[OpenSIPS-Users] no transport protocol

2016-10-18 Thread ‪H Yavari‬ ‪
Hi to all,
I installed version 2.2.1, I'm using default configs. But when I start Opensips 
I receive this error:
Oct 18 10:29:51 [2569] ERROR:core:main: no trasnport protocol loaded

Can you please give me some hints?
"Note: transport spelled wrong too!"

Regards,H.Yavari
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Re: [OpenSIPS-Users] Why am I gettin error to terminate long calls and not short ones? Why it is "Ignoring callid"?

2016-10-18 Thread Răzvan Crainea

Hi, Rodrigo!

Most likely A closes the connection to OpenSIPS. You can check that by 
tracing the communication between A and OpenSIPS.
In order to solve that, make sure that the TCP keepalive[1] is enabled. 
Also, you can use the dialog pinging[2] feature ('Pp' flag to 
create_dialog()) to keep the dialog connections open.


[1] http://www.opensips.org/Documentation/Script-CoreParameters-2-2#toc103
[2] http://www.opensips.org/html/docs/modules/2.2.x/dialog.html#id295792

Best regards,

Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com

On 10/17/2016 11:20 PM, Rodrigo Pimenta Carvalho wrote:

Dear OpenSIPS users,


In my hardware, with IP = 192.168.0.101, I have OpenSIPS and softphone 
A. Thre is softphone B also, in another hardware.


A calls B.
B accept the call.
After *_t_* minutes...B hungs up the call.

In this moment, A enters in a wrong state, because OpenSIPS reports a 
problem and probably due to it the proxy doesn't communicate with 
softphone A in such moment. So, my softphone A considers that the call 
is not ended.


See what OpenSIPS reports in this moment:

Jan 05 04:14:29 colibri-imx6-jfl opensips[431]: Jan  5 04:14:29 [438] 
INFO:core:probe_max_sock_buff: using snd buffer of 320 kb
Jan 05 04:14:29 colibri-imx6-jfl opensips[431]: Jan  5 04:14:29 [438] 
INFO:core:init_sock_keepalive: TCP keepalive enabled on socket 20
Jan 05 04:14:29 colibri-imx6-jfl opensips[431]: Jan  5 04:14:29 [438] 
ERROR:core:tcpconn_async_connect: poll error: flags 1c
Jan 05 04:14:29 colibri-imx6-jfl opensips[431]: Jan  5 04:14:29 [438] 
ERROR:core:tcpconn_async_connect: failed to retrieve SO_ERROR 
[server=192.168.0.101:57985] (111) Connection refused
Jan 05 04:14:29 colibri-imx6-jfl opensips[431]: Jan  5 04:14:29 [438] 
ERROR:core:proto_tcp_send: async TCP connect failed
Jan 05 04:14:29 colibri-imx6-jfl opensips[431]: Jan  5 04:14:29 [438] 
ERROR:tm:msg_send: send() for proto 2 failed
Jan 05 04:14:29 colibri-imx6-jfl opensips[431]: Jan  5 04:14:29 [438] 
ERROR:tm:t_forward_nonack: sending request failed


If *_t_* is just few minutes, let's say 2 minutes, there is no any issue.

However, if *_t_* is bigger, let's say 4 minutes, his issue is present.


What is happening here? Can someone give some help, please!

Any hint will be very helpful!




Some more details:

User B is g1r2u3p4o5@192.168.0.102.

User A is intercomA_5dtUWgwgqzR6@192.168.0.101.

Callid was "ec4548a8-4207-4fc2-8ed8-81897ff62175".


