Re: [OpenSIPS-Users] xlog adding information to output

[Ben Newlin]

Bogdan,

I can understand the desire for consistency.

Unfortunately in our case we are using an external log service vendor and the 
application that is collecting the logs is provided by them. It simply collects 
all logs and does not have any option to preprocess individual log entries. We 
don’t have the ability to pre-process the logs prior to ingestion. We can, I 
think, filter out the extra info while searching the logs, but it will make the 
searches more complex and slower.

It would be nice to be able to disable this so there is some option to only log 
what is provided without manipulation. It could be a flag that affects xlog, or 
a different log function that doesn’t use the prefix. If you think it is 
worthy, I’d be happy to open a feature request.

Thanks,
Ben Newlin

From: Bogdan-Andrei Iancu 
Date: Thursday, April 12, 2018 at 4:02 PM
To: OpenSIPS users mailling list , Ben Newlin 

Subject: Re: [OpenSIPS-Users] xlog adding information to output

Hi Ben,

Yes, this change was done on purpose for logs consistency reasons. As when you 
log via syslog you will get *all* the time the process name, pid and data as 
suffix, we considered that printing to strerr should follow the same model and 
have *all* the time the same prefix for the messages.

Now, I guess you grep the logs by some sort of an application (before having 
them pushed as jsons for later processing) - maybe you can do some extra 
pre-processing the strip the prefix before parsing it as json.

Regards,


Bogdan-Andrei Iancu



OpenSIPS Founder and Developer

  http://www.opensips-solutions.com

OpenSIPS Summit 2018

  
http://www.opensips.org/events/Summit-2018Amsterdam
On 04/06/2018 10:58 PM, Ben Newlin wrote:
Hi,

Yet another question. ☺

We use the json module to structure our output to and then print using xlog. 
Our log consumer processes the json format and can process and index our logs.

It appears that in 2.x the xlog module now always prepends a date string and 
the process number before every log. This is causing the logs to not be valid 
json format and they are not parsed properly by our consumer.

I know that OpenSIPS’ internal logs always print in this format with the date 
and process number in front. Was this an intentional change to make xlog print 
this way as well? Can it be disabled in any way?

I’m pretty sure I remember someone else asking about this not too long ago but 
I couldn’t find the thread in the archive.

Thanks,
Ben Newlin





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Re: [OpenSIPS-Users] codec stripping with rtpengine

[Esty, Ryan]

Razvan,

Rtpengine is printing out this: rtpengine:parse_flags: error processing flag 
`codec-strip-VP8': unknown error. As I look at it now I don't see how it could 
be your issue. You aren't modifying the flag I sent to rtpengine. If it helps 
my version of opensips is 2.3.3.

Ryan


-Original Message-
From: Users [mailto:users-boun...@lists.opensips.org] On Behalf Of Razvan 
Crainea
Sent: Friday, April 13, 2018 9:17 AM
To: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] codec stripping with rtpengine

Hi, Ryan!

I think the issue you are talking about is related to OpenSIPS, rather than 
rtpengine, since you are getting the error in OpenSIPS, is that right?

Can you confirm what version of OpenSIPS you are using?

Best regards,
Răzvan

On 04/13/2018 03:42 PM, Esty, Ryan wrote:
> Bogdan-Andrei,
> 
> Thanks for the information just in case someone else looks for this, 
> this is the tracker https://github.com/sipwise/rtpengine/issues/525.
> 
> Ryan
> 
> *From:*Bogdan-Andrei Iancu [mailto:bog...@opensips.org]
> *Sent:* Thursday, April 12, 2018 4:08 PM
> *To:* OpenSIPS users mailling list ; Esty, 
> Ryan 
> *Subject:* Re: [OpenSIPS-Users] codec stripping with rtpengine
> 
> Hi Ryan,
> 
> yeah, this happens because OpenSIPS applies all the changes at the 
> end, when the message is about to be sent out. As a side effect, when 
> sending the SDP to rtpengine, opensips does not see its own previous 
> changes over the body (changes are still pending).
> Usually there are easy workarounds for this, but in this case it looks 
> like bug to me. Could you please open a bug report the the github tracker.
> 
> Best regards,
> 
> Bogdan-Andrei Iancu
> 
> OpenSIPS Founder and Developer
> 
>http://www.opensips-solutions.com
> 
> OpenSIPS Summit 2018
> 
>http://www.opensips.org/events/Summit-2018Amsterdam
> 
> On 04/09/2018 05:22 PM, Esty, Ryan wrote:
> 
> Hi opensips list,
> 
> First some background I’m trying to use opensips as a webrtc proxy.
> I found out that the packets for the invite going to my sip server
> are too big for my sip server. It doesn’t like packets to be over
> 4000 bytes. I’m trying to take what I can out of the sip packets
> like codes I know the other side can’t do. First codec stripping
> works but only with the audio codecs. If I try to strip a video
> codec the packet gets mangled. This is probably a bug in rtpengine
> and not opensips. I was hoping if anyone has any idea how I might
> get my invite packets smaller? The webrtc side is generating ssrc
> lines in my sdp. I’m trying to strip them out but I’m not sure if
> rtpengine is putting them back in or not. Before my rtpengine_offer
> I do a replace_body_all(“a=ssrc.*\r\n,” “”) but the invite still has
> all the ssrc lines in it.
> 
> Ryan
> 
> 
> 
> 
> ___
> 
> Users mailing list
> 
> Users@lists.opensips.org 
> 
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> 
> 
> 
> ___
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> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> 

