[sr-dev] git:master:1018b363: modules: readme files regenerated - rtp_media_server ... [skip ci]

2019-02-22 Thread Kamailio Dev
Module: kamailio
Branch: master
Commit: 1018b3631ac629ca79ccade7d36d35af273a3267
URL: 
https://github.com/kamailio/kamailio/commit/1018b3631ac629ca79ccade7d36d35af273a3267

Author: Kamailio Dev 
Committer: Kamailio Dev 
Date: 2019-02-23T00:01:34+01:00

modules: readme files regenerated - rtp_media_server ... [skip ci]

---

Modified: src/modules/rtp_media_server/README

---

Diff:  
https://github.com/kamailio/kamailio/commit/1018b3631ac629ca79ccade7d36d35af273a3267.diff
Patch: 
https://github.com/kamailio/kamailio/commit/1018b3631ac629ca79ccade7d36d35af273a3267.patch


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[sr-dev] git:master:4103e6b0: rtp_media_server: proper handling on CANCEL

2019-02-22 Thread Julien Chavanton
Module: kamailio
Branch: master
Commit: 4103e6b0043b5befb3bd9043629ec6a4d1112e29
URL: 
https://github.com/kamailio/kamailio/commit/4103e6b0043b5befb3bd9043629ec6a4d1112e29

Author: Julien Chavanton 
Committer: Julien Chavanton 
Date: 2019-02-22T22:55:12Z

rtp_media_server: proper handling on CANCEL

 while bridging.
 doc update for bridge command

---

Modified: src/modules/rtp_media_server/config_example/kamailio.cfg
Modified: src/modules/rtp_media_server/doc/design_notes.txt
Modified: src/modules/rtp_media_server/doc/rtp_media_server.xml
Modified: src/modules/rtp_media_server/doc/rtp_media_server_admin.xml
Modified: src/modules/rtp_media_server/rtp_media_server.c

---

Diff:  
https://github.com/kamailio/kamailio/commit/4103e6b0043b5befb3bd9043629ec6a4d1112e29.diff
Patch: 
https://github.com/kamailio/kamailio/commit/4103e6b0043b5befb3bd9043629ec6a4d1112e29.patch


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Re: [sr-dev] git:master:4b7e6089: modules: readme files regenerated - rtp_media_server ... [skip ci]

2019-02-22 Thread Julien Chavanton
 I can see there was an error, I think I can not see the details, not sure
how to reproduce, I will try


All checks have failed
1 failing check

ci/dockercloud (/pkg/docker/alpine) — Your image failed to build in Docker
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Re: [sr-dev] git:master:4b7e6089: modules: readme files regenerated - rtp_media_server ... [skip ci]

2019-02-22 Thread Julien Chavanton
Hi, I wonder why the HTML documentation is not updated yet ?

when I test with the following command I do not detect any problem, is
there any pipeline failing ?

make modules-doc doc_format=html modules=modules/rtp_media_server

Thanks !


On Fri, Feb 22, 2019 at 9:32 AM Kamailio Dev 
wrote:

> Module: kamailio
> Branch: master
> Commit: 4b7e6089e32ed71897396b95fed60b2461f14434
> URL:
> https://github.com/kamailio/kamailio/commit/4b7e6089e32ed71897396b95fed60b2461f14434
>
> Author: Kamailio Dev 
> Committer: Kamailio Dev 
> Date: 2019-02-22T18:31:45+01:00
>
> modules: readme files regenerated - rtp_media_server ... [skip ci]
>
> ---
>
> Modified: src/modules/rtp_media_server/README
>
> ---
>
> Diff:
> https://github.com/kamailio/kamailio/commit/4b7e6089e32ed71897396b95fed60b2461f14434.diff
> Patch:
> https://github.com/kamailio/kamailio/commit/4b7e6089e32ed71897396b95fed60b2461f14434.patch
>
> ---
>
> diff --git a/src/modules/rtp_media_server/README
> b/src/modules/rtp_media_server/README
> index bc47d7311e..742264f366 100644
> --- a/src/modules/rtp_media_server/README
> +++ b/src/modules/rtp_media_server/README
> @@ -1,4 +1,3 @@
> -
>  rtp_media_server Module
>
>  Julien Chavanton
> @@ -38,8 +37,9 @@ Julien Chavanton
>
>4.1. rms_answer ()
>4.2. rms_hangup ()
> -  4.3. rms_media_stop ()
> -  4.4. rms_play ()
> +  4.3. rms_session_check ()
> +  4.4. rms_sip_request ()
> +  4.5. rms_play ()
>
> List of Examples
>
> @@ -48,6 +48,7 @@ Julien Chavanton
> 1.3. usage example
> 1.4. usage example
> 1.5. usage example
> +   1.6. usage example
>
>  Chapter 1. Admin Guide
>
> @@ -67,8 +68,9 @@ Chapter 1. Admin Guide
>
>  4.1. rms_answer ()
>  4.2. rms_hangup ()
> -4.3. rms_media_stop ()
> -4.4. rms_play ()
> +4.3. rms_session_check ()
> +4.4. rms_sip_request ()
> +4.5. rms_play ()
>
>  1. Overview
>
> @@ -111,6 +113,10 @@ Chapter 1. Admin Guide
>   * mediastreamer2 git clone git://git.linphone.org/mediastreamer2.git
> Mediastreamer2 is a powerful and lightweight streaming engine
> specialized for voice/video telephony applications.
> + * bcunit git clone
> +   https://github.com/BelledonneCommunications/bcunit.git
> +   fork of the defunct project CUnit, with several fixes and patches
> +   applied. CUnit is a Unit testing framework for C.
>
>  3. Parameters
>
> @@ -132,8 +138,9 @@ modparam("rtp_media_server", "log_file_name",
> "/var/log/rms/rms_ortp.log")
>
> 4.1. rms_answer ()
> 4.2. rms_hangup ()
> -   4.3. rms_media_stop ()
> -   4.4. rms_play ()
> +   4.3. rms_session_check ()
> +   4.4. rms_sip_request ()
> +   4.5. rms_play ()
>
>  4.1. rms_answer ()
>
> @@ -166,11 +173,7 @@ route {
>  t_reply("503", "server error");
>  }
>  }
> -
> -if (is_method("BYE")){
> -xnotice("BYE RECEIVED [$ci]\n");
> -rms_media_stop();
> -}
> +rms_sip_request();
>  ...
>
>  4.2. rms_hangup ()
> @@ -184,10 +187,27 @@ route {
>  rms_hangup();
>  ...
>
> -4.3. rms_media_stop ()
> +4.3. rms_session_check ()
> +
> +   Returns true if the current SIP message it handled/known by the RMS
> +   module, else it may be handle in any other way by Kamailio.
> +
> +   This function can be used from REQUEST_ROUTE, REPLY_ROUTE and
> +   FAILURE_ROUTE.
> +
> +   Example 1.4. usage example
> +...
> +if (rms_session_check()) {
> +xnotice("This session is handled by the RMS module\n");
> +rms_sip_request();
> +}
> +...
> +
> +4.4. rms_sip_request ()
>
> -   This should be called on reception of a BYE, this will delete the RTP
> -   session and the media ressources. and reply "200 OK".
> +   This should be called for every in-dialog SIP request, it will be
> +   forwarded behaving as a B2BUA, the transaction will be suspended until
> +   the second leg replies.
>
> If the SIP session is not found "481 Call/Transaction Does Not Exist"
> is returned.
> @@ -195,14 +215,14 @@ route {
> This function can be used from REQUEST_ROUTE, REPLY_ROUTE and
> FAILURE_ROUTE.
>
> -   Example 1.4. usage example
> +   Example 1.5. usage example
>  ...
> -if (is_method("BYE")){
> -rms_media_stop();
> +if (rms_session_check()) {
> +rms_sip_request();
>  }
>  ...
>
> -4.4. rms_play ()
> +4.5. rms_play ()
>
> Play a wav file, a resampler is automaticaly configured to resample and
> convert stereo to mono if needed.
> @@ -212,7 +232,7 @@ route {
>
> This function can be used from EVENT_ROUTE.
>
> -   Example 1.5. usage example
> +   Example 1.6. usage example
>  ...
>  rms_play("file.wav", "event_route_name");
>  ...
>
>
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Re: [sr-dev] [kamailio/kamailio] Cannot use db_flatstore with siptrace module (#1850)

2019-02-22 Thread Chris Ayoup
I tried it out and it works, thanks!