Before getting such error log, I saw another messages in the log like 
this:



Jan 05 04:13:28 colibri-imx6-jfl opensips[431]: new branch at 
sip:g1r2u3p4o5@192.168.0.102:61230;transport=TCP;ob

Jan 05 04:13:28 colibri-imx6-jfl opensips[431]: od: invalid option -- 'A'
Jan 05 04:13:28 colibri-imx6-jfl opensips[431]: BusyBox v1.22.1 
(2016-03-29 09:43:20 BRT) multi-call binary.
Jan 05 04:13:28 colibri-imx6-jfl opensips[431]: Usage: od 
[-aBbcDdeFfHhIiLlOovXx] [FILE]
Jan 05 04:13:29 colibri-imx6-jfl opensips[431]: Ignoring callid 
"ec4548a8-4207-4fc2-8ed8-81897ff62175"



Jan 05 04:13:34 colibri-imx6-jfl opensips[431]: Jan  5 04:13:34 [442] 
INFO:core:probe_max_sock_buff: using snd buffer of 320 kb
an 05 04:13:34 colibri-imx6-jfl opensips[431]: Jan  5 04:13:34 [442] 
INFO:core:init_sock_keepalive: TCP keepalive enabled on socket 25


an 05 04:14:29 colibri-imx6-jfl opensips[431]: Jan  5 04:14:29 [438] 
ERROR:dialog:dlg_validate_dialog: failed to validate remote contact: 
dlg=[sip:intercomA_5dtUWgwgqzR6@127.0.0.1:36427;transport=TCP;ob] , 
req=[sip:intercomA_5dtUWgwgqzR6@192.168.0.101:57985;transport=TCP;ob]
Jan 05 04:14:29 colibri-imx6-jfl opensips[431]: In-Dialog BYE from 
192.168.0.102 (callid=ec4548a8-4207-4fc2-8ed8-81897ff62175) is not 
valid according to dialog




Best regards.



RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979


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Re: [OpenSIPS-Users] no transport protocol

2016-10-18 Thread Răzvan Crainea

Hi, Yavari!

Make sure you are loading at least one protocol module. For example:

loadmodule "proto_udp.so" # loads the UDP protocol module
loadmodule "proto_tcp.so" # loads the TCP protocol module

Best regards,
Răzvan

Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com

On 10/18/2016 10:03 AM, ‪H Yavari‬ ‪ wrote:

Hi to all,

I installed version 2.2.1, I'm using default configs. But when I start 
Opensips I receive this error:


Oct 18 10:29:51 [2569] ERROR:core:main: no trasnport protocol loaded

Can you please give me some hints?

"Note: transport spelled wrong too!"


Regards,
H.Yavari


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Re: [OpenSIPS-Users] OpenSIPS in the market.

2016-10-18 Thread Bogdan-Andrei Iancu

Hello Rodrigo,

The questions your are asking are hard to answer and the Open Source 
world is most of the times opaque when comes to who is using and how 
much is used. As anyone can simply download and start using (information 
on how to do it) is quite enough, you may have many cases were companies 
do use an Open Source software without "leaking" or "leaving traces" 
about that.


This is also for OpenSIPS project - it is hard to say who is using and 
how the popularity is fluctuating and mainly because there is no much of 
a feedback (as it is free to use).


Still, take a look at :
 http://www.opensips.org/About/WhoIsUsing

This is an attempt to list the companies willing to do it, it is not a 
full listing.


Best regards,

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

On 13.10.2016 15:44, Rodrigo Pimenta Carvalho wrote:


Dear OpenSIPS users,


As a teacher, I'm going to explain somethings about SIP and SIP 
proxies to my students.


So, I intend to talk about OpenSIPS too.


1- Is there some ranking in any web page showing what is the most used 
SIP proxies in the telecommunication industries?



2 - Does the adoption of OpenSIPS, by telecom players, have been 
increasing ?



3 - What are examples of most remarkable use cases with OpenSIPS in 
telecommunications?



Any information will  be very helpful!


Best regards.


RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979


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Re: [OpenSIPS-Users] no transport protocol

2016-10-18 Thread ‪H Yavari‬ ‪
Hi Razvan,
Thank you. You're right. 
Just one point: "port" global parameter is deprecated ? because when I set it, 
I receive syntax error.

Regards,H.Yavari


  From: Răzvan Crainea 
 
   
 Hi, Yavari!
 
