
L1i cache: 32K
L2 cache:  256K
L3 cache:  20480K
NUMA node0 CPU(s): 0-7,16-23
NUMA node1 CPU(s): 8-15,24-31

egrep --color 'Mem|Cache|Swap' /proc/meminfo

MemTotal:   49530560 kB
MemFree:29677688 kB
MemAvailable:   45436220 kB



Let me know if you need anything else.

Thanks.

On 22 January 2016 at 21:18, Bogdan-Andrei Iancu
mailto:bog...@opensips.org>> wrote:

Yes, the "children" option - 10 should be more than ok.

What is the CPU usage from opensips during the test ?

Regards,

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

On 22.01.2016 18:08, Aqs Younas wrote:

Hi, Bogdan

You mean children? First I thought it is due to children(10 
default) which I increased to 500 but no avail.
This I have in my configuration file.

debug=3
log_stderror=no
log_facility=LOG_LOCAL3

fork=yes
children=500
open_files_limit=9

Thanks for replying.

On 22 January 2016 at 20:49, Bogdan-Andrei Iancu
mailto:bog...@opensips.org>>
wrote:

How many workers have you configured into opensips ?

Regards,

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

On 22.01.2016 17:43, Aqs Younas wrote:

I see this warning when i am sending calls with
more than 10 cps. On 5 to 10 cps everything
seems ok.

On 22 January 2016 a