[asterisk-users] http.conf - haproxy

2016-08-22 Thread Marek Červenka 

hello,

is it possible move asterisk http server behind haproxy (haproxy as SSL 
endpoint, asterisk http only)


any examples?

my current http.conf

[general]
enabled=yes
bindaddr=0.0.0.0
tlsenable=yes
tlsbindaddr=0.0.0.0:8089
tlscertfile=/etc/pki/tls/certs/some.crt
tlsprivatekey=/etc/pki/tls/private/some.key
tlscipher=ALL

tnx

marek


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Re: [asterisk-users] max concurrent calls with bundled pjproject

2016-08-18 Thread Marek Červenka 

you can patch it in

[cervenka@matrix asterisk-13.9.1]$ ll third-party/pjproject/
total 24
-rwxrwxr-x. 1 cervenka cervenka  877 May 13 19:41 apply_patches
-rw-rw-r--. 1 cervenka cervenka 1794 May 13 19:41 configure.m4
-rw-rw-r--. 1 cervenka cervenka 5352 May 13 19:41 Makefile
-rw-rw-r--. 1 cervenka cervenka  428 May 13 19:41 Makefile.rules
drwxrwxr-x. 2 cervenka cervenka 4096 May 13 19:41 patches


Dne 18.8.2016 v 11:33 ian gilmour napsal(a):


Hi,

PJSIP in the past had limitations on the max concurrent calls, etc. 
There were ways to overcome them by changing the source code. (e.g. 
http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/2013-February/015721.html)


Do any similar tweaks need to be done to the bundled pjproject to 
handle high volumes of concurrent calls with Asterisk?


What (if any) are the current default asterisk 13 + pjproject audio + 
video concurrent call limits if using the bundled pjproject + asterisk 
patches as is?


Thanks in advance.

Regards,

IanG





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[asterisk-users] CDR replacement with CEL

2016-07-14 Thread Marek Červenka 

hi,

i'm trying replace CDR with CEL

reasons:

- minimize Stasis listeners (CDR)

- CEL, CDR produces "similar" data

- own logic of CDR meaning like "calldate,src,dst,direction,.." dst is 
always first connected point in PBX - real user or IVR/queue etc., 
numbers are only attributes of object "user"



do you have any tips/logic/comments for this goal?


my custom cdr table

CREATE TABLE `cdr` (
  `id` bigint(20) NOT NULL AUTO_INCREMENT,
  `user_id` int(11) NOT NULL COMMENT 'user id',
  `tenant_id` int(11) NOT NULL COMMENT 'tenant id',
  `calldate` datetime NOT NULL DEFAULT '-00-00 00:00:00',
  `clid` varchar(80) NOT NULL DEFAULT '',
  `src` varchar(80) NOT NULL DEFAULT '',
  `dst` varchar(80) NOT NULL DEFAULT '',
  `duration` int(11) NOT NULL DEFAULT '0',
  `billsec` int(11) NOT NULL DEFAULT '0',
  `disposition` varchar(45) NOT NULL DEFAULT '' COMMENT 'asterisk hangup cause',
  `way` enum('loc','in','out') NOT NULL DEFAULT 'loc' COMMENT 'call direction 
(loc - local, in - incoming, out - outgoing)',
  `trunk` varchar(80) NOT NULL COMMENT 'used SIP trunk',
  `hangupcause` varchar(10) NOT NULL COMMENT 'hangup cause',
  `hangupside` varchar(10) NOT NULL COMMENT 'hangup on which side',
  `uniqueid` varchar(64) NOT NULL DEFAULT '',
  `linkedid` varchar(64) NOT NULL,
  `data` json NOT NULL COMMENT 'metadata',
  `stamp` timestamp NOT NULL DEFAULT CURRENT_TIMESTAMP ON UPDATE 
CURRENT_TIMESTAMP COMMENT 'creation date',
  PRIMARY KEY (`id`),
  KEY `dst` (`dst`),
  KEY `uniqueid` (`uniqueid`),
) ENGINE=InnoDB DEFAULT CHARSET=utf8;


CEL pairing for simple call scenario

user_id = own variable (from cel userfield or CELGenUserEvent app)
tenant_id = own variable (from cel accountcode or CELGenUserEvent app)
calldate = eventtime
src = cid_num
dst = exten
duration =  eventtime(event HANGUP) - eventtime(eventtype BRIDGE_ENTER)  (no 
eventtype PICKUP,FORWARD,*TRANSFER)(or howto identify event RINGING?)
billsec =  eventtime(event HANGUP) - eventtime(eventtype ANSWER)  (no eventtype 
PICKUP,FORWARD,*TRANSFER)
way = own variable (CELGenUserEvent app)
disposition = extra: 
{"hangupcause":16,"hangupsource":"SIP/siptrunk-0a80","dialstatus":"ANSWER"}
trunk = own variable (CELGenUserEvent app)
hangupcause = extra: 
{"hangupcause":16,"hangupsource":"SIP/siptrunk-0a80","dialstatus":"ANSWER"}
hangupside = ???

 



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Re: [asterisk-users] log debug (SOLVED)

2016-06-24 Thread Marek Červenka 


some very bad guy (maybe me) set global debug to 3 and forgot

hint

check

asterisk*CLI> core show settings

PBX Core settings
-
  Version: 13.9.1
  Build Options:   LOADABLE_MODULES, OPTIONAL_API
  Maximum calls:   Not set
  Maximum open file handles:   Not set
  Root console verbosity:  0
  Current console verbosity:   0
  Debug level: 0
  Maximum load average:0.00
  Minimum free memory: 0 MB
  Startup time:19:03:23


question
is performance worse if debug level is higher than 1 but no logger.conf 
options logs debug data?



Dne 24.6.2016 v 11:16 Marek Červenka napsal(a):

hi,

i want debug only app_queue  (asterisk 13.9)

i have this configuration

[general]
[logfiles]
console => notice,warning,error
messages => notice,warning,error
;full => notice,warning,error,debug,verbose
debug => debug

syslog.local1 => warning,error

but after

asterisk*CLI> core set debug 1 app_queue
Core debug was 0 and has been set to 1 for 'app_queue'.

i see a ton of

[Jun 24 11:11:45] DEBUG[26980][C-251e] audiohook.c: Write factory 
0xc86229c was pretty quick last time, waiting for them.
[Jun 24 11:11:45] DEBUG[26434][C-2516] audiohook.c: Read factory 
0x91fc247c was pretty quick last time, waiting for them.
[Jun 24 11:11:45] DEBUG[26980][C-251e] audiohook.c: Write factory 
0xc86229c was pretty quick last time, waiting for them.
[Jun 24 11:11:45] DEBUG[26980][C-251e] audiohook.c: Write factory 
0xc86229c was pretty quick last time, waiting for them.


is it possible to see only debug for app_queue?





