Re: [OpenSIPS-Users] Remote ip change in-dialog

[Saioa Perurena]

We have our own sip client ...


On 10/10/16 17:02, Russell Treleaven wrote:
> What mobile sip client supports this?
>
>
> On Mon, Oct 10, 2016 at 6:50 AM, Saioa Perurena
> > wrote:
>
> Hi,
>
> I've the following problem, any advice will be welcome.
>
> A calls B, A changes his ip because of network change (3G to wifi for
> example) sends again an INVITE and the call continous ok with the same
> callid, did, tags... but if B sends and INVITE (to enable video for
> example), opensips validate_dialog returns error because the ip
> has changed.
>
> I've tried fix_route_dialog() but it doesn't work.
>
> Thanks in advance!
>
> Saioa.
>
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>
>
>
>
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Re: [OpenSIPS-Users] OpensSIPS 2.2 - avps with an index of -1

[Pat Burke]

Bogdan,


I implemented your change in my 2.2.1 code, and it works great.  


Thanks for the implementation.


Pat


-Original Message-
From: "Bogdan-Andrei Iancu" 
To: "Pat Burke" , "OpenSIPS users mailling list" 

Date: 10/13/16 05:31
Subject: Re: [Spam] Re: [OpenSIPS-Users] OpensSIPS 2.2 - avps with an index of 
-1

Hi Pat,

OK :).

Well, the usage of the -1 index is inconsistent with the rest of the indexes - 
for positive indexes are used to "overwrite" an existing value, not to add a 
new one. This is why the usage of -1 to add may be misleading, as you would 
expect to actually overwrite the last value :).

I committed a patch on trunk/2.3 that adds the "append" index for AVPs :
$(avp(name)[append]) = "last value";

https://github.com/OpenSIPS/opensips/commit/466fbf045179188c39fefe5b48ec581c37023a5b

You may consider using it (it is fully 2.2 compatible) and starting with 2.3 
version you will get rid of additional patches ;).

Regards,

Bogdan-Andrei Iancu OpenSIPS Founder and 
Developerhttp://www.opensips-solutions.com
On 12.10.2016 18:02, Pat Burke wrote:


Hello Bogdan,


You are exactly correct.  We have a patch in pv_set_avp in pvar.c that allows 
-1.  I don't know who put that in, but it works great.  I will put the patch 
into my 2.2 code.


The method was updated to
if(idxf == PV_IDX_INT || idxf == PV_IDX_PVAR) /* if the avp is indexed 
*/
{
if (idx == -1) { 
if (add_avp_last(flags, avp_name, avp_val)<0) {
LM_ERR("error - cannot add AVP\n");
goto error;
}
} else if(replace_avp(flags, avp_name, avp_val, idx)< 0) {
LM_ERR("Failed to replace avp\n");
goto error;
}
}
  


from


if(idxf == PV_IDX_INT || idxf == PV_IDX_PVAR) /* if the avp is indexed 
*/
{
if(replace_avp(flags, avp_name, avp_val, idx)< 0)
{
LM_ERR("Failed to replace avp\n");
goto error;
}
}
  
Thanks again,
Pat Burke




-Original Message-
From: "Bogdan-Andrei Iancu" 
To: "OpenSIPS users mailling list" , "Pat Burke" 

Date: 10/12/16 03:47
Subject: [Spam] Re: [OpenSIPS-Users] OpensSIPS 2.2 - avps with an index of -1

Hi Pat,

Are you sure the -1 index works for you? I just tested with an 1.11 code and 
got:

Oct 12 11:35:12 [29913] ERROR:core:replace_avp: Index with negative value
Oct 12 11:35:12 [29913] ERROR:core:pv_set_avp: Failed to replace avp
Oct 12 11:35:12 [29913] ERROR:core:do_assign: setting PV failed

I don't think negative indexes were ever supported for AVPs.

Are you sure you do not have a patch for 1.11 to allow negative values ?

Regards,

Bogdan-Andrei Iancu OpenSIPS Founder and 
Developerhttp://www.opensips-solutions.com
On 12.10.2016 03:04, Pat Burke wrote:

In OpenSIPS 1.11 you were able to use the index of -1 on AVP's to put items 
into an AVP as a FIFO instead of a LIFO.