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Re: [sr-dev] [kamailio/kamailio] Kamailio 5.2.0 - Segmentation fault in libcrypto.so.1.1 (#1860)

2019-02-22 Thread Henning Westerholt
About the question "Are there any significant drawbacks setting 'children' to 
1? Thus far everything appears to be working as expected."
You will not get a good performance this way, as Kamailio can't process any new 
messages if there is e.g. a database operation running in the one child. You 
will also not use your processors effectively. It  is not recommended to run a 
regular production environment with children=1.

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[sr-dev] git:master:4b7e6089: modules: readme files regenerated - rtp_media_server ... [skip ci]

2019-02-22 Thread Kamailio Dev
Module: kamailio
Branch: master
Commit: 4b7e6089e32ed71897396b95fed60b2461f14434
URL: 
https://github.com/kamailio/kamailio/commit/4b7e6089e32ed71897396b95fed60b2461f14434

Author: Kamailio Dev 
Committer: Kamailio Dev 
Date: 2019-02-22T18:31:45+01:00

modules: readme files regenerated - rtp_media_server ... [skip ci]

---

Modified: src/modules/rtp_media_server/README

---

Diff:  
https://github.com/kamailio/kamailio/commit/4b7e6089e32ed71897396b95fed60b2461f14434.diff
Patch: 
https://github.com/kamailio/kamailio/commit/4b7e6089e32ed71897396b95fed60b2461f14434.patch

---

diff --git a/src/modules/rtp_media_server/README 
b/src/modules/rtp_media_server/README
index bc47d7311e..742264f366 100644
--- a/src/modules/rtp_media_server/README
+++ b/src/modules/rtp_media_server/README
@@ -1,4 +1,3 @@
-
 rtp_media_server Module
 
 Julien Chavanton
@@ -38,8 +37,9 @@ Julien Chavanton
 
   4.1. rms_answer ()
   4.2. rms_hangup ()
-  4.3. rms_media_stop ()
-  4.4. rms_play ()
+  4.3. rms_session_check ()
+  4.4. rms_sip_request ()
+  4.5. rms_play ()
 
List of Examples
 
@@ -48,6 +48,7 @@ Julien Chavanton
1.3. usage example
1.4. usage example
1.5. usage example
+   1.6. usage example
 
 Chapter 1. Admin Guide
 
@@ -67,8 +68,9 @@ Chapter 1. Admin Guide
 
 4.1. rms_answer ()
 4.2. rms_hangup ()
-4.3. rms_media_stop ()
-4.4. rms_play ()
+4.3. rms_session_check ()
+4.4. rms_sip_request ()
+4.5. rms_play ()
 
 1. Overview
 
@@ -111,6 +113,10 @@ Chapter 1. Admin Guide
  * mediastreamer2 git clone git://git.linphone.org/mediastreamer2.git
Mediastreamer2 is a powerful and lightweight streaming engine
specialized for voice/video telephony applications.
+ * bcunit git clone
+   https://github.com/BelledonneCommunications/bcunit.git
+   fork of the defunct project CUnit, with several fixes and patches
+   applied. CUnit is a Unit testing framework for C.
 
 3. Parameters
 
@@ -132,8 +138,9 @@ modparam("rtp_media_server", "log_file_name", 
"/var/log/rms/rms_ortp.log")
 
4.1. rms_answer ()
4.2. rms_hangup ()
-   4.3. rms_media_stop ()
-   4.4. rms_play ()
+   4.3. rms_session_check ()
+   4.4. rms_sip_request ()
+   4.5. rms_play ()
 
 4.1. rms_answer ()
 
@@ -166,11 +173,7 @@ route {
 t_reply("503", "server error");
 }
 }
-
-if (is_method("BYE")){
-xnotice("BYE RECEIVED [$ci]\n");
-rms_media_stop();
-}
+rms_sip_request();
 ...
 
 4.2. rms_hangup ()
@@ -184,10 +187,27 @@ route {
 rms_hangup();
 ...
 
-4.3. rms_media_stop ()
+4.3. rms_session_check ()
+
+   Returns true if the current SIP message it handled/known by the RMS
+   module, else it may be handle in any other way by Kamailio.
+
+   This function can be used from REQUEST_ROUTE, REPLY_ROUTE and
+   FAILURE_ROUTE.
+
+   Example 1.4. usage example
+...
+if (rms_session_check()) {
+xnotice("This session is handled by the RMS module\n");
+rms_sip_request();
+}
+...
+
+4.4. rms_sip_request ()
 
-   This should be called on reception of a BYE, this will delete the RTP
-   session and the media ressources. and reply "200 OK".
+   This should be called for every in-dialog SIP request, it will be
+   forwarded behaving as a B2BUA, the transaction will be suspended until
+   the second leg replies.
 
If the SIP session is not found "481 Call/Transaction Does Not Exist"
is returned.
@@ -195,14 +215,14 @@ route {
This function can be used from REQUEST_ROUTE, REPLY_ROUTE and
FAILURE_ROUTE.
 
-   Example 1.4. usage example
+   Example 1.5. usage example
 ...
-if (is_method("BYE")){
-rms_media_stop();
+if (rms_session_check()) {
+rms_sip_request();
 }
 ...
 
-4.4. rms_play ()
+4.5. rms_play ()
 
Play a wav file, a resampler is automaticaly configured to resample and
convert stereo to mono if needed.
@@ -212,7 +232,7 @@ route {
 
This function can be used from EVENT_ROUTE.
 
-   Example 1.5. usage example
+   Example 1.6. usage example
 ...
 rms_play("file.wav", "event_route_name");
 ...


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[sr-dev] git:master:108f2cd8: rtp_media_server: adding call bridging

2019-02-22 Thread Julien Chavanton
Module: kamailio
Branch: master
Commit: 108f2cd8272dcf10e912c6f81bc81548b4603484
URL: 
https://github.com/kamailio/kamailio/commit/108f2cd8272dcf10e912c6f81bc81548b4603484

Author: Julien Chavanton 
Committer: Julien Chavanton 
Date: 2019-02-08T23:24:41Z

rtp_media_server: adding call bridging

- forward in dialog messages using RMS bridge in B2BUA manner
- move all code related to rms_session_info to dedicated source file
- session_list refactoring to always include all dialog information
  needed
- suspend transaction to have non blocking bridging
- random rtp port
- stop using shared memory for MS2 and oRTP, manage all the allocation
  from one dedicated process

---

Added: src/modules/rtp_media_server/doc/design_notes.txt
Added: src/modules/rtp_media_server/rms_session_info.c
Added: src/modules/rtp_media_server/rms_session_info.h
Modified: src/modules/rtp_media_server/Makefile
Modified: src/modules/rtp_media_server/config_example/kamailio.cfg
Modified: src/modules/rtp_media_server/doc/rtp_media_server_admin.xml
Modified: src/modules/rtp_media_server/rms_media.c
Modified: src/modules/rtp_media_server/rms_media.h
Modified: src/modules/rtp_media_server/rms_sdp.c
Modified: src/modules/rtp_media_server/rms_sdp.h
Modified: src/modules/rtp_media_server/rms_util.h
Modified: src/modules/rtp_media_server/rtp_media_server.c
Modified: src/modules/rtp_media_server/rtp_media_server.h

---

Diff:  
https://github.com/kamailio/kamailio/commit/108f2cd8272dcf10e912c6f81bc81548b4603484.diff
Patch: 
https://github.com/kamailio/kamailio/commit/108f2cd8272dcf10e912c6f81bc81548b4603484.patch


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Re: [sr-dev] [kamailio/kamailio] rtp_media_server: adding call bridging (#1823)

2019-02-22 Thread Julien Chavanton
Merged #1823 into master.

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[sr-dev] git:master:f6493ba1: Merge pull request #1823 from kamailio/rtp_media_server

2019-02-22 Thread GitHub
Module: kamailio
Branch: master
Commit: f6493ba109565dcb14cc72a9697c1d7c3014c246
URL: 
https://github.com/kamailio/kamailio/commit/f6493ba109565dcb14cc72a9697c1d7c3014c246

Author: Julien Chavanton 
Committer: GitHub 
Date: 2019-02-22T09:27:16-08:00

Merge pull request #1823 from kamailio/rtp_media_server

rtp_media_server: adding call bridging

---

Added: src/modules/rtp_media_server/doc/design_notes.txt
Added: src/modules/rtp_media_server/rms_session_info.c
Added: src/modules/rtp_media_server/rms_session_info.h
Modified: src/modules/rtp_media_server/Makefile
Modified: src/modules/rtp_media_server/config_example/kamailio.cfg
Modified: src/modules/rtp_media_server/doc/rtp_media_server_admin.xml
Modified: src/modules/rtp_media_server/rms_media.c
Modified: src/modules/rtp_media_server/rms_media.h
Modified: src/modules/rtp_media_server/rms_sdp.c
Modified: src/modules/rtp_media_server/rms_sdp.h
Modified: src/modules/rtp_media_server/rms_util.h
Modified: src/modules/rtp_media_server/rtp_media_server.c
Modified: src/modules/rtp_media_server/rtp_media_server.h

---

Diff:  
https://github.com/kamailio/kamailio/commit/f6493ba109565dcb14cc72a9697c1d7c3014c246.diff
Patch: 
https://github.com/kamailio/kamailio/commit/f6493ba109565dcb14cc72a9697c1d7c3014c246.patch


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Re: [sr-dev] [kamailio/kamailio] Infinite loop inside htable module during dmq synch (#1863)

2019-02-22 Thread Charles Chance
Thanks for testing so extensively!

My (limited) understanding is that the call to srjson_DeleteItemFromObject() - 
by virtue of its internal call to srjson_Delete() - takes care of freeing the 
entire "cells" structure.

I will do some testing here to be sure there is no leak but in the meantime, 
can you confirm whether you see the memory increase linearly in line with the 
number of times you launch a new Kamailio instance? For example, if you 
increase from 100 to 200 times, does the memory usage then settle at 2.7%?

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Re: [sr-dev] [kamailio/kamailio] Kamailio 5.2.0 - Segmentation fault in libcrypto.so.1.1 (#1860)

2019-02-22 Thread shaunjstokes
We tried changing children to 8 and the problem still occurs, 1 is the only 
safe value thus far.

Not sure exactly on the relationship between Kamailio and Libcrypto, does 
Kamailio need to interact with Libcrypto in a specific order? For example, if 
there are multiple messages sent to Kamailio both of which require some form of 
interaction with Libcrypto but one thread takes slightly than another, does it 
matter the order in which Kamailio interacts with Libcrypto?

Will look at installing debug symbols for libssl/libcrypto.

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[sr-dev] git:master:1959a8f2: modules: readme files regenerated - evrexec ... [skip ci]

2019-02-22 Thread Kamailio Dev
Module: kamailio
Branch: master
Commit: 1959a8f23ff0fd78f1a34045addfc5a096db6556
URL: 
https://github.com/kamailio/kamailio/commit/1959a8f23ff0fd78f1a34045addfc5a096db6556

Author: Kamailio Dev 
Committer: Kamailio Dev 
Date: 2019-02-22T12:32:12+01:00

modules: readme files regenerated - evrexec ... [skip ci]

---

Modified: src/modules/evrexec/README

---

Diff:  
https://github.com/kamailio/kamailio/commit/1959a8f23ff0fd78f1a34045addfc5a096db6556.diff
Patch: 
https://github.com/kamailio/kamailio/commit/1959a8f23ff0fd78f1a34045addfc5a096db6556.patch

---

diff --git a/src/modules/evrexec/README b/src/modules/evrexec/README
index 02ae3c86b8..f7363c4c43 100644
--- a/src/modules/evrexec/README
+++ b/src/modules/evrexec/README
@@ -28,6 +28,10 @@ Daniel-Constantin Mierla
 
   3.1. exec (str)
 
+4. RPC Commands
+
+  4.1. evrexec.run
+
List of Examples
 
1.1. Set exec parameter
@@ -46,6 +50,10 @@ Chapter 1. Admin Guide
 
 3.1. exec (str)
 
+   4. RPC Commands
+
+4.1. evrexec.run
+
 1. Overview
 
The module executes event route blocks or KEMI functions on dedicated
@@ -108,3 +116,30 @@ event_route[evrexec:timer] {
   }
 }
 ...
+
+4. RPC Commands
+
+   4.1. evrexec.run
+
+4.1. evrexec.run
+
+   Run an event_route block or a KEMI function upon an RPC command.
+
+   Name: evroute.run
+
+   Parameters:
+ * evname
+   - the name of the event route block or the KEMI function.
+ * evdata
+   - (optional) - arbitrary data passed as a string, which is made
+   available inside event route block as $evr(data).
+
+   RPC Command Format:
+...
+event_route[evrexec:test] {
+   xlog("rpc command data: $evr(data)\n");
+}
+...
+kamctl rpc evroute.run evroute:test
+kamctl rpc evroute.run evroute:test mydata
+...


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[sr-dev] git:master:4eb1c1b3: evrexec: documentation for rpc command evrexec.run

2019-02-22 Thread Daniel-Constantin Mierla
Module: kamailio
Branch: master
Commit: 4eb1c1b33edaf889cd10d62e7c5f21c9a260fd0d
URL: 
https://github.com/kamailio/kamailio/commit/4eb1c1b33edaf889cd10d62e7c5f21c9a260fd0d

Author: Daniel-Constantin Mierla 
Committer: Daniel-Constantin Mierla 
Date: 2019-02-22T12:16:40+01:00

evrexec: documentation for rpc command evrexec.run

---

Modified: src/modules/evrexec/doc/evrexec_admin.xml

---

Diff:  
https://github.com/kamailio/kamailio/commit/4eb1c1b33edaf889cd10d62e7c5f21c9a260fd0d.diff
Patch: 
https://github.com/kamailio/kamailio/commit/4eb1c1b33edaf889cd10d62e7c5f21c9a260fd0d.patch

---

diff --git a/src/modules/evrexec/doc/evrexec_admin.xml 
b/src/modules/evrexec/doc/evrexec_admin.xml
index 3d39cf6ab4..bcd6af5f49 100644
--- a/src/modules/evrexec/doc/evrexec_admin.xml
+++ b/src/modules/evrexec/doc/evrexec_admin.xml
@@ -123,5 +123,43 @@ event_route[evrexec:timer] {


 
+   
+   RPC Commands
+   
+   evrexec.run
+   
+   Run an event_route block or a KEMI function upon an RPC command.
+   
+   
+   Name: evroute.run
+   
+   Parameters:
+   
+   
+   evname - the name of the event route block 
or the KEMI
+   function.
+   
+   
+   evdata - (optional) - arbitrary data 
passed as a string,
+   which is made available inside event route block as 
$evr(data).
+   
+   
+   
+
+   
+   RPC Command Format:
+   
+   
+...
+event_route[evrexec:test] {
+   xlog("rpc command data: $evr(data)\n");
+}
+...
+kamctl rpc evroute.run evroute:test
+kamctl rpc evroute.run evroute:test mydata
+...
+   
+   
+   
 
 


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[sr-dev] git:master:04ae96ab: evrexec: added rpc command evrexec.run to run an event_route

2019-02-22 Thread Daniel-Constantin Mierla
Module: kamailio
Branch: master
Commit: 04ae96ab5fa3e219ab9d326fec7055ef3b7a212f
URL: 
https://github.com/kamailio/kamailio/commit/04ae96ab5fa3e219ab9d326fec7055ef3b7a212f

Author: Daniel-Constantin Mierla 
Committer: Daniel-Constantin Mierla 
Date: 2019-02-22T11:54:13+01:00

evrexec: added rpc command evrexec.run to run an event_route

- kamctl rpc evrexec.run evname [evdata]
- first parameter is the event route name (or KEMI function)
- second parameter (optional, string) is an arbitrary data that becomes
available inside the event route via $evr(data)

---

Modified: src/modules/evrexec/evrexec_mod.c

---

Diff:  
https://github.com/kamailio/kamailio/commit/04ae96ab5fa3e219ab9d326fec7055ef3b7a212f.diff
Patch: 
https://github.com/kamailio/kamailio/commit/04ae96ab5fa3e219ab9d326fec7055ef3b7a212f.patch


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[sr-dev] Planning next IRC devel meeting

2019-02-22 Thread Daniel-Constantin Mierla
Hello,

wondering if we we should do a new IRC devel meeting in the near future
to sync on development plans.

One of the decisions we should take is whether we should target to
release the new major version (5.3) before the summer holidays or we
leave it for the autumn.

If many wants to do it, then a first proposal for a date would be: March
07, 2019, at 15:00 UTC (16:00 Berlin time).

As usual, I created a wiki page to track the availability and the topics
that should be approched:

  * https://www.kamailio.org/wiki/devel/irc-meetings/2019a

Feel free to add yourself there, propose topics, etc...

Cheers,
Daniel

-- 
Daniel-Constantin Mierla -- www.asipto.com
www.twitter.com/miconda -- www.linkedin.com/in/miconda
Kamailio World Conference - May 6-8, 2019 -- www.kamailioworld.com
Kamailio Advanced Training - Mar 4-6, 2019 in Berlin; Mar 25-27, 2019, in 
Washington, DC, USA -- www.asipto.com


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Re: [sr-dev] [kamailio/kamailio] Infinite loop inside htable module during dmq synch (#1863)

2019-02-22 Thread Enrico Bandiera
Thank you @charlesrchance , I did some tests with this setup:
kamailio.cfg (meaningful lines):
```
fork=yes
children=1
tcp_connection_lifetime=3605
pv_buffer_size=8192

# - dmq params -
modparam("dmq", "server_address", DMQ_SERVER_ADDRESS)
modparam("dmq", "notification_address", DMQ_NOTIFICATION_ADDRESS)
modparam("dmq", "multi_notify", 1)
modparam("dmq", "num_workers", 1)
modparam("dmq", "ping_interval", 15)
modparam("dmq", "worker_usleep", 1000)

# - htable params -
modparam("htable", "enable_dmq", 1)
modparam("htable", "dmq_init_sync", 1)
modparam("htable", "htable", "ht=>size=16;dmqreplicate=1;autoexpire=10800;")
   # Keep track of concurrent channels for accounts. Should be same as 
dialog
modparam("htable", "htable", "ht1=>size=16;dmqreplicate=1;autoexpire=10800;")   
# Keep track of concurrent channels for accounts. Should be same as 
dialog
modparam("htable", "htable", "ht2=>size=16;dmqreplicate=1;autoexpire=10800;")   
# Keep track of concurrent channels for accounts. Should be same as 
dialog
modparam("htable", "htable", "ht3=>size=16;dmqreplicate=1;autoexpire=10800;")   
# Keep track of concurrent channels for accounts. Should be same as 
dialog

#!define ONEK "really 1 k chars, believe me :)"

event_route[htable:mod-init] {
  $var(name) = POD_NAME + "\n";
  xlog("L_ALERT", "$var(name)");
  if(POD_NAME == "kama-0") {
$var(count) = 0;
while($var(count) < 99) {
  $sht(ht=>$var(count)) = ONEK;
  $sht(ht1=>$var(count)) = ONEK;
  $sht(ht2=>$var(count)) = ONEK;
  $sht(ht3=>$var(count)) = ONEK;
  $var(count) = $var(count)+1;
}
  }
}

request_route {
  if ($rm == "KDMQ"){
dmq_handle_message();
  }
  exit;
}
```

Started kama-0 which has now 4 htables of ~99K size each
Started 10 kubernetes pods and launched kamailio 100 times with a timeout of 3 
seconds on each pod
So we have roughly 1000 kamailios trying to get these htables from kama-0
I didn't see any dangerous CPU spike and the loop doesn't happen anymore.

There's something I'm worried of though: the memory of the DMQ worker (measured 
from top), which usually stays around 0.1% is now stable at 1.4% and it's not 
going down again

I fear there's a memory leak somewhere but I'm not sure where, I had some 
doubts while debugging the loop issue about how the json_t structures are freed 
but it could be caused by me not knowing well the code; can you give us any 
hint to help you understand this issue better?

Thanks


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Re: [sr-dev] [kamailio/kamailio] Can't install kamailio-sipcapture-daemon-config package due to missing dependency (#1861)

2019-02-22 Thread sergey-safarov
Is sipcapture config started at server boot?

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