 Make sure you are loading at least one protocol module. For example:
 
 loadmodule "proto_udp.so" # loads the UDP protocol module
 loadmodule "proto_tcp.so" # loads the TCP protocol module
 
 Best regards,
 Răzvan
  Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com On 10/18/2016 10:03 AM, ‪H Yavari‬ ‪ wrote:
  
  Hi to all, 
  I installed version 2.2.1, I'm using default configs. But when I start 
Opensips I receive this error: 
  Oct 18 10:29:51 [2569] ERROR:core:main: no trasnport protocol loaded
  
  Can you please give me some hints? 
  "Note: transport spelled wrong too!" 
  
  Regards, H.Yavari
   
  
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Re: [OpenSIPS-Users] OpenSIPS in the market.

2016-10-18 Thread Olle E. Johansson

> On 18 Oct 2016, at 08:46, Bogdan-Andrei Iancu  wrote:
> 
> Hello Rodrigo,
> 
> The questions your are asking are hard to answer and the Open Source world is 
> most of the times opaque when comes to who is using and how much is used. As 
> anyone can simply download and start using (information on how to do it) is 
> quite enough, you may have many cases were companies do use an Open Source 
> software without "leaking" or "leaving traces" about that.
Yes, I agree. These kind of questions are really hard to answer. During the 
last ten years, acceptance of Open Source in production use for
commercial companies have changed worldwide. Open Source proxys like OpenSIPS 
and the relatives (SER, Kamailio) and other code bases
like resiprocate are taking the market together with media servers like Janus, 
Asterisk and FreeSwitch.

> 
> This is also for OpenSIPS project - it is hard to say who is using and how 
> the popularity is fluctuating and mainly because there is no much of a 
> feedback (as it is free to use).
> 
> Still, take a look at :
>  http://www.opensips.org/About/WhoIsUsing 
> 
> 
> This is an attempt to list the companies willing to do it, it is not a full 
> listing.
> 
> Best regards,
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com 
> On 13.10.2016 15:44, Rodrigo Pimenta Carvalho wrote:
>> Dear OpenSIPS users,
>> 
>> As a teacher, I'm going to explain somethings about SIP and SIP proxies to 
>> my students.
>> So, I intend to talk about OpenSIPS too.
>> 
>> 1- Is there some ranking in any web page showing what is the most used SIP 
>> proxies in the telecommunication industries?
The interesting part would be Open Source platforms vs commercial and non-Open 
platforms. But no, I am not either aware of such data.
>> 
>> 2 - Does the adoption of OpenSIPS, by telecom players, have been increasing ?
Yes
>> 
>> 3 - What are examples of most remarkable use cases with OpenSIPS in 
>> telecommunications?
If you widen the scope to “Open Source proxys” I would say that SIP as a 
protocol is making headway in interesting
areas. SIP is part of a European standard for live radio broadcast. It’s also 
part of a standard for air traffic communication,
connecting radio masts to a flight control tower.

I’ve built solutions for both of this standards with Open Source SIP software.

So it’s no longer just about “telecommunications” - SIP and the open source 
software is starting to deliver on
the original SIP vision - a realtime communication platform for Internet users.

To stay in telecommunications there are many presentations from our various 
conferences that will
give you ideas on where and how it is used. The new black is of course Open 
Source mobile network
platforms - after ISDN and SS7 this is the new interesting area to cover with 
Open solutions.

Regards,
/O


>> 
>> Any information will  be very helpful!
>> 
>> Best regards.
>> 
>> RODRIGO PIMENTA CARVALHO
>> Inatel Competence Center
>> Software
>> Ph: +55 35 3471 9200 RAMAL 979
>> 
>> 
>> ___
>> Users mailing list
>> Users@lists.opensips.org 
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users 
>> 
> 
> ___
> Users mailing list
> Users@lists.opensips.org 
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users 
> 



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Re: [OpenSIPS-Users] no transport protocol

2016-10-18 Thread Răzvan Crainea

Yes, it has been remove starting with OpenSIPS 2.1.

Best regards,

Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com

On 10/18/2016 11:02 AM, ‪H Yavari‬ ‪ wrote:

Hi Razvan,

Thank you. You're right.
Just one point: "port" global parameter is deprecated ? because when I 
set it, I receive syntax error.