--
Răzvan Crainea
OpenSIPS Core Developer
   http://www.opensips-solutions.com
OpenSIPS Summit 2018
   http://www.opensips.org/events/Summit-2018Amsterdam

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Re: [OpenSIPS-Users] sip options pings are empty

[Bogdan-Andrei Iancu]

Hi Tito,

A ping cannot be empty - if it is empty, it is not a ping :) . Maybe it 
is the UDP ping (the 4 bytes of zero value) ?? If so, it means you did 
not set the NAT_SIP_PINGS branch flag when the contact was registered 
(before the save(location)).


IF you do opensipsctl ul show, you should see the saved branch flags.

Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
  http://www.opensips-solutions.com
OpenSIPS Summit 2018
  http://www.opensips.org/events/Summit-2018Amsterdam

On 04/13/2018 01:28 AM, Tito Cumpen wrote:

Hello,


I am using opensips 2.3 git rev. 3a66b9c and I am noticing that sip 
options pings to wss clients are sent with no body at all. I am using 
SIP.js/0.7. and see that the options sent when using :



 modparam("nathelper", "natping_interval", 10)
modparam("nathelper", "natping_tcp", 1)
modparam("nathelper", "sipping_from", "sip:pin...@mydomain.net 
")

modparam("nathelper", "remove_on_timeout_bflag", "RM_ONTO_FLAG")
modparam("nathelper", "sipping_bflag", "NAT_SIP_PINGS")

are empty and therefore the these clients cannot be monitored because 
sipjs dismisses the empty messages
Thu Apr 12 2018 18:27:14 GMT-0400 (EDT) | sip.parser | no CRLF found, 
not a SIP message, discarded
sip-0.7.7.min.js:36 Thu Apr 12 2018 18:27:23 GMT-0400 (EDT) | 
sip.transport | received WebSocket text message:



Thanks,
Tito


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Re: [OpenSIPS-Users] codec stripping with rtpengine

[Răzvan Crainea]

Hi, Ryan!

I think the issue you are talking about is related to OpenSIPS, rather 
than rtpengine, since you are getting the error in OpenSIPS, is that right?


Can you confirm what version of OpenSIPS you are using?

Best regards,
Răzvan

On 04/13/2018 03:42 PM, Esty, Ryan wrote:

Bogdan-Andrei,

Thanks for the information just in case someone else looks for this, 
this is the tracker https://github.com/sipwise/rtpengine/issues/525.


Ryan

*From:*Bogdan-Andrei Iancu [mailto:bog...@opensips.org]
*Sent:* Thursday, April 12, 2018 4:08 PM
*To:* OpenSIPS users mailling list ; Esty, 
Ryan 

*Subject:* Re: [OpenSIPS-Users] codec stripping with rtpengine

Hi Ryan,

yeah, this happens because OpenSIPS applies all the changes at the end, 
when the message is about to be sent out. As a side effect, when sending 
the SDP to rtpengine, opensips does not see its own previous changes 
over the body (changes are still pending).
Usually there are easy workarounds for this, but in this case it looks 
like bug to me. Could you please open a bug report the the github tracker.


Best regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

   http://www.opensips-solutions.com

OpenSIPS Summit 2018

   http://www.opensips.org/events/Summit-2018Amsterdam

On 04/09/2018 05:22 PM, Esty, Ryan wrote:

Hi opensips list,

First some background I’m trying to use opensips as a webrtc proxy.
I found out that the packets for the invite going to my sip server
are too big for my sip server. It doesn’t like packets to be over
4000 bytes. I’m trying to take what I can out of the sip packets
like codes I know the other side can’t do. First codec stripping
works but only with the audio codecs. If I try to strip a video
codec the packet gets mangled. This is probably a bug in rtpengine
and not opensips. I was hoping if anyone has any idea how I might
get my invite packets smaller? The webrtc side is generating ssrc
lines in my sdp. I’m trying to strip them out but I’m not sure if
rtpengine is putting them back in or not. Before my rtpengine_offer
I do a replace_body_all(“a=ssrc.*\r\n,” “”) but the invite still has
all the ssrc lines in it.