Here is what I mean.


$(avp(list)[-1]) = 1;
$(avp(list)[-1]) = 2;

$(avp(list)[-1]) = 3;
$(avp(list)[-1]) = 4;
$(avp(list)[-1]) = 5;


this would result the values coming off as a FIFO
$avp(list) is 1
$avp(list) = NULL;
$avp(list) is 2
$avp(list) = NULL;
$avp(list) is 3
$avp(list) = NULL;
$avp(list) is 4
$avp(list) = NULL;
$avp(list) is 5






Just pushing the items onto the AVP creates a LIFO
$avp(list) = 1;
$avp(list) = 2;
$avp(list) = 3;
$avp(list) = 4;
$avp(list) = 5;


this would result the values coming off as
$avp(list) is 5
$avp(list) = NULL;
$avp(list) is 4
$avp(list) = NULL;
$avp(list) is 3
$avp(list) = NULL;
$avp(list) is 2
$avp(list) = NULL;
$avp(list) is 1




So my question is how do I create a FIFO without having to keep an index to 
ensure that I am adding the item to the end. When porting the code from 1.11 to 
2.2 I get an error "ERROR:core:replace_avp: Index with negative value" when I 
use the -1.




Regards,
Pat Burke






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Re: [OpenSIPS-Users] Latest AVP's in the cdr when call fails in branch

[Pat Burke]

Hi Bogdan,


I got it figured out.  It was a series of configuration issues and not 
realizing how some of the new cdr process worked.  


1)  The branching was just bad timing on my part.  It is working fine.  The 
issue I saw just happened to be after I implemented branching.


2) The real issue was in the call to do_accouting, I set the cdr and 
missed_call flags, but not the failed flag.


Where I got confused was that the acc table had the correct record when the 
call completed successfully.  If there was an error and the call didn't 
complete successfully, there was a record in the missed_call table that didn't 
have the final avp values.   I did notice that after putting the failed flag in 
place and having a call fail, the acc record is correct (contains the lastest 
avp values)  but the missed_call record does not contain all of the lastest avp 
values.




I do have one more question on the migration from 1.11 to 2.2.


        In  1.11 we had the following set
             modparam("dispatcher", "flags", 2)
             modparam("dispatcher", "force_dst", 1)


        in 2.2, I use the following in 
             ds_select_dst("$var(server_group_carrier_setid)","6","FS")


Do I have the parameters correct in the ds_select_dst call?


Thanks,
Pat

-Original Message-
From: "Bogdan-Andrei Iancu" 
To: "OpenSIPS users mailling list" , "Pat Burke" 

Date: 10/13/16 06:10
Subject: Re: [OpenSIPS-Users] Latest AVP's in the cdr when call fails in branch

Hi Pat,

It is a bit confusion and the most of the report is about adding an AVP as hdr 
via append_hf() , but you also mention something about CDR...and I do not see 
the relation between the two.

First of all, is the append_hf() working fine ? is the header (in the 
sequential attempts) containing the correct aggregated value ?

Best regards,

Bogdan-Andrei Iancu OpenSIPS Founder and 
Developerhttp://www.opensips-solutions.com
On 12.10.2016 22:59, Pat Burke wrote:


Hello,


I am in process of migrating from 1.11 to 2.2.  We have a situation where we 
have a routing opensips that can make multiple attempts to complete a call.  
The information specific to each attempt is passed to the SBC opensips through 
headers.  In order to not double up on headers, I moved the code that calls 
append_hf to a branch.  As the call progresses, I update various AVP's and 
print them into the log (so I can see that they are being updated correctly).  
Below is an example scenario of what I am doing.






initial call setup logic
set the t_on_failure


setup for first carrier 
branch 
   setup headers with carrier information used by the SBC
   $avp(test) := "Step 1;";
t_relay


on_failure (for example 480 - timeout)
   setup for the next carrier
   branch
  setup header for next carrier information used by the SBC
  $avp(test) := $avp(test) + "Step 2;";  # avp(test) now equal "Step 1;Step 
2;"
t_relay



on_failure (for example 408 - timeout)
   no more carriers, end the call






What gets put into the cdr is "Step 1;" not "Step 1;Step 2;"


Also, the sip_code is set to the first failure received from the SBC, not the 
last.