Regards,
H.Yavari



*From:* Răzvan Crainea 


Hi, Yavari!

Make sure you are loading at least one protocol module. For example:

loadmodule "proto_udp.so" # loads the UDP protocol module
loadmodule "proto_tcp.so" # loads the TCP protocol module

Best regards,
Răzvan
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com 
On 10/18/2016 10:03 AM, ‪H Yavari‬ ‪ wrote:

Hi to all,

I installed version 2.2.1, I'm using default configs. But when I 
start Opensips I receive this error:


Oct 18 10:29:51 [2569] ERROR:core:main: no trasnport protocol loaded

Can you please give me some hints?

"Note: transport spelled wrong too!"


Regards,
H.Yavari


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Re: [OpenSIPS-Users] OpenSIPS in the market.

2016-10-18 Thread Rodrigo Pimenta Carvalho
Hi people.


Thank all of you very much!


I need read more and more about the use of SIP in the industry. This will serve 
as complementary information that I will present in my next class.

So, i will search about that "European standard for live radio broadcast" and 
also about that "standard for air traffic communication,  connecting radio 
masts to a flight control tower". These sound very interesting.


Here, where I work we are developing a intercom (door bell) for a company 
specialized in residential security. The intercom will have the OpenSIPS 
inside. As a SIP proxy, such intercom will connect the house with smartphones, 
for example.


OpenSIPS is really helping me a lot! Because several project requirements I 
have programmed inside the opensips.cfg configuration file.


Thanks for the support!



RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979



De: Olle E. Johansson 
Enviado: terça-feira, 18 de outubro de 2016 06:06
Para: OpenSIPS users mailling list
Cc: Olle E Johansson; Rodrigo Pimenta Carvalho
Assunto: Re: [OpenSIPS-Users] OpenSIPS in the market.


On 18 Oct 2016, at 08:46, Bogdan-Andrei Iancu 
mailto:bog...@opensips.org>> wrote:

Hello Rodrigo,

The questions your are asking are hard to answer and the Open Source world is 
most of the times opaque when comes to who is using and how much is used. As 
anyone can simply download and start using (information on how to do it) is 
quite enough, you may have many cases were companies do use an Open Source 
software without "leaking" or "leaving traces" about that.
Yes, I agree. These kind of questions are really hard to answer. During the 
last ten years, acceptance of Open Source in production use for
commercial companies have changed worldwide. Open Source proxys like OpenSIPS 
and the relatives (SER, Kamailio) and other code bases
like resiprocate are taking the market together with media servers like Janus, 
Asterisk and FreeSwitch.


This is also for OpenSIPS project - it is hard to say who is using and how the 
popularity is fluctuating and mainly because there is no much of a feedback (as 
it is free to use).

Still, take a look at :
 http://www.opensips.org/About/WhoIsUsing

This is an attempt to list the companies willing to do it, it is not a full 
listing.

Best regards,

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

On 13.10.2016 15:44, Rodrigo Pimenta Carvalho wrote:
Dear OpenSIPS users,

As a teacher, I'm going to explain somethings about SIP and SIP proxies to my 
students.
So, I intend to talk about OpenSIPS too.

1- Is there some ranking in any web page showing what is the most used SIP 
proxies in the telecommunication industries?
The interesting part would be Open Source platforms vs commercial and non-Open 
platforms. But no, I am not either aware of such data.

2 - Does the adoption of OpenSIPS, by telecom players, have been increasing ?
Yes

3 - What are examples of most remarkable use cases with OpenSIPS in 
telecommunications?
If you widen the scope to “Open Source proxys” I would say that SIP as a 
protocol is making headway in interesting
areas. SIP is part of a European standard for live radio broadcast. It’s also 
part of a standard for air traffic communication,
connecting radio masts to a flight control tower.

I’ve built solutions for both of this standards with Open Source SIP software.

So it’s no longer just about “telecommunications” - SIP and the open source 
software is starting to deliver on
the original SIP vision - a realtime communication platform for Internet users.