Ryan




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--
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OpenSIPS Core Developer
  http://www.opensips-solutions.com
OpenSIPS Summit 2018
  http://www.opensips.org/events/Summit-2018Amsterdam

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Re: [OpenSIPS-Users] Fwd: [SR-Users] MSILO: SIP stored offline MESSAGE repetitive delivery

[Bogdan-Andrei Iancu]

Hi Abdul,

Not sure if this is applicable for us. According to the script sample, 
the fr_timer (in seconds) is 10Is this value too low for a server to 
properly reply back ??


I guess that bug is Kamailio specific. We are on the safe side ;)

Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
  http://www.opensips-solutions.com
OpenSIPS Summit 2018
  http://www.opensips.org/events/Summit-2018Amsterdam

On 04/13/2018 12:42 AM, Abdul Basit wrote:

Hi Opensips team,

I faced instant message timeout issues with MSILO module while setting 
SIP IM server using kamailio.
It turned out that fr_timer values was too low to wait for MESSAGE 
delivery to the destination (online/offline both cases) that server 
was replying with 408.
Which result in message sending failed on sending side where as IM was 
reaching to the destination.


The same issue is true for msilo sample configuration 
 
for opensips.
Kindly update it at your end as well. This will save lot of time for 
the ones who want to use opensips as IM server.


--
regards,

abdul basit | p: +92 32 1416 4196 | o: +92 30 0841 1445

-- Forwarded message --
From: *Abdul Basit* >
Date: 13 April 2018 at 02:28
Subject: Re: [SR-Users] MSILO: SIP stored offline MESSAGE repetitive 
delivery

To: Henning Westerholt >


Thanks Henning.

Good move. You removed the timers so that they use their default values :)

Same need to be done in sample msilo script
https://kamailio.org/docs/modules/5.1.x/modules/msilo.html#idp45433660 



Who will be doing that?


--
regards,

abdul basit | p: +92 32 1416 4196 | o: +92 30 0841 1445

On 13 April 2018 at 00:37, Henning Westerholt > wrote:


On Thursday, 12 April 2018 10:15:19 CEST Abdul Basit wrote:
> My issue for IM handling has been resolved.
>
> @MS helped to look into the matter. From to the example
> [..]
>
> I replaced it as below
>
> # -- tm params --
>
> modparam("tm", "fr_timer", 1 )
> modparam("tm", "fr_inv_timer", 15 )
> modparam("tm", "wt_timer", 10 )
>
>
> Lower fr_timer was initiating 408 without waiting for 200 OK from
> destination because 10ms is too low. This was confusing msilo
module and
> the sender device that MESSAGE sent was failed.
> Increasing the fr_timer value resolved the issue.
> This also resolved a ripple effect. i.e, kamailio delivery of
offline
> messages from DB store to the destination party.
> Since kamailio was get message delivery error, it was
> executing failure_route[1] that was storing the message again in
DB store
> as offline message and so on.
>
> lower fr_timer values are also exits in 3.x, 4.x, 5x and dev
branches
> documentation.
> [..]

Hello Abdul,

great that you found the issue. I have fixed this issue in git
master and also
5.1 and 5.0 branch.

Best regards,

Henning





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Re: [OpenSIPS-Users] codec stripping with rtpengine

[Esty, Ryan]

Bogdan-Andrei,

Thanks for the information just in case someone else looks for this, this is 
the tracker https://github.com/sipwise/rtpengine/issues/525.

Ryan

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org]
Sent: Thursday, April 12, 2018 4:08 PM
To: OpenSIPS users mailling list ; Esty, Ryan 

Subject: Re: [OpenSIPS-Users] codec stripping with rtpengine

Hi Ryan,

yeah, this happens because OpenSIPS applies all the changes at the end, when 
the message is about to be sent out. As a side effect, when sending the SDP to 
rtpengine, opensips does not see its own previous changes over the body 
(changes are still pending).
Usually there are easy workarounds for this, but in this case it looks like bug 
to me. Could you please open a bug report the the github tracker.

Best regards,


Bogdan-Andrei Iancu



OpenSIPS Founder and Developer

  http://www.opensips-solutions.com

OpenSIPS Summit 2018

  http://www.opensips.org/events/Summit-2018Amsterdam
On 04/09/2018 05:22 PM, Esty, Ryan wrote:
Hi opensips list,

First some background I'm trying to use opensips as a webrtc proxy. I found out 
that the packets for the invite going to my sip server are too big for my sip 
server. It doesn't like packets to be over 4000 bytes. I'm trying to take what 
I can out of the sip packets like codes I know the other side can't do. First 
codec stripping works but only with the audio codecs. If I try to strip a video 
codec the packet gets mangled. This is probably a bug in rtpengine and not 
opensips. I was hoping if anyone has any idea how I might get my invite packets 
smaller? The webrtc side is generating ssrc lines in my sdp. I'm trying to 
strip them out but I'm not sure if rtpengine is putting them back in or not. 
Before my rtpengine_offer I do a replace_body_all("a=ssrc.*\r\n," "") but the 
invite still has all the ssrc lines in it.