If the call completes successfully, the cdr data is correct.


Thanks,
Pat Burke



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

[Răzvan Crainea]

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 customcode you made.


Best regards,

Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com

On 10/13/2016 04:55 PM, Ramachandran, Agalya (Contractor) wrote:


Hi Liviu,

Thanks for sharing the process for source code contribution.

Please find my response inline.

Just to be sure: we're talking about REST POST now, right?

Agalya: No. Am using REST PUT API.

Regarding the issue: are there any OpenSIPS log errors that might help us?

Agalya: No errors in the logs.

Also, did you deduce the 99986 number by grepping the logs, or by 
looking at the completed SIP calls? I'd recommend the latter, since 
log lines may be rate-limited / overlapped, etc.


Agalya: By grepping logs as well by looking completed SIP calls. I did 
this for 50,000 calls and the number matched for SIP logs and call 
completed in sipp.


In Wireshark, I see response for all 50,000 HTTP requests.

I couldn’t get complete logs for 100K calls may be because as you said 
rate-limited / overlapped, etc.



Regards,
Agalya

*From:*Liviu Chircu [mailto:li...@opensips.org]
*Sent:* Thursday, October 13, 2016 4:10 AM
*To:* Ramachandran, Agalya (Contractor) 
; OpenSIPS users mailling list 

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


Hi, Agalya!

If I get green signal from my management, I will contribute code
for REST_PUT. Can you share me  the process to contribute code ?

First, you create a GitHub account. Then you fork the OpenSIPS repo 
[1] to your account. This allows you to work on it independently and 
push changes back to GitHub when you're done, so they are visible for 
everyone.


Once the fork is done, you clone the forked project on your machine so 
you can work on its code. You then apply your custom patch(es), make 
the necessary commits and push these changes back to GitHub.


The process of proposing the merge of a forked project back into the 
main project is called a "Pull Request" [2]. This is the final step of 
contributing code, and you can easily do it with a few clicks via 
GitHub's web interface.


Out of 2 times, I tested I observed the below issue for once.
Before I used to have it for every test.

1.Tried to load 100,000 calls - But route[resume_http] is called
only for 99986 calls.

Every time approximately 10-20 calls, route[resume_http]  is not
called. But if I see the tcpdump, I am seeing 100,000 HTTP request
and 100,000 HTTP 200 OK responses.

When printing the response in resume_http for every call-id, 10-20
calls response is not printed - which means resume is not called
for these calls.

Am not filtering any response code.

Any clue on this one?


Just to be sure: we're talking about REST POST now, right?

Regarding the issue: are there any OpenSIPS log errors that might help 
us? Also, did you deduce the 99986 number by grepping the logs, or by 
looking at the completed SIP calls? I'd recommend the latter, since 
log lines may be rate-limited / overlapped, etc.


[1]: https://github.com/OpenSIPS/opensips
[2]: https://github.com/OpenSIPS/opensips/pulls



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

[Ramachandran, Agalya (Contractor)]

Hi Liviu,

Thanks for sharing the process for source code contribution.
Please find my response inline.

Just to be sure: we're talking about REST POST now, right?
Agalya: No. Am using REST PUT API.

Regarding the issue: are there any OpenSIPS log errors that might help us?
Agalya: No errors in the logs.

Also, did you deduce the 99986 number by grepping the logs, or by looking at 
the completed SIP calls? I'd recommend the latter, since log lines may be 
rate-limited / overlapped, etc.
Agalya: By grepping logs as well by looking completed SIP calls. I did this for 
50,000 calls and the number matched for SIP logs and call completed in sipp.
In Wireshark, I see response for all 50,000 HTTP requests.
I couldn't get complete logs for 100K calls may be because as you said 
rate-limited / overlapped, etc.

Regards,
Agalya

From: Liviu Chircu [mailto:li...@opensips.org]
Sent: Thursday, October 13, 2016 4:10 AM
To: Ramachandran, Agalya (Contractor) ; 
OpenSIPS users mailling list 
Subject: Re: [OpenSIPS-Users] Pending OpenSIPS minor releases: Last minute bug 
fixes!