To stay in telecommunications there are many presentations from our various 
conferences that will
give you ideas on where and how it is used. The new black is of course Open 
Source mobile network
platforms - after ISDN and SS7 this is the new interesting area to cover with 
Open solutions.

Regards,
/O



Any information will  be very helpful!

Best regards.

RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979



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Re: [OpenSIPS-Users] Why am I gettin error to terminate long calls and not short ones? Why it is "Ignoring callid"?

2016-10-18 Thread Rodrigo Pimenta Carvalho

Ok.


Thank you very much for the hint.


Best regards.



RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979



De: users-boun...@lists.opensips.org  em nome 
de Răzvan Crainea 
Enviado: terça-feira, 18 de outubro de 2016 05:18
Para: users@lists.opensips.org
Assunto: Re: [OpenSIPS-Users] Why am I gettin error to terminate long calls and 
not short ones? Why it is "Ignoring callid"?

Hi, Rodrigo!

Most likely A closes the connection to OpenSIPS. You can check that by tracing 
the communication between A and OpenSIPS.
In order to solve that, make sure that the TCP keepalive[1] is enabled. Also, 
you can use the dialog pinging[2] feature ('Pp' flag to create_dialog()) to 
keep the dialog connections open.

[1] http://www.opensips.org/Documentation/Script-CoreParameters-2-2#toc103
[2] http://www.opensips.org/html/docs/modules/2.2.x/dialog.html#id295792

Best regards,

Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com

On 10/17/2016 11:20 PM, Rodrigo Pimenta Carvalho wrote:
Dear OpenSIPS users,


In my hardware, with IP = 192.168.0.101, I have OpenSIPS and softphone A. Thre 
is softphone B also, in another hardware.

A calls B.
B accept the call.
After t minutes...B hungs up the call.

In this moment, A enters in a wrong state, because OpenSIPS reports a problem 
and probably due to it the proxy doesn't communicate with softphone A in such 
moment. So, my softphone A considers that the call is not ended.

See what OpenSIPS reports in this moment:

Jan 05 04:14:29 colibri-imx6-jfl opensips[431]: Jan  5 04:14:29 [438] 
INFO:core:probe_max_sock_buff: using snd buffer of 320 kb
Jan 05 04:14:29 colibri-imx6-jfl opensips[431]: Jan  5 04:14:29 [438] 
INFO:core:init_sock_keepalive: TCP keepalive enabled on socket 20
Jan 05 04:14:29 colibri-imx6-jfl opensips[431]: Jan  5 04:14:29 [438] 
ERROR:core:tcpconn_async_connect: poll error: flags 1c
Jan 05 04:14:29 colibri-imx6-jfl opensips[431]: Jan  5 04:14:29 [438] 
ERROR:core:tcpconn_async_connect: failed to retrieve SO_ERROR 
[server=192.168.0.101:57985] (111) Connection refused
Jan 05 04:14:29 colibri-imx6-jfl opensips[431]: Jan  5 04:14:29 [438] 
ERROR:core:proto_tcp_send: async TCP connect failed
Jan 05 04:14:29 colibri-imx6-jfl opensips[431]: Jan  5 04:14:29 [438] 
ERROR:tm:msg_send: send() for proto 2 failed
Jan 05 04:14:29 colibri-imx6-jfl opensips[431]: Jan  5 04:14:29 [438] 
ERROR:tm:t_forward_nonack: sending request failed


If t is just few minutes, let's say 2 minutes, there is no any issue.

However, if t is bigger, let's say 4 minutes, his issue is present.


What is happening here? Can someone give some help, please!

Any hint will be very helpful!




Some more details:

User B is g1r2u3p4o5@192.168.0.102.

User A is 
intercomA_5dtUWgwgqzR6@192.168.0.101.

Callid was "ec4548a8-4207-4fc2-8ed8-81897ff62175".