Ryan





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Re: [OpenSIPS-Users] register to a sip server

[Alexey K . via Users]

Hello

As the simplest example, having 10 or smth like that DIDs/inner phone numbers,
you may use just core functions - 
https://www.opensips.org/Documentation/Script-CoreFunctions-2-3#toc18

and core variables   
https://www.opensips.org/Documentation/Script-CoreVar-2-3#toc75   

to determine the username in incoming INVITE (e.g.  00498970043400)
and forward this request to desirable destination '1000'

if(method=="REGISTER" && $rU==00498970043400) {
forward("10.0.0.10:5060"); # ip of 1000
}
OpenSIPS is extremely flexible, you may set/rewrite almost any SIP parameter 
using it - 
destination IP, destination username in uri and so on.


But of course hardcoding in the script is not the best way. If you have a lot 
of relations
"DID - inner exten" it's better to have a static code with AVP variables and 
get the values from
some database, using some of CACHEDB_ module.

PS: I'm not a experienced user of OpenSIPS, so check my advice twice...

---
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Re: [OpenSIPS-Users] register to a sip server

[Alexey K . via Users]

Hello

As the simplest example, having 10 or smth like that DIDs/inner phone numbers,
you may use just core functions - 
https://www.opensips.org/Documentation/Script-CoreFunctions-2-3#toc18

and core variables  
https://www.opensips.org/Documentation/Script-CoreVar-2-3#toc75  

to determine the username in incoming INVITE (e.g.  00498970043400)
and forward this request to desirable destination '1000'

if(method=="REGISTER" && $rU==00498970043400) {
forward("10.0.0.10:5060");# ip of 1000
}
OpenSIPS is extremely flexible, you may set/rewrite almost any SIP parameter 
using it - 
destination IP, destination username in uri and so on.


But of course hardcoding in the script is not the best way. If you have a lot 
of relations
"DID - inner exten" it's better to have a static code with AVP variables and 
get the values from
some database, using some of CACHEDB_ module.

PS: I'm not a experienced user of OpenSIPS, so check my advice twice...

---
BR, Alexey
http://alexeyka.zantsev.com/
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Re: [OpenSIPS-Users] register to a sip server

[Alexey K . via Users]

Hello

As the simplest example, having 10 or smth like that DIDs/inner phone numbers,
you may use just core functions - 
https://www.opensips.org/Documentation/Script-CoreFunctions-2-3#toc18

and core variables  
https://www.opensips.org/Documentation/Script-CoreVar-2-3#toc75  

to determine the username in incoming INVITE (e.g.  00498970043400)
and forward this request to desirable destination '1000'

if(method=="REGISTER" && $rU==00498970043400) {
forward("10.0.0.10:5060");# ip of 1000
}
OpenSIPS is extremely flexible, you may set/rewrite almost any SIP parameter 
using it - 
destination IP, destination username in uri and so on.


But of course hardcoding in the script is not the best way. If you have a lot 
of relations
"DID - inner exten" it's better to have a static code with AVP variables and 
get the values from
some database, using some of CACHEDB_ module.

PS: I'm not a experienced user of OpenSIPS, so check my advice twice...

---
BR, Alexey
http://alexeyka.zantsev.com/
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Re: [OpenSIPS-Users] drouting: need more information about a feature

[Abdoul Osséni]

Hi,
I hope, it will be completed in the future.
Thank you for your feedback.

Abdoul OSSENI
Ingénieur DevOps chez Néo-Soft
Co-Fondateur de ON SERVICES
Tél : +33 601 135 167


Le jeu. 12 avr. 2018 à 22:03, Bogdan-Andrei Iancu  a
écrit :

> Hi,
>
> This module was never completed and never pushed to the public tree. Maybe
> in the future :)
>
> Best regards,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
>   http://www.opensips-solutions.com
> OpenSIPS Summit 2018
>   http://www.opensips.org/events/Summit-2018Amsterdam
>
> On 04/09/2018 11:53 AM, Abdoul Osséni wrote:
>
> Hello Dear list,
>
> According to this email
> https://opensips.org/pipermail/users/2014-September/029817.html, drouting
> can reorder the gateways according PDD, ASR and ACD stats.
>
> Can you explain me how it works and how can I enable this feature?
>
> Thank you for your works.
>
> Abdoul.
>
>
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>
>
>
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