Hi, Agalya!
If I get green signal from my management, I will contribute code for REST_PUT. 
Can you share me  the process to contribute code ?
First, you create a GitHub account. Then you fork the OpenSIPS repo [1] to your 
account. This allows you to work on it independently and push changes back to 
GitHub when you're done, so they are visible for everyone.

Once the fork is done, you clone the forked project on your machine so you can 
work on its code. You then apply your custom patch(es), make the necessary 
commits and push these changes back to GitHub.

The process of proposing the merge of a forked project back into the main 
project is called a "Pull Request" [2]. This is the final step of contributing 
code, and you can easily do it with a few clicks via GitHub's web interface.

Out of 2 times, I tested I observed the below issue for once. Before I used to 
have it for every test.

1.   Tried to load 100,000 calls - But route[resume_http] is called only 
for 99986 calls.
Every time approximately 10-20 calls, route[resume_http]  is not called. But if 
I see the tcpdump, I am seeing 100,000 HTTP request and 100,000 HTTP 200 OK 
responses.
When printing the response in resume_http for every call-id, 10-20 calls 
response is not printed - which means resume is not called for these calls.
Am not filtering any response code.

Any clue on this one?

Just to be sure: we're talking about REST POST now, right?

Regarding the issue: are there any OpenSIPS log errors that might help us? 
Also, did you deduce the 99986 number by grepping the logs, or by looking at 
the completed SIP calls? I'd recommend the latter, since log lines may be 
rate-limited / overlapped, etc.

[1]: https://github.com/OpenSIPS/opensips
[2]: https://github.com/OpenSIPS/opensips/pulls
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[OpenSIPS-Users] OpenSIPS in the market.

[Rodrigo Pimenta Carvalho]

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] opensips 2.1 call_center queue position

[Jonathan Hunter]

Hi Bogdan,
Thanks for the response, and yes I understand you are envisioning a solution, 
which I appreciate as usual :)
Yes the addition of cc_pos would work for me, as I can extract any number of 
variables form the ruri, this would do the job.
Thanks
Jon

Subject: Re: [OpenSIPS-Users] opensips 2.1 call_center queue position
To: hunter...@hotmail.com; users@lists.opensips.org
From: bog...@opensips.org
Date: Thu, 13 Oct 2016 13:59:40 +0300


  

  
  
Hi Jonathan,

  

  No, currently this is not possible. I was trying to envision a solution
  for your need.

  

  But, checking the code, it is really difficult to add the headers
  to the INVITEs originated by OpenSIPS (via the B2BUA), as we need
  some flexibility (different headers to different INVITEs belonging
  to the same B2B scenario , and even more, we need to traverse
  couple of internal APIs - to propagate the hdrs from Call center
  module all the way to TM).

  

  So, a simpler approach may be to add such extra info as URI params
  to the RURI. Like if you have the RURI
  "sip:queue@192.168.1.10:5060" for the queue/waiting playback, the
  RURI in the INVITE to the media server will look like :  
sip:queue@192.168.1.10:5060;cc_eta=40;cc_pos=10 
  - cc_eta being the estimated time to wait in seconds and cc_pos
  the position in the queue.

  

  What do you think of this ?

  

  Regards,


Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 12.10.2016 17:21, Jonathan Hunter
  wrote:



  
  Hi Bogdan,



Yes being able to grab the queue position would be perfect.



Is that possible?



Thanks



Jon

  

  
Subject: Re: [OpenSIPS-Users] opensips
2.1 call_center queue position

To: hunter...@hotmail.com; users@lists.opensips.org

From: bog...@opensips.org

Date: Wed, 12 Oct 2016 15:42:43 +0300



Hi Jonathan,

  

  When a call is mapped to a flow / queue (before playing
  the welcome message), we know the ETA (estimated time to
  wait) and when is placed in the queue (before playing the
  queuing) we internally know the position in the queue.

  

  Would it help to have the position in the queue placed
  into a custome SIP header, when sending the INVITE to the
  message_queue URL ? or to the welcome message ?

  

  Regards,


Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 12.10.2016 12:06,
  Jonathan Hunter wrote:



  
  Hello Bogdan,



Thanks for the response.



In terms of my question, with a number of queuing
  platforms, they have the capability to tell the
  caller, what position they are in , and when they are
  likely to be answered.