Before getting such error log, I saw another messages in the log like this:


Jan 05 04:13:28 colibri-imx6-jfl opensips[431]: new branch at 
sip:g1r2u3p4o5@192.168.0.102:61230;transport=TCP;ob
Jan 05 04:13:28 colibri-imx6-jfl opensips[431]: od: invalid option -- 'A'
Jan 05 04:13:28 colibri-imx6-jfl opensips[431]: BusyBox v1.22.1 (2016-03-29 
09:43:20 BRT) multi-call binary.
Jan 05 04:13:28 colibri-imx6-jfl opensips[431]: Usage: od 
[-aBbcDdeFfHhIiLlOovXx] [FILE]
Jan 05 04:13:29 colibri-imx6-jfl opensips[431]: Ignoring callid  
"ec4548a8-4207-4fc2-8ed8-81897ff62175"


Jan 05 04:13:34 colibri-imx6-jfl opensips[431]: Jan  5 04:13:34 [442] 
INFO:core:probe_max_sock_buff: using snd buffer of 320 kb
an 05 04:13:34 colibri-imx6-jfl opensips[431]: Jan  5 04:13:34 [442] 
INFO:core:init_sock_keepalive: TCP keepalive enabled on socket 25

an 05 04:14:29 colibri-imx6-jfl opensips[431]: Jan  5 04:14:29 [438] 
ERROR:dialog:dlg_validate_dialog: failed to validate remote contact: 
dlg=[sip:intercomA_5dtUWgwgqzR6@127.0.0.1:36427;transport=TCP;ob]
 , 
req=[sip:intercomA_5dtUWgwgqzR6@192.168.0.101:57985;transport=TCP;ob]
Jan 05 04:14:29 colibri-imx6-jfl opensips[431]: In-Dialog BYE from 
192.168.0.102 (callid=ec4548a8-4207-4fc2-8ed8-81897ff62175) is not valid 
according to dialog




Best regards.



RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979



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Re: [OpenSIPS-Users] Pending OpenSIPS minor releases: Last minute bug fixes!

2016-10-18 Thread Ramachandran, Agalya (Contractor)
Hi Liviu,

Should I enable debug logs when capturing the OpenSIPS logs?  I have logs only 
for INFO message. i.e log level is 3 in my case.
Also is there any FTP site available to upload the logs and tcpdump? Because in 
e-mail it doesn’t allow to send more than 40 KB.

Regards,
Agalya


From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Liviu Chircu
Sent: Monday, October 17, 2016 12:09 PM
To: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] Pending OpenSIPS minor releases: Last minute bug 
fixes!


Hi, Agalya!

Thank you for including the POST method in your tests as well. In order to 
understand more about the problem, we need to print some information, in order 
to correlate the SIP flow of those calls with the fact that resume_route was 
not called, although the HTTP requests are correctly handled. Could you please 
do the following info to your testing framework, and attempt to replicate the 
missing resume_routes again?

- trace all traffic (pcap format) on your server's SIP port (e.g. 5060). 
Example command: "tcpdump -i  -s 5060 -w "

- in your OpenSIPS script:

* for each INVITE, print the $ci variable

* for each resume_route, print the $ci variable again

If, by any chance, all the above information is already available, could you 
please email the pcap and problematic Call-IDs to 
li...@opensips.org, so I can take a look? Thanks!

Best regards,

Liviu Chircu

OpenSIPS Developer

http://www.opensips-solutions.com
On 17.10.2016 18:33, Ramachandran, Agalya (Contractor) wrote:
Hi Razvan/Liviu,

My test observance as follows:

REST_PUT:

1)  In the sipp client if I have pause value after 200 OK between 12 to 24 
sec, am seeing that for ~15 calls resume_route is not called.

2)  I have increased pause time to  24 -45 sec and also 180-240 sec. This 
both cases,  call setup is complete. resume_route is executed for all 100K 
calls.
REST_POST:

3)  In the sipp client, pause of 12 to 24 sec after 200OK. For 10 calls out 
of 100K calls, resume_route is not called.

This happened only once out of  three times. Other 2 times call set is 
complete. REST_POST is returning 4xx in my case.

4)  In the sipp client, pause of 24 to 45 sec after 200OK.Call set up is 
complete.


Regards,
Agalya

From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Razvan Crainea
Sent: Thursday, October 13, 2016 10:16 AM
To: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] Pending OpenSIPS minor releases: Last minute bug 
fixes!

Hi, Agalya!

Can you try running the tests with the stock code, without the PUT API code? 
Just run the tests with the POST method, to confirm whether there is a bug in 
the stock code, or in the custom code you made.

Best regards,



Răzvan Crainea

OpenSIPS Solutions

www.opensips-solutions.com
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Re: [OpenSIPS-Users] Pending OpenSIPS minor releases: Last minute bug fixes!

2016-10-18 Thread Liviu Chircu

> Should I enable debug logs when capturing the OpenSIPS logs?

I'd just stick with INFO, otherwise the log file might grow too much.

> Also is there any FTP site available to upload the logs and tcpdump?

Try pastebin.comfor logs. I'm fairly sure Dropbox and Google Driveboth 
have some sort of public upload feature as well.


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

On 18.10.2016 17:31, Ramachandran, Agalya (Contractor) wrote:


Hi Liviu,

Should I enable debug logs when capturing the OpenSIPS logs?  I have 
logs only for INFO message. i.e log level is 3 in my case.


Also is there any FTP site available to upload the logs and tcpdump? 
Because in e-mail it doesn’t allow to send more than 40 KB.


Regards,

Agalya

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

*Sent:* Monday, October 17, 2016 12:09 PM
*To:* users@lists.opensips.org
*Subject:* Re: [OpenSIPS-Users] Pending OpenSIPS minor releases: Last 
minute bug fixes!


Hi, Agalya!

Thank you for including the POST method in your tests as well. In 
order to understand more about the problem, we need to print some 
information, in order to correlate the SIP flow of those calls with 
the fact that resume_route was not called, although the HTTP requests 
are correctly handled. Could you please do the following info to your 
testing framework, and attempt to replicate the missing resume_routes 
again?


- trace all traffic (pcap format) on your server's SIP port (e.g. 
5060). Example command: "tcpdump -i  -s 5060 -w "


- in your OpenSIPS script:

* for each INVITE, print the $ci variable

* for each resume_route, print the $ci variable again

If, by any chance, all the above information is already available, 
could you please email the pcap and problematic Call-IDs to 
li...@opensips.org , so I can take a look? 
Thanks!


Best regards,

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

On 17.10.2016 18:33, Ramachandran, Agalya (Contractor) wrote:

Hi Razvan/Liviu,

My test observance as follows:

REST_PUT:

1)In the sipp client if I have pause value after 200 OK between 12
to 24 sec, am seeing that for ~15 calls resume_route is not called.

2)I have increased pause time to  24 -45 sec and also 180-240 sec.
This both cases,  call setup is complete. resume_route is executed
for all 100K calls.

REST_POST:

3)In the sipp client, pause of 12 to 24 sec after 200OK. For 10
calls out of 100K calls, resume_route is not called.

This happened only once out of  three times. Other 2 times call
set is complete. REST_POST is returning 4xx in my case.

4)In the sipp client, pause of 24 to 45 sec after 200OK.Call set
up is complete.

Regards,

Agalya

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

[mailto:users-boun...@lists.opensips.org] *On Behalf Of *Razvan
Crainea
*Sent:* Thursday, October 13, 2016 10:16 AM
*To:* users@lists.opensips.org 
*Subject:* Re: [OpenSIPS-Users] Pending OpenSIPS minor releases:
Last minute bug fixes!

Hi, Agalya!

Can you try running the tests with the stock code, without the PUT
API code? Just run the tests with the POST method, to confirm
whether there is a bug in the stock code, or in the custom code
you made.