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[asterisk-users] log debug

2016-06-24 Thread Marek Červenka 

hi,

i want debug only app_queue  (asterisk 13.9)

i have this configuration

[general]
[logfiles]
console => notice,warning,error
messages => notice,warning,error
;full => notice,warning,error,debug,verbose
debug => debug

syslog.local1 => warning,error

but after

asterisk*CLI> core set debug 1 app_queue
Core debug was 0 and has been set to 1 for 'app_queue'.

i see a ton of

[Jun 24 11:11:45] DEBUG[26980][C-251e] audiohook.c: Write factory 
0xc86229c was pretty quick last time, waiting for them.
[Jun 24 11:11:45] DEBUG[26434][C-2516] audiohook.c: Read factory 
0x91fc247c was pretty quick last time, waiting for them.
[Jun 24 11:11:45] DEBUG[26980][C-251e] audiohook.c: Write factory 
0xc86229c was pretty quick last time, waiting for them.
[Jun 24 11:11:45] DEBUG[26980][C-251e] audiohook.c: Write factory 
0xc86229c was pretty quick last time, waiting for them.


is it possible to see only debug for app_queue?


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[asterisk-users] queue_log - odbc vs AMI

2016-06-16 Thread Marek Červenka 

hi,

can someone confirm that queue_log data are the same if are received via 
AMI as if they are saved via ODBC


thanks


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[asterisk-users] pbx testsuite

2016-06-15 Thread Marek Červenka 

hi,

we have in house developed pbx testsuite based on

 * node.js
 * selenium
 * protractor
 * gulp
 * pjsip - pjsua python
 * docker

 there are helpers for testing

 * sip
 * web
 * api

you can create end-to-end scenarios like
- create 2 users via web
- call from first user to second
- check CDR result via API

but
we have some problems in "burning" tests with frozen jasmine reporter, 
with account management in pjsua python, ...


my questions are:
is there some similiar testsuite based on node.js technology? (i know 
about asterisk-testsuite and xivo-testsuite)

is there interest in publishing our testsuite on github?

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Re: [asterisk-users] Recent UnixODBC Issues

2016-06-07 Thread Marek Červenka 

Dne 6.6.2016 v 17:42 Joshua Colp napsal(a):

Happy Monday all,

Since I sent my previous email a lot has been learnt about our 
UnixODBC problem and a path has emerged ensuring both better 
performance while
making sure people are not required to upgrade their UnixODBC unless 
they want to.


So what's this mean?

As of Asterisk 13.10.0 our own connection pool will be used in the 
res_odbc module. Under testing this has shown to be more performant 
than the UnixODBC implementation and also does not suffer the slowdown 
present in UnixODBC 2.3.3 and above. As well since our own connection 
pool has a fixed size we can restore behavior to that of previous 
versions so those using UnixODBC 2.3.1 will not experience the crashes 
that have been seen.


This is done by having the default be 1 connection, thus disabling 
connection pooling. To turn connection pooling on you will need to 
ensure you are using UnixODBC 2.3.2 or above with latest database 
connectors and configure a maximum limit on concurrent connections in 
res_odbc.conf. To facilitate figuring out the right limit for your 
environment I've made the current count and limit available using the 
"odbc show" CLI command.


This change is up for review[1] and any feedback would be welcome, 
both from code review itself and testing.


Cheers,

[1] https://gerrit.asterisk.org/#/c/2943/



whats the recommended settings in odbcinst.ini
Pooling = no
threading = 0

?

do you think it is possible backport this to 13.9 or are there some 
problematic dependencies?



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Re: [asterisk-users] asterisk odbc segfaults (SOLVED)

2016-05-30 Thread Marek Červenka 
/cel_custom.so
b3b7e000-b3b7f000 rw-p 2000 08:01 31147 
/usr/lib/asterisk/modules/cel_custom.so
b3b7f000-b3b82000 r-xp  08:01 31186 
/usr/lib/asterisk/modules/format_pcm.so
b3b82000-b3b83000 rw-p 2000 08:01 31186 
/usr/lib/asterisk/modules/format_pcm.so
b3b83000-b3b84000 r-xp  08:01 31163 
/usr/lib/asterisk/modules/codec_a_mu.so
b3b84000-b3b85000 rw-p 1000 08:01 31163 
/usr/lib/asterisk/modules/codec_a_mu.so
b3b85000-b3b86000 r-xp  08:01 31074 
/usr/lib/asterisk/modules/app_echo.so
b3b86000-b3b87000 rw-p  08:01 31074 
/usr/lib/asterisk/modules/app_echo.so
b3b87000-b3b89000 r-xp  08:01 31299 
/usr/lib/asterisk/modules/res_manager_presencestate.so
b3b89000-b3b8a000 rw-p 1000 08:01 31299 
/usr/lib/asterisk/modules/res_manager_presencestate.so
b3b8a000-b3b8c000 r-xp  08:01 31274 
/usr/lib/asterisk/modules/res_clialiases.so
b3b8c000-b3b8d000 rw-p 2000 08:01 31274 
/usr/lib/asterisk/modules/res_clialiases.so
b3b8d000-b3b95000 r-xp  08:01 31149 
/usr/lib/asterisk/modules/cel_odbc.so
b3b95000-b3b96000 rw-p 7000 08:01 31149 
/usr/lib/asterisk/modules/cel_odbc.so
b3b96000-b3b98000 r-xp  08:01 31132 
/usr/lib/asterisk/modules/app_zapateller.so
b3b98000-b3b99000 rw-p 1000 08:01 31132 
/usr/lib/asterisk/modules/app_zapateller.so
b3b99000-b3b9b000 r-xp  08:01 31346 
/usr/lib/asterisk/modules/res_pjsip_send_to_voicemail.so/usr/sbin/safe_asterisk: 
line 164:  3901 Aborted (core dumped) nice -n $PRIORITY 
"${ASTSBINDIR}/asterisk" -f ${CLIARGS} ${ASTARGS} > /dev/${TTY} 2>&1 < 
/dev/${TTY}





Dne 29.5.2016 v 22:04 Marek Červenka napsal(a):

Dne 29.5.2016 v 21:31 Marek Červenka napsal(a):

doesnt work for me


strange. combination of your tip with

Option  = 3

in odbc.ini solved my problem

i'm trying find what option=3 means



Dne 29.5.2016 v 17:48 Niklas Larsson napsal(a):

Hi,


On 2016-05-27 18:28, Marek Červenka wrote:

after downgrade to 13.8.2
May 27 18:21:06 ast kernel: asterisk[16286]: segfault at 1010024 ip 
b49162cd sp bfac0940 error 4 in 
libmysqlclient.so.16.0.0[b48f1000+12e000]


after downgrade to 13.7.2
asterisk is ok


Could be https://issues.asterisk.org/jira/browse/ASTERISK-25957 - a 
solution is to change the order in modules.conf to:


preload => res_odbc.so
preload => res_config_odbc.so
preload => chan_local.so
preload => cdr_adaptive_odbc.so

/niklas




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Re: [asterisk-users] asterisk odbc segfaults (SOLVED)

2016-05-29 Thread Marek Červenka 

Dne 29.5.2016 v 21:31 Marek Červenka napsal(a):

doesnt work for me


strange. combination of your tip with

Option  = 3

in odbc.ini solved my problem

i'm trying find what option=3 means



Dne 29.5.2016 v 17:48 Niklas Larsson napsal(a):