I just wondered if this logic was already within
  the module, or if I would need to use an external
  code/script to facilitate this function?



As I presume call_center tracks the number of calls
  currently in a queue ? I would then want to be able to
  extract that information, and if a caller was for
  example in 3rd place in a queue, I could inject the
  relevant audio from freeswitch to tell them their
  current position?



Does that make sense? :)   Just wanted to know if
  its something this module can do?



Thanks



Jon

  

  
Subject: Re:
[OpenSIPS-Users] opensips 2.1 call_center queue
position

To: users@lists.opensips.org;
hunter...@hotmail.com

From: bog...@opensips.org

Date: Wed, 12 Oct 2016 11:23:45 +0300



Hello Jon,

  

  The message_queue is a SIP URI pointing to an
  audio announcement to play to roll of the
  waiting/in-queue playback. This needs to be an
  announcements that never ends (from the
  perspective of the media 

[OpenSIPS-Users] Softphone with by-default AutoAnswer active?

[Mandar Khire]

Hi,
I know that This is not Asterisk related question but then also I ask this
question here due to opensips users know about softphones & here lots of
user present.

Question:-

I am looking for Softphone which work on Windows platform.

Softphone must have 'default AutoAnswer on'.

Means example:- When I install softphone, it does not have any SIP
registration. But it has some by-default settings.

In that I am looking active AutoAnswer option.

I tried Ekiga, Linphone, mizuphone, x-lite etc but all have default
AutoAnswer off.

I have to click on settings & active it.

So all these softphones not useful for me.

Need help.
Mandar P. Khire
+919769419340
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Re: [OpenSIPS-Users] Latest AVP's in the cdr when call fails in branch

[Bogdan-Andrei Iancu]

Hi Pat,

It is a bit confusion and the most of the report is about adding an AVP 
as hdr via append_hf() , but you also mention something about CDR...and 
I do not see the relation between the two.


First of all, is the append_hf() working fine ? is the header (in the 
sequential attempts) containing the correct aggregated value ?


Best regards,

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

On 12.10.2016 22:59, Pat Burke wrote:

Hello,

I am in process of migrating from 1.11 to 2.2.  We have a situation 
where we have a routing opensips that can make multiple attempts to 
complete a call.  The information specific to each attempt is passed 
to the SBC opensips through headers.  In order to not double up on 
headers, I moved the code that calls append_hf to a branch.  As the 
call progresses, I update various AVP's and print them into the log 
(so I can see that they are being updated correctly).  Below is an 
example scenario of what I am doing.




initial call setup logic
set the t_on_failure

setup for first carrier
branch
   setup headers with carrier information used by the SBC
   $avp(test) := "Step 1;";
t_relay

on_failure (for example 480 - timeout)
   setup for the next carrier
   branch
  setup header for next carrier information used by the SBC
  $avp(test) := $avp(test) + "Step 2;";  # avp(test) now equal 
"Step 1;Step 2;"

t_relay

on_failure (for example 408 - timeout)
   no more carriers, end the call



What gets put into the cdr is "Step 1;" not "Step 1;Step 2;"

Also, the sip_code is set to the first failure received from the SBC, 
not the last.


If the call completes successfully, the cdr data is correct.

Thanks,
Pat Burke


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Re: [OpenSIPS-Users] opensips 2.1 call_center queue position

[Bogdan-Andrei Iancu]

Hi Jonathan,

No, currently this is not possible. I was trying to envision a solution 
for your need.


But, checking the code, it is really difficult to add the headers to the 
INVITEs originated by OpenSIPS (via the B2BUA), as we need some 
flexibility (different headers to different INVITEs belonging to the 
same B2B scenario , and even more, we need to traverse couple of 
internal APIs - to propagate the hdrs from Call center module all the 
way to TM).


So, a simpler approach may be to add such extra info as URI params to 
the RURI. Like if you have the RURI "sip:queue@192.168.1.10:5060" for 
the queue/waiting playback, the RURI in the INVITE to the media server 
will look like : sip:queue@192.168.1.10:5060;cc_eta=40;cc_pos=10 - 
cc_eta being the estimated time to wait in seconds and cc_pos the 
position in the queue.


What do you think of this ?