Best regards,


Răzvan Crainea

OpenSIPS Solutions

www.opensips-solutions.com 



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[OpenSIPS-Users] little help with this bug

2016-10-18 Thread Dani Popa
Hi,

I need a little help with this bug:

 gdb /usr/local/unified_opensips/sbin/opensips /core
GNU gdb (GDB) 7.4.1-debian
Copyright (C) 2012 Free Software Foundation, Inc.
License GPLv3+: GNU GPL version 3 or later 
This is free software: you are free to change and redistribute it.
There is NO WARRANTY, to the extent permitted by law.  Type "show copying"
and "show warranty" for details.
This GDB was configured as "i486-linux-gnu".
For bug reporting instructions, please see:
...
Reading symbols from /usr/local/unified_opensips/sbin/opensips...done.
[New LWP 18111]
[New LWP 18112]
[New LWP 17585]

warning: Can't read pathname for load map: Input/output error.
[Thread debugging using libthread_db enabled]
Using host libthread_db library
"/lib/i386-linux-gnu/i686/cmov/libthread_db.so.1".
Core was generated by `/usr/local/unified_opensips/sbin/opensips -f
/usr/local/unified_opensips/etc/op'.
Program terminated with signal 6, Aborted.
#0  0xb77e2424 in __kernel_vsyscall ()
(gdb) bt
#0  0xb77e2424 in __kernel_vsyscall ()
#1  0xb7468661 in raise () from /lib/i386-linux-gnu/i686/cmov/libc.so.6
#2  0xb746ba92 in abort () from /lib/i386-linux-gnu/i686/cmov/libc.so.6
#3  0x081404a7 in qm_free (qm=0xb7222008, p=p@entry=0xb729571c,
file=file@entry=0x8203506 "parser/hf.c", func=func@entry=0x820366a
"free_hdr_field_lst", line=line@entry=243) at mem/q_malloc.c:448
#4  0x08143a9e in free_hdr_field_lst (hf=0xb7295680, hf@entry=0xb729571c)
at parser/hf.c:243
#5  0xb71c8d86 in mi_tm_uac_dlg (cmd_tree=0xb72942ec, param=0x0) at mi.c:539
#6  0xb6985fc5 in run_mi_cmd (param=0xb485c150, f=,
t=0xb72942ec, cmd=) at ../../mi/mi.h:109
#7  default_method (env=0xb485c150, host=0xb485c270 "\001",
methodName=, paramArray=0xa2b8880, serverInfo=0x0) at
xr_server.c:223
#8  0xb68c2ac7 in xmlrpc_dispatchCall () from /usr/lib/libxmlrpc_server.so.3
#9  0xb68c2c16 in xmlrpc_registry_process_call2 () from
/usr/lib/libxmlrpc_server.so.3
#10 0xb68c96e0 in ?? () from /usr/lib/libxmlrpc_server_abyss.so.3
#11 0xb68b7f57 in ?? () from /usr/lib/libxmlrpc_abyss.so.3
#12 0xb68b1e30 in ?? () from /usr/lib/libxmlrpc_abyss.so.3
#13 0xb68bac93 in ?? () from /usr/lib/libxmlrpc_abyss.so.3
#14 0xb6e0bc39 in start_thread () from
/lib/i386-linux-gnu/i686/cmov/libpthread.so.0
#15 0xb7514c6e in clone () from /lib/i386-linux-gnu/i686/cmov/libc.so.6
(gdb) quit
root@sp01:/# cd /usr/local/^C
root@sp01:/# cd /home/openips/opensips_1_11/
root@sp01:/home/openips/opensips_1_11# git pull
remote: Counting objects: 43, done.
remote: Compressing objects: 100% (43/43), done.
remote: Total 43 (delta 14), reused 0 (delta 0), pack-reused 0
Unpacking objects: 100% (43/43), done.
>From https://github.com/OpenSIPS/opensips
   f880642..66ae29f  1.11   -> origin/1.11
   1dfeb25..522a9e3  2.1-> origin/2.1
   9ed154b..757389d  2.2-> origin/2.2
   7e2d6e4..c63e14d  master -> origin/master
Updating f880642..66ae29f
Fast-forward
 modules/sst/sst_handlers.c |   42
+-
 modules/sst/sst_handlers.h |1 +
 2 files changed, 34 insertions(+), 9 deletions(-)
root@sp01:/home/openips/opensips_1_11#


Thanks,
-- 
Dani Popa
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