Hi,


On 2016-05-27 18:28, Marek Červenka wrote:

after downgrade to 13.8.2
May 27 18:21:06 ast kernel: asterisk[16286]: segfault at 1010024 ip 
b49162cd sp bfac0940 error 4 in 
libmysqlclient.so.16.0.0[b48f1000+12e000]


after downgrade to 13.7.2
asterisk is ok


Could be https://issues.asterisk.org/jira/browse/ASTERISK-25957 - a 
solution is to change the order in modules.conf to:


preload => res_odbc.so
preload => res_config_odbc.so
preload => chan_local.so
preload => cdr_adaptive_odbc.so

/niklas




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Re: [asterisk-users] asterisk odbc segfaults

2016-05-29 Thread Marek Červenka 

doesnt work for me

Dne 29.5.2016 v 17:48 Niklas Larsson napsal(a):

Hi,


On 2016-05-27 18:28, Marek Červenka wrote:

after downgrade to 13.8.2
May 27 18:21:06 ast kernel: asterisk[16286]: segfault at 1010024 ip 
b49162cd sp bfac0940 error 4 in 
libmysqlclient.so.16.0.0[b48f1000+12e000]


after downgrade to 13.7.2
asterisk is ok


Could be https://issues.asterisk.org/jira/browse/ASTERISK-25957 - a 
solution is to change the order in modules.conf to:


preload => res_odbc.so
preload => res_config_odbc.so
preload => chan_local.so
preload => cdr_adaptive_odbc.so

/niklas




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Re: [asterisk-users] asterisk odbc segfaults

2016-05-27 Thread Marek Červenka 

after downgrade to 13.8.2
May 27 18:21:06 ast kernel: asterisk[16286]: segfault at 1010024 ip 
b49162cd sp bfac0940 error 4 in libmysqlclient.so.16.0.0[b48f1000+12e000]


after downgrade to 13.7.2
asterisk is ok

Dne 27.5.2016 v 18:09 Marek Červenka napsal(a):

btw info from my segfault

Core was generated by `/usr/sbin/asterisk -f -vvvg -c'.
Program terminated with signal 11, Segmentation fault.
#0  fix_result_types (stmt=0x9741730)
at /usr/src/debug/mysql-connector-odbc-5.3.6-src/driver/utility.c:139
139 irrec->row.field= field;


Dne 27.5.2016 v 17:58 Marek Červenka napsal(a):


hi,

i have the same problems as in 
https://issues.asterisk.org/jira/browse/ASTERISK-25833


my current combination is centos 6 32-bit, unixODBC 2.3.2 (recompiled 
from fedora20), mysql 5.1.73, mysql-connector-odbc 5.1.5, asterisk 13.9.1


i tried update to mysql-connector-odbc 5.3.6 from oracle but it 
segfaults every time



can you share your working mysql+odbc combination?

thanks


odbcinst.ini

[ODBC]
Trace=No
Pooling=Yes
Threading=0

[MySQL ODBC 5.3 Unicode Driver]
Driver=/usr/lib/libmyodbc5w.so
UsageCount=1


odbc.ini

[pbxdb-connector]
Description   = MySQL connection to 'pbxdb' database
Driver= MySQL ODBC 5.3 Unicode Driver
Database  = pbxdb
Server= localhost
UserName  = top
Password  = secret
Port  = 3306
Socket= /var/lib/mysql/mysql.sock
#SSLCIPHER = AES256-SHA
Charset   = utf8



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Re: [asterisk-users] asterisk odbc segfaults

2016-05-27 Thread Marek Červenka 

btw info from my segfault

Core was generated by `/usr/sbin/asterisk -f -vvvg -c'.
Program terminated with signal 11, Segmentation fault.
#0  fix_result_types (stmt=0x9741730)
at /usr/src/debug/mysql-connector-odbc-5.3.6-src/driver/utility.c:139
139 irrec->row.field= field;


Dne 27.5.2016 v 17:58 Marek Červenka napsal(a):


hi,

i have the same problems as in 
https://issues.asterisk.org/jira/browse/ASTERISK-25833


my current combination is centos 6 32-bit, unixODBC 2.3.2 (recompiled 
from fedora20), mysql 5.1.73, mysql-connector-odbc 5.1.5, asterisk 13.9.1


i tried update to mysql-connector-odbc 5.3.6 from oracle but it 
segfaults every time



can you share your working mysql+odbc combination?

thanks


odbcinst.ini

[ODBC]
Trace=No
Pooling=Yes
Threading=0

[MySQL ODBC 5.3 Unicode Driver]
Driver=/usr/lib/libmyodbc5w.so
UsageCount=1


odbc.ini

[pbxdb-connector]
Description   = MySQL connection to 'pbxdb' database
Driver= MySQL ODBC 5.3 Unicode Driver
Database  = pbxdb
Server= localhost
UserName  = top
Password  = secret
Port  = 3306
Socket= /var/lib/mysql/mysql.sock
#SSLCIPHER = AES256-SHA
Charset   = utf8



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[asterisk-users] asterisk odbc segfaults

2016-05-27 Thread Marek Červenka 

hi,

i have the same problems as in 
https://issues.asterisk.org/jira/browse/ASTERISK-25833


my current combination is centos 6 32-bit, unixODBC 2.3.2 (recompiled 
from fedora20), mysql 5.1.73, mysql-connector-odbc 5.1.5, asterisk 13.9.1


i tried update to mysql-connector-odbc 5.3.6 from oracle but it 
segfaults every time



can you share your working mysql+odbc combination?

thanks


odbcinst.ini

[ODBC]
Trace=No
Pooling=Yes
Threading=0

[MySQL ODBC 5.3 Unicode Driver]
Driver=/usr/lib/libmyodbc5w.so
UsageCount=1


odbc.ini

[pbxdb-connector]
Description   = MySQL connection to 'pbxdb' database
Driver= MySQL ODBC 5.3 Unicode Driver
Database  = pbxdb
Server= localhost
UserName  = top
Password  = secret
Port  = 3306
Socket= /var/lib/mysql/mysql.sock
#SSLCIPHER = AES256-SHA
Charset   = utf8



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Re: [asterisk-users] pjsip segfault problem

2016-05-26 Thread Marek Červenka 
no, i added patch https://trac.pjsip.org/repos/changeset/5233 to 
pjproject bundled


testsuite with wss calls is still running

Dne 26.5.2016 v 16:11 Matt Fredrickson napsal(a):

Have you tried updating to pjproject version 2.5.x?  It should have
the patch that you listed in your other email, which I believe should
be included in that branch.

Hope that helps, and best of luck.

Matthew Fredrickson

On Thu, May 26, 2016 at 4:11 AM, Marek Červenka <cerv...@fpf.slu.cz> wrote:

hi,

after switch from 13.7 + pjproject 2.4.5 to 13.9.1 pjproject bundled i have
problem with segfault  (centos 6)

Program terminated with signal 11, Segmentation fault.
#0  0xb7665695 in check_cached_response (sess=0xafbd688c, packet=0xb07676d8,
pkt_size=132, options=1, token=0xafecc2bc, parsed_len=0x0,
src_addr=0xb0e47a20, src_addr_len=16)
 at ../src/pjnath/stun_session.c:1287
1287if (t->msg_magic == msg->hdr.magic &&


it was only once after 2 days.  i dont know how to repeat it now :(

any similiar experience?