Regards,

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

On 12.10.2016 17:21, Jonathan Hunter wrote:

Hi Bogdan,

Yes being able to grab the queue position would be perfect.

Is that possible?

Thanks

Jon


Subject: Re: [OpenSIPS-Users] opensips 2.1 call_center queue position
To: hunter...@hotmail.com; users@lists.opensips.org
From: bog...@opensips.org
Date: Wed, 12 Oct 2016 15:42:43 +0300

Hi Jonathan,

When a call is mapped to a flow / queue (before playing the welcome 
message), we know the ETA (estimated time to wait) and when is placed 
in the queue (before playing the queuing) we internally know the 
position in the queue.


Would it help to have the position in the queue placed into a custome 
SIP header, when sending the INVITE to the message_queue URL ? or to 
the welcome message ?


Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 12.10.2016 12:06, Jonathan Hunter wrote:

Hello Bogdan,

Thanks for the response.

In terms of my question, with a number of queuing platforms, they
have the capability to tell the caller, what position they are in
, and when they are likely to be answered.

I just wondered if this logic was already within the module, or if
I would need to use an external code/script to facilitate this
function?

As I presume call_center tracks the number of calls currently in a
queue ? I would then want to be able to extract that information,
and if a caller was for example in 3rd place in a queue, I could
inject the relevant audio from freeswitch to tell them their
current position?

Does that make sense? :)   Just wanted to know if its something
this module can do?

Thanks

Jon


Subject: Re: [OpenSIPS-Users] opensips 2.1 call_center queue position
To: users@lists.opensips.org ;
hunter...@hotmail.com 
From: bog...@opensips.org 
Date: Wed, 12 Oct 2016 11:23:45 +0300

Hello Jon,

The message_queue is a SIP URI pointing to an audio announcement
to play to roll of the waiting/in-queue playback. This needs to be
an announcements that never ends (from the perspective of the
media server); only the the OpenSIPS Queue may terminate the
playback, when it decides to take out the call from waiting and to
deliver it to an agent.

As for your question, I'm not sure I understand what you mean by
"inject a message with queue position for the caller in question"
- could you detail please ?

Regards,

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

On 11.10.2016 13:36, Jonathan Hunter wrote:

Hi guys,

I have implemented an opensips/freeswitch environment, and I
wish to add call queues to it, and I like the look of
call_center, so just checking this out in comparison to
mod_callcenter in FS world.

My main question is if using the call_center module if you can
inject a message with queue position for the caller in
question, as I cant see that in documentation, I only see
message_queue which I assume could be used to report the
callers position, but just wondered if anyone has done this
and if they could give me some tips as to if possible?

Many thanks

Jon


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

[Liviu Chircu]

Hi, Agalya!

If I get green signal from my management, I will contribute code for 
REST_PUT. Can you share me  the process to contribute code ?
First, you create a GitHub account. Then you fork the OpenSIPS repo [1] 
to your account. This allows you to work on it independently and push 
changes back to GitHub when you're done, so they are visible for everyone.


Once the fork is done, you clone the forked project on your machine so 
you can work on its code. You then apply your custom patch(es), make the 
necessary commits and push these changes back to GitHub.


The process of proposing the merge of a forked project back into the 
main project is called a "Pull Request" [2]. This is the final step of 
contributing code, and you can easily do it with a few clicks via 
GitHub's web interface.
Out of 2 times, I tested I observed the below issue for once. Before I 
used to have it for every test.


 1. Tried to load 100,000 calls - But route[resume_http] is called
only for 99986 calls.

Every time approximately 10-20 calls, route[resume_http]  is not 
called. But if I see the tcpdump, I am seeing 100,000 HTTP request and 
100,000 HTTP 200 OK responses.
When printing the response in resume_http for every call-id, 10-20 
calls response is not printed - which means resume is not called for 
these calls.

Am not filtering any response code.
Any clue on this one?


Just to be sure: we're talking about REST POST now, right?

Regarding the issue: are there any OpenSIPS log errors that might help 
us? Also, did you deduce the 99986 number by grepping the logs, or by 
looking at the completed SIP calls? I'd recommend the latter, since log 
lines may be rate-limited / overlapped, etc.


[1]: https://github.com/OpenSIPS/opensips
[2]: https://github.com/OpenSIPS/opensips/pulls
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