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Re: [asterisk-users] pjsip segfault problem

2016-05-26 Thread Marek Červenka 

it looks like i faced this problem
https://trac.pjsip.org/repos/changeset/5233
https://issues.asterisk.org/jira/browse/ASTERISK-25275

Dne 26.5.2016 v 11:11 Marek Červenka napsal(a):


hi,

after switch from 13.7 + pjproject 2.4.5 to 13.9.1 pjproject bundled i 
have problem with segfault  (centos 6)


Program terminated with signal 11, Segmentation fault.
#0  0xb7665695 in check_cached_response (sess=0xafbd688c, 
packet=0xb07676d8, pkt_size=132, options=1, token=0xafecc2bc, 
parsed_len=0x0, src_addr=0xb0e47a20, src_addr_len=16)

at ../src/pjnath/stun_session.c:1287
1287if (t->msg_magic == msg->hdr.magic &&


it was only once after 2 days.  i dont know how to repeat it now :(

any similiar experience?


--
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===





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[asterisk-users] pjsip segfault problem

2016-05-26 Thread Marek Červenka 

hi,

after switch from 13.7 + pjproject 2.4.5 to 13.9.1 pjproject bundled i 
have problem with segfault  (centos 6)


Program terminated with signal 11, Segmentation fault.
#0  0xb7665695 in check_cached_response (sess=0xafbd688c, 
packet=0xb07676d8, pkt_size=132, options=1, token=0xafecc2bc, 
parsed_len=0x0, src_addr=0xb0e47a20, src_addr_len=16)

at ../src/pjnath/stun_session.c:1287
1287if (t->msg_magic == msg->hdr.magic &&


it was only once after 2 days.  i dont know how to repeat it now :(

any similiar experience?


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Re: [asterisk-users] Asterisk 13.8.0 Now Available

2016-03-31 Thread Marek Červenka 

Dne 30.3.2016 v 14:34 Joshua Colp napsal(a):

Marek Červenka wrote:

and what about
https://www.asterisk-blog.com/2016/02/17/odbc_gutting/


While not in the email these are listed in the CHANGES and UPGRADE.txt 
file. Going forward we'll try to ensure we include such things in the 
release notes as well.




can you add this example from centos 6 ?

[ODBC]
Trace = No

#http://www.unixodbc.org/doc/conn_pool.html
Pooling = Yes

#http://stackoverflow.com/questions/4207458/using-unixodbc-in-a-multithreaded-concurrent-setting
Threading = 0

# Setup from the unixODBC package
[PostgreSQL]
Description = ODBC for PostgreSQL
Driver  = /usr/lib/psqlodbcw.so
Setup   = /usr/lib/libodbcpsqlS.so
Driver64= /usr/lib64/psqlodbcw.so
Setup64 = /usr/lib64/libodbcpsqlS.so
FileUsage   = 1

# Driver from the mysql-connector-odbc package
# Setup from the unixODBC package
[MySQL]
Description = ODBC for MySQL
Driver  = /usr/lib/libmyodbc5.so
Setup   = /usr/lib/libodbcmyS.so
Driver64= /usr/lib64/libmyodbc5.so
Setup64 = /usr/lib64/libodbcmyS.so
FileUsage   = 1

to https://wiki.asterisk.org/wiki/display/AST/ODBC

and the text from https://www.asterisk-blog.com/2016/02/17/odbc_gutting/

thanks

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Re: [asterisk-users] Asterisk 13.8.0 Now Available

2016-03-30 Thread Marek Červenka 

and what about
https://www.asterisk-blog.com/2016/02/17/odbc_gutting/

Dne 30.3.2016 v 0:12 Asterisk Development Team napsal(a):

The Asterisk Development Team has announced the release of Asterisk 13.8.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 13.8.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following are the issues resolved in this release:

New Features made in this release:
---
  * ASTERISK-24919 - res_pjsip_config_wizard: Ability to write
   contents to file (Reported by Ray Crumrine)
  * ASTERISK-25670 - Add regcontext to PJSIP (Reported by Daniel
   Journo)
  * ASTERISK-25480 - [patch]Add field PauseReason on
   QueueMemberStatus (Reported by Rodrigo Ramirez Norambuena)

Bugs fixed in this release:
---
  * ASTERISK-25849 - chan_pjsip: transfers with direct media
   sometimes drops audio (Reported by Kevin Harwell)
  * ASTERISK-25113 - install_prereq in Debian 8 without "standard
   system utilities" (Reported by Rodrigo Ramirez Norambuena)
  * ASTERISK-25814 - Segfault at f ip in res_pjsip_refer.so
   (Reported by Sergio Medina Toledo)
  * ASTERISK-25023 - Deadlock in chan_sip in
   update_provisional_keepalive (Reported by Arnd Schmitter)
  * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
   channel (Reported by Filip Frank)
  * ASTERISK-25829 - res_pjsip: PJSIP does not accept spaces when
   separating multiple AORs (Reported by Mateusz Kowalski)
  * ASTERISK-25771 - ARI:Crash - Attended transfers of channels into
   Stasis application. (Reported by Javier Riveros )
  * ASTERISK-25830 - Revision 2451d4e breaks NAT (Reported by Sean
   Bright)
  * ASTERISK-25582 - Testsuite: Reactor timeout error in
   tests/fax/pjsip/directmedia_reinvite_t38 (Reported by Matt
   Jordan)
  * ASTERISK-25811 - Unable to delete object from sorcery cache
   (Reported by Ross Beer)
  * ASTERISK-25800 - [patch] Calculate talktime when is first call
   answered (Reported by Rodrigo Ramirez Norambuena)
  * ASTERISK-25727 - RPM build requires OPTIONAL_API cflag due to
   PJSIP requirement (Reported by Gergely Dömsödi)
  * ASTERISK-25337 - Crash on PJSIP_HEADER Add P-Asserted-Identity
   when calling from Gosub (Reported by Jacques Peacock)
  * ASTERISK-25738 - res_pjsip_pubsub: Crash while executing
   OutboundSubscriptionDetail ami action (Reported by Kevin
   Harwell)
  * ASTERISK-25721 - [patch] res_phoneprov: memory leak and
   heap-use-after-free (Reported by Badalian Vyacheslav)
  * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
   returns garbage (Reported by Etienne Lessard)
  * ASTERISK-25751 - res_pjsip: Support
   pjsip_dlg_create_uas_and_inc_lock (Reported by Joshua Colp)
  * ASTERISK-25606 - Core dump when using transports in sorcery
   (Reported by Martin Moučka)
  * ASTERISK-20987 - non-admin users, who join muted conference are
   not being muted (Reported by hristo)
  * ASTERISK-25737 - res_pjsip_outbound_registration: line option
   not in Alembic (Reported by Joshua Colp)
  * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
   udptl_rx_packet cause ast_frdup crash (Reported by Walter
   Doekes)
  * ASTERISK-25742 - Secondary IFP Packets can result in accessing
   uninitialized pointers and a crash (Reported by Torrey Searle)
  * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
   Vulnerability - Investigate vulnerability of HTTP server
   (Reported by Alex A. Welzl)
  * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
   non-default timert1 (Reported by Alexander Traud)
  * ASTERISK-25702 - PjSip realtime DB and Cache Errors since
   upgrade to asterisk-13.7.0 from asterisk-13.7.0-rc2 (Reported by
   Nic Colledge)
  * ASTERISK-25730 - build:  make uninstall after make distclean
   tries to remove root (Reported by George Joseph)
  * ASTERISK-25725 - core: Incorrect XML documentation may result in
   weird behavior (Reported by Joshua Colp)
  * ASTERISK-25722 - ASAN & testsute: stack-buffer-overflow in
   sip_sipredirect (Reported by Badalian Vyacheslav)
  * ASTERISK-25709 - ARI: Crash can occur due to race condition when
   attempting to operate on a hung up channel (Reported by Mark
   Michelson)
  * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
   by Badalian Vyacheslav)
  * ASTERISK-25685 - infrastructure: Run alembic in Jenkins build
   script (Reported by Joshua Colp)
  * ASTERISK-25712 - Second call to already-on-call phone and
   Asterisk sends "Ready" (Reported by Richard Mudgett)
  * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
   (Reported by Badalian Vyacheslav)
  * ASTERISK-25179 -

Re: [asterisk-users] Asterisk 13.8.0 Now Available

2016-03-30 Thread Marek Červenka 

there is no info about --with-pjproject-bundled

i tried it (centos6 32bit)
./configure --with-pjproject-bundled

checking for SSL_library_init in -lssl... yes
OpenSSL library found, SSL support enabled
./aconfigure: line 14995: syntax error near unexpected token `fi'
./aconfigure: line 14995: `fi'
make: *** [build.mak] Error 2
make: Leaving directory 
`/root/rpmbuild/SOURCES/asterisk-13.8.0/third-party/pjproject'


vim ./third-party/pjproject/source/aconfigure
# Check whether --with-opencore-amrnb was given.
if test "${with_opencore_amrnb+set}" = set; then
  withval=$with_opencore_amrnb; { { $as_echo "$as_me:$LINENO: error: 
This option is obsolete and replaced by --with-opencore-amr=DIR" >&5
$as_echo "$as_me: error: This option is obsolete and replaced by 
--with-opencore-amr=DIR" >&2;}

   { (exit 1); exit 1; }; }
else
???missing something???

fi



Dne 30.3.2016 v 0:12 Asterisk Development Team napsal(a):

The Asterisk Development Team has announced the release of Asterisk 13.8.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 13.8.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following are the issues resolved in this release:

New Features made in this release:
---
  * ASTERISK-24919 - res_pjsip_config_wizard: Ability to write
   contents to file (Reported by Ray Crumrine)
  * ASTERISK-25670 - Add regcontext to PJSIP (Reported by Daniel
   Journo)
  * ASTERISK-25480 - [patch]Add field PauseReason on
   QueueMemberStatus (Reported by Rodrigo Ramirez Norambuena)

Bugs fixed in this release:
---
  * ASTERISK-25849 - chan_pjsip: transfers with direct media
   sometimes drops audio (Reported by Kevin Harwell)
  * ASTERISK-25113 - install_prereq in Debian 8 without "standard
   system utilities" (Reported by Rodrigo Ramirez Norambuena)
  * ASTERISK-25814 - Segfault at f ip in res_pjsip_refer.so
   (Reported by Sergio Medina Toledo)
  * ASTERISK-25023 - Deadlock in chan_sip in
   update_provisional_keepalive (Reported by Arnd Schmitter)
  * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
   channel (Reported by Filip Frank)
  * ASTERISK-25829 - res_pjsip: PJSIP does not accept spaces when
   separating multiple AORs (Reported by Mateusz Kowalski)
  * ASTERISK-25771 - ARI:Crash - Attended transfers of channels into
   Stasis application. (Reported by Javier Riveros )
  * ASTERISK-25830 - Revision 2451d4e breaks NAT (Reported by Sean
   Bright)
  * ASTERISK-25582 - Testsuite: Reactor timeout error in
   tests/fax/pjsip/directmedia_reinvite_t38 (Reported by Matt
   Jordan)
  * ASTERISK-25811 - Unable to delete object from sorcery cache
   (Reported by Ross Beer)
  * ASTERISK-25800 - [patch] Calculate talktime when is first call
   answered (Reported by Rodrigo Ramirez Norambuena)
  * ASTERISK-25727 - RPM build requires OPTIONAL_API cflag due to
   PJSIP requirement (Reported by Gergely Dömsödi)
  * ASTERISK-25337 - Crash on PJSIP_HEADER Add P-Asserted-Identity
   when calling from Gosub (Reported by Jacques Peacock)
  * ASTERISK-25738 - res_pjsip_pubsub: Crash while executing
   OutboundSubscriptionDetail ami action (Reported by Kevin
   Harwell)
  * ASTERISK-25721 - [patch] res_phoneprov: memory leak and
   heap-use-after-free (Reported by Badalian Vyacheslav)
  * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
   returns garbage (Reported by Etienne Lessard)
  * ASTERISK-25751 - res_pjsip: Support
   pjsip_dlg_create_uas_and_inc_lock (Reported by Joshua Colp)
  * ASTERISK-25606 - Core dump when using transports in sorcery
   (Reported by Martin Moučka)
  * ASTERISK-20987 - non-admin users, who join muted conference are
   not being muted (Reported by hristo)
  * ASTERISK-25737 - res_pjsip_outbound_registration: line option
   not in Alembic (Reported by Joshua Colp)
  * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
   udptl_rx_packet cause ast_frdup crash (Reported by Walter
   Doekes)
  * ASTERISK-25742 - Secondary IFP Packets can result in accessing
   uninitialized pointers and a crash (Reported by Torrey Searle)
  * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
   Vulnerability - Investigate vulnerability of HTTP server
   (Reported by Alex A. Welzl)
  * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
   non-default timert1 (Reported by Alexander Traud)
  * ASTERISK-25702 - PjSip realtime DB and Cache Errors since
   upgrade to asterisk-13.7.0 from asterisk-13.7.0-rc2 (Reported by
   Nic Colledge)
  * ASTERISK-25730 - build:  make uninstall after make distclean
   tries to remove root (Reported by George Joseph)
  * ASTERISK-25725 - core: Incorrect XML documentation

Re: [asterisk-users] Asterisk 13 and WebRTC. Is wiki page still valid ?

2016-02-19 Thread Marek Červenka 

on my own server

i want try jssip
https://github.com/versatica/JsSIP
it looks like a lot "livelier" than sipml5

any experience with jssip?


Dne 18.2.2016 v 16:01 Olivier napsal(a):



2016-02-18 15:42 GMT+01:00 Marek Červenka <cerv...@fpf.slu.cz 
<mailto:cerv...@fpf.slu.cz>>:


my experience with pjsip for webrtc
http://lists.digium.com/pipermail/asterisk-users/2015-September/287562.html


Yes I saw this post earlier today.
Having to fight 14 days scared me a bit !

Did you set sipml5 on your own server or did you use Live demo 
(https://www.doubango.org/sipml5/call.htm?svn=241) ?


Dne 18.2.2016 v 15:36 Olivier napsal(a):



2016-02-18 14:57 GMT+01:00 Simon Hohberg
<simon.hohb...@mcs-datalabs.com
<mailto:simon.hohb...@mcs-datalabs.com>>:


Is it implied here that both HTTPS and WSS must also come
from the same server (Same Origin Policy) ?

No, the same origin policy does not apply to web sockets.

Then, can I also install my own WebRTC demo page on my
own private  Asterisk server and access this demo page
through HTTPS ?
If I'm not mistaken, this should fulfill all requirements.

It doesn't matter where the asterisk server is hosted. It is
important where the web application comes from. If you don't
want to use https and wss you only have the option to host
the web app locally (on the same machine as the browser that
loads the page), which probably makes sense only for
development. Otherwise you have to use https and wss for the
reasons discussed earlier.

Hope it helps.



At least, it helped me to realize I still have several more
things to learn ;-)

My setup is the following:
- an asterisk server,
- a PC,
- asterisk server and PC are installed on the same LAN
- sipM5 live demo outside my LAN
- no NAT/PAT configuration allowing incoming communications from
the outside.

Is using sipML live demo as a way to rapidly test private
Asterisk WebRTC capabilies, something achievable ?
What would keep this from working ?



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Re: [asterisk-users] Asterisk 13 and WebRTC. Is wiki page still valid ?

2016-02-18 Thread Marek Červenka 

my experience with pjsip for webrtc
http://lists.digium.com/pipermail/asterisk-users/2015-September/287562.html


Dne 18.2.2016 v 15:36 Olivier napsal(a):



2016-02-18 14:57 GMT+01:00 Simon Hohberg 
>:



Is it implied here that both HTTPS and WSS must also come from
the same server (Same Origin Policy) ?

No, the same origin policy does not apply to web sockets.

Then, can I also install my own WebRTC demo page on my own
private  Asterisk server and access this demo page through HTTPS ?
If I'm not mistaken, this should fulfill all requirements.

It doesn't matter where the asterisk server is hosted. It is
important where the web application comes from. If you don't want
to use https and wss you only have the option to host the web app
locally (on the same machine as the browser that loads the page),
which probably makes sense only for development. Otherwise you
have to use https and wss for the reasons discussed earlier.

Hope it helps.



At least, it helped me to realize I still have several more things to 
learn ;-)


My setup is the following:
- an asterisk server,
- a PC,
- asterisk server and PC are installed on the same LAN
- sipM5 live demo outside my LAN
- no NAT/PAT configuration allowing incoming communications from the 
outside.


Is using sipML live demo as a way to rapidly test private Asterisk 
WebRTC capabilies, something achievable ?

What would keep this from working ?





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[asterisk-users] siemens openstage provisioning

2016-02-17 Thread Marek Červenka 

hi,

one of my client have hundreds of siemens openstage phones
i want implement provisioning (1) for Asterisk and donate the code to 
some OSS provisioning project


can you recommend some "live" provisioning project?

thanks

(1) 
http://wiki.unify.com/images/c/c7/OpenStage_Provisioning_Interface_Developer%27s_Guide.pdf


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Re: [asterisk-users] sql schema without alembic

2016-02-08 Thread Marek Červenka 

Dne 4.2.2016 v 12:17 A J Stiles napsal(a):

On Thursday 04 Feb 2016, Marek Červenka wrote:

hi,

is there way to get SQL schema for Asterisk 13.7.0 without alembic?
thanks

Assuming you already have Asterisk up and running, you can just use

$ mysqldump -d -uroot DATABASE TABLE1 TABLE2 TABLE3 ...

will print  (on STDOUT, so you can just use > to write it to a new file, >> to
join onot an existing file or | to pass it through a program)  the SQL
statements required to recreate the given tables in the given database.  (If
you don't specify any tables, it will assume you mean all of them.)

The -d means "don't dump any data", so it will dump just the CREATE statements
and not the INSERT statements that would actually populate the database.


i have asterisk 13.3.0 running on box where i cannot install alembic
i need upgrade to latest asterisk.
for me is the best way apply only "alter table ..." commands for upgrade 
13.3.0->13.7.2


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[asterisk-users] sql schema without alembic

2016-02-04 Thread Marek Červenka 

hi,

is there way to get SQL schema for Asterisk 13.7.0 without alembic?
thanks

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[asterisk-users] missing https://github.com/asterisk/asterisk/blob/13.7/asterisk-13.7.0-summary

2016-02-04 Thread Marek Červenka 

hi,

there is missing 
https://github.com/asterisk/asterisk/blob/13.7/asterisk-13.7.0-summary.html
is it a mistake or "feature" of security releases summary ?

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Re: [asterisk-users] asterisk 13 mixmonitor - random missing syllables (SOLVED)

2016-02-02 Thread Marek Červenka 

solved - https://issues.asterisk.org/jira/browse/ASTERISK-25734?filter=13140

Dne 29.1.2016 v 11:46 Brian :: napsal(a):

12 calls isn't under any type of load.

Someone with better understanding of Asterisk internals may chime in here.

Could it be vmware timing? Is timing critical when using mixmonitor?

I've seen > 100 concurrent calls being recorded wtihout issue.



On Fri, Jan 29, 2016 at 10:39 AM, Marek Červenka <cerv...@fpf.slu.cz 
<mailto:cerv...@fpf.slu.cz>> wrote:


Dne 28.1.2016 v 13:37 Brian :: napsal(a):

when you say load - how many concurrent calls? Is there
transcoding happening? sip / PRIs ? what load?



12  concurrent calls

no transcoding

SIP

under 1.5 with 4x  1Ghz vcpus  (its vmware VPS)



On Thu, Jan 28, 2016 at 9:57 AM, Marek Červenka
<cerv...@fpf.slu.cz <mailto:cerv...@fpf.slu.cz>> wrote:

Dne 27.1.2016 v 17:50 A J Stiles napsal(a):

    On Wednesday 27 Jan 2016, Marek Červenka wrote:

Dne 27.1.2016 v 13:14 A J Stiles napsal(a):

    On Wednesday 27 Jan 2016, Marek Červenka wrote:

hi,

i have strange problem with asterisk 13
mixmonitor, recording to wav
(centos6)
when the system is under load, there are
sometimes missing syllable

there arent BIG spikes on cpus
recordings are to ramdisk (/dev/shm)

any hints?

First, try recording to a real disk (preferrably
a separate drive, so
nothing else will be seeking the heads about; and
connected by SATA, not
USB, for full speed).  Does that work any better?

i tried before. IO is not the problem

Are you saying that it records fine when you use a real
disk, but not with a
ramdisk?

And why are you using a ramdisk for your mixmonitor
recordings?


i have problem in both scenarios
im using ramdisk because is faster and IO cannot be problem





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Re: [asterisk-users] asterisk 13 mixmonitor - random missing syllables

2016-01-29 Thread Marek Červenka 

Dne 28.1.2016 v 13:37 Brian :: napsal(a):
when you say load - how many concurrent calls? Is there transcoding 
happening? sip / PRIs ? what load?




12  concurrent calls

no transcoding

SIP

under 1.5 with 4x  1Ghz vcpus  (its vmware VPS)

On Thu, Jan 28, 2016 at 9:57 AM, Marek Červenka <cerv...@fpf.slu.cz 
<mailto:cerv...@fpf.slu.cz>> wrote:


Dne 27.1.2016 v 17:50 A J Stiles napsal(a):

On Wednesday 27 Jan 2016, Marek Červenka wrote:

Dne 27.1.2016 v 13:14 A J Stiles napsal(a):

On Wednesday 27 Jan 2016, Marek Červenka wrote:

hi,

i have strange problem with asterisk 13
mixmonitor, recording to wav
(centos6)
when the system is under load, there are sometimes
missing syllable

there arent BIG spikes on cpus
recordings are to ramdisk (/dev/shm)

any hints?

First, try recording to a real disk  (preferrably a
separate drive, so
nothing else will be seeking the heads about; and
connected by SATA, not
USB, for full speed).  Does that work any better?

i tried before. IO is not the problem

Are you saying that it records fine when you use a real disk,
but not with a
ramdisk?

And why are you using a ramdisk for your mixmonitor recordings?


i have problem in both scenarios
im using ramdisk because is faster and IO cannot be problem



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Re: [asterisk-users] asterisk 13 mixmonitor - random missing syllables

2016-01-28 Thread Marek Červenka 

Dne 27.1.2016 v 17:50 A J Stiles napsal(a):

On Wednesday 27 Jan 2016, Marek Červenka wrote:

Dne 27.1.2016 v 13:14 A J Stiles napsal(a):

On Wednesday 27 Jan 2016, Marek Červenka wrote:

hi,

i have strange problem with asterisk 13 mixmonitor, recording to wav
(centos6)
when the system is under load, there are sometimes missing syllable

there arent BIG spikes on cpus
recordings are to ramdisk (/dev/shm)

any hints?

First, try recording to a real disk  (preferrably a separate drive, so
nothing else will be seeking the heads about; and connected by SATA, not
USB, for full speed).  Does that work any better?

i tried before. IO is not the problem

Are you saying that it records fine when you use a real disk, but not with a
ramdisk?

And why are you using a ramdisk for your mixmonitor recordings?



i have problem in both scenarios
im using ramdisk because is faster and IO cannot be problem

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[asterisk-users] asterisk 13 mixmonitor - random missing syllables

2016-01-27 Thread Marek Červenka 

hi,

i have strange problem with asterisk 13 mixmonitor, recording to wav 
(centos6)

when the system is under load, there are sometimes missing syllable

there arent BIG spikes on cpus
recordings are to ramdisk (/dev/shm)

any hints?

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Re: [asterisk-users] asterisk 13 mixmonitor - random missing syllables

2016-01-27 Thread Marek Červenka 

Dne 27.1.2016 v 13:14 A J Stiles napsal(a):

On Wednesday 27 Jan 2016, Marek Červenka wrote:

hi,

i have strange problem with asterisk 13 mixmonitor, recording to wav
(centos6)
when the system is under load, there are sometimes missing syllable

there arent BIG spikes on cpus
recordings are to ramdisk (/dev/shm)

any hints?

First, try recording to a real disk  (preferrably a separate drive, so nothing
else will be seeking the heads about; and connected by SATA, not USB, for full
speed).  Does that work any better?



i tried before. IO is not the problem

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[asterisk-users] best practices - ari reconnect

2016-01-18 Thread Marek Červenka 

hi,

can you share your best practices for ARI reconnect when asterisk is 
restarted or when ari app is started before asterisk is fullybooted?


we are using node.js + ari-client so we are thinking about these options:
1) wait for AMI event FullyBooted
2) wait for AMI reconnect and then run ARI reconnect

thanks

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[asterisk-users] asterisk 13 chan_pjsip tcp transport

2015-12-04 Thread Marek Červenka 

hi,

before i fill bug in asterisk issue tracker, is there someone who is 
using chan_pjsip + transport tcp in production with endpoints behind NAT?

thanks

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Re: [asterisk-users] bad performance centos6 ->centos7

2015-11-30 Thread Marek Červenka 

found interesting difference

centos6 have increasing "Function call interrupts" only on cpu4 (its 2x 
quad core system with HT cpu0-cpu15)
 RES:62866052569745188591918425142177536 2138628
17617181838163366894725973741849095 1679889
2736591252322717514321611744   Rescheduling interrupts
 CAL:248362366362 40923894
349357361345 305364367
363337356 352   Function call interrupts


centos7 have "Function call interrupts" distributed to all cpus
RES:  18659  22826  17808  14357  42245 36973  
29927  24703  15279   9098   7984 6564  27054  
23757  19585  16525   Rescheduling interrupts
 CAL: 304288  30969  30869  30555  39649 34481  
32515  28972  11286  22461  22623 22792  25012  
24027  22610  21718   Function call interrupts


any ideas?

Dne 9.11.2015 v 13:28 Marek Červenka napsal(a):

found this interesting article
http://stackoverflow.com/questions/12111954/context-switches-much-slower-in-new-linux-kernels 



running with network-latency profile and its better now but still not 
good as centos6


[root@ast1 ~]# tuned-adm active
Current active profile: network-latency


Dne 6.11.2015 v 10:18 Marek Červenka napsal(a):

hi,

i'm evaluating performance of centos7
i did tests on centos6 x86_64/distro kernel 2.6.32, asterisk 11.16.0 
with 500calls (7sec alaw, simple dialplan, pass trough - sipp 
generators/asterisk receiver with answer/playback)
scenario - sipp generators - asterisk - asterisk receiver (i wrote 
ansible scenario for this if you are interested)


then i reinstalled system to
centos7 x86_64/distro kernel 3.10, asterisk 11.20.0 and run the test 
again


there is big performance hit
https://dl.dropboxusercontent.com/u/44105720/context_interrupts.PNG
https://dl.dropboxusercontent.com/u/44105720/cpu.PNG
https://dl.dropboxusercontent.com/u/44105720/load.PNG

any ideas what tweaks can help?  (it looks like the main problem is 
in interrupts from network card)

your experience with centos7?
any experience with kernel 4.2 from 
http://elrepo.org/linux/kernel/el7/x86_64/RPMS/ ?










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Re: [asterisk-users] asterisk 13 systemd - SOLVED

2015-11-11 Thread Marek Červenka 

Dne 8.11.2015 v 9:13 Tzafrir Cohen napsal(a):

On Sat, Nov 07, 2015 at 09:34:33AM +0100, Ludovic Gasc wrote:

I've some Asterisk 13 on production, it's a custom compilation + I've
retrieved systemd configuration file from asterisk Debian package of
unstable.

After a small adaptation, I've no issue like you, however, I use Debian
Jessie.

Speaking of those (that use the pending review for a systemd unit):

   https://bugs.debian.org/801629



my problem was in highpriority = yes in asterisk.conf ( is equal to -p )

as pointed Tzafrir in https://issues.asterisk.org/jira/browse/ASTERISK-21991

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Re: [asterisk-users] bad performance centos6 ->centos7

2015-11-09 Thread Marek Červenka 

found this interesting article
http://stackoverflow.com/questions/12111954/context-switches-much-slower-in-new-linux-kernels

running with network-latency profile and its better now but still not 
good as centos6


[root@ast1 ~]# tuned-adm active
Current active profile: network-latency


Dne 6.11.2015 v 10:18 Marek Červenka napsal(a):

hi,

i'm evaluating performance of centos7
i did tests on centos6 x86_64/distro kernel 2.6.32, asterisk 11.16.0 
with 500calls (7sec alaw, simple dialplan, pass trough - sipp 
generators/asterisk receiver with answer/playback)
scenario - sipp generators - asterisk - asterisk receiver (i wrote 
ansible scenario for this if you are interested)


then i reinstalled system to
centos7 x86_64/distro kernel 3.10, asterisk 11.20.0 and run the test 
again


there is big performance hit
https://dl.dropboxusercontent.com/u/44105720/context_interrupts.PNG
https://dl.dropboxusercontent.com/u/44105720/cpu.PNG
https://dl.dropboxusercontent.com/u/44105720/load.PNG

any ideas what tweaks can help?  (it looks like the main problem is in 
interrupts from network card)

your experience with centos7?
any experience with kernel 4.2 from 
http://elrepo.org/linux/kernel/el7/x86_64/RPMS/ ?







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[asterisk-users] bad performance centos6 ->centos7

2015-11-06 Thread Marek Červenka 

hi,

i'm evaluating performance of centos7
i did tests on centos6 x86_64/distro kernel 2.6.32, asterisk 11.16.0 
with 500calls (7sec alaw, simple dialplan, pass trough - sipp 
generators/asterisk receiver with answer/playback)
scenario - sipp generators - asterisk - asterisk receiver (i wrote 
ansible scenario for this if you are interested)


then i reinstalled system to
centos7 x86_64/distro kernel 3.10, asterisk 11.20.0 and run the test again

there is big performance hit
https://dl.dropboxusercontent.com/u/44105720/context_interrupts.PNG
https://dl.dropboxusercontent.com/u/44105720/cpu.PNG
https://dl.dropboxusercontent.com/u/44105720/load.PNG

any ideas what tweaks can help?  (it looks like the main problem is in 
interrupts from network card)

your experience with centos7?
any experience with kernel 4.2 from 
http://elrepo.org/linux/kernel/el7/x86_64/RPMS/ ?




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[asterisk-users] asterisk 13 systemd

2015-10-30 Thread Marek Červenka 

hi,

is there somebody using systemd start script on fedora/centos7 + 
asterisk 13 in production?
i have strange problem with high cpu usage when asterisk is started via 
systemd


thanks for feedback

p.s. systemd script is not in vanilla asterisk. only in fedora package
info https://reviewboard.asterisk.org/r/2730/

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Re: [asterisk-users] Calendar integration : Could not authenticate to server: rejected Basic challenge

2015-10-26 Thread Marek Červenka 

try ical url

caldav switched to Oauth
https://blog.mozilla.org/calendar/2013/09/google-is-changing-the-location-url-of-their-caldav-calendars/

and this looks like you must use Oauth 2.0
https://developers.google.com/google-apps/calendar/caldav/v2/guide

Dne 26.10.2015 v 12:17 Jonas Kellens napsal(a):

Hello


I find very little feedback on the following warning/error when trying 
to connect to Google calendar :



[Oct 26 12:11:14] WARNING[24926]: res_calendar_caldav.c:118 
auth_credentials: Invalid username or password for CalDAV calendar 'cal1'
[Oct 26 12:11:14] WARNING[24926]: res_calendar_caldav.c:157 
caldav_request: Unknown response to CalDAV calendar cal1, request 
REPORT to /calendar/dav/i...@mydomain.tld/events/: Could not 
authenticate to server: rejected Basic challenge



[cal1]
type = caldav
url = https://www.google.com/calendar/dav/i...@mydomain.tld/events/
user = i...@mydomain.tld
secret = mysecret
refresh = 15
timeframe = 60



When I go to the URL 
https://www.google.com/calendar/dav/i...@mydomain.tld/events/ I can 
log in with the credentials to the calendar (and get a download window 
for the calendar file).


So it seems not a problem of authentication to me.


But what then could be the real issue here ?





Thanks

Kind regards

Jonas.





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Re: [asterisk-users] Storing HANGUPCAUSE in CDR

2015-10-09 Thread Marek Červenka 

search in archives
save the records to another table like cdr_extended


Dne 7.10.2015 v 15:26 Ross Beer napsal(a):

Hi,

I have the following code that operates when a channel is hung-up:

[record-hangupcause]
exten => 1,n,Set(CDR(hangupcause)=${HANGUPCAUSE})
exten => s,n,Return()


Before the dial a hangup handler is registered:

Set(CHANNEL(hangup_handler_push)=record-hangupcause,s,1)


The routine is called and the variables are being set, however not on 
the channel's CDR which made the call. I believe this is due to the 
CDR being closes as soon as the dial returns.


By changing the cdr option 'endbeforehexten=no' this should keep the 
CDR accessible, however all this does is cause another CDR to be 
created for the 'h' extension.


Is there a way to update the CDR so that a result can be stored per dial?

Thank you in advance,

Ross









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[asterisk-users] Asterisk 13 WebRTC Status report

2015-09-15 Thread Marek Červenka 

hi,

i'm fighting with webrtc for 14 days
reporting my experience to minimize number of crazy asterisk users

i have working webrtc with simpl5 + asterisk 13 + pjproject 2.4.5 + 
chan_pjsip + secure websockets + secure audio + audio in both ways


problems
first, i needed run chan_sip for old hard phones and wss with chan_pjsip 
only for webrtc. this is possible with patch from

https://issues.asterisk.org/jira/browse/ASTERISK-24106

chan_sip is not usable for webrtc because of
https://issues.asterisk.org/jira/browse/ASTERISK-24602

another problem arise with RTP/SAVPF negotiation
this can be solved with patch for Asterisk from
https://issues.asterisk.org/jira/browse/ASTERISK-24602
and for pjsip
http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/2015-September/018607.html

i hope this info helps

what is your experience with WebRTC?

See you at WebRTC Expo Paris :)

p.s. many thanks to my colleague martin tomec for debugging support

p.s.2 relevant part of pjsip.conf

[global]
[transport-wss]
type=transport
protocol=wss;udp,tcp,tls,ws,wss
bind=0.0.0.0

;===ENDPOINT TEMPLATES

[endpoint-basic](!)
type=endpoint
transport=transport-wss
context=route_phones
disallow=all
allow=alaw
allow=ulaw
force_avp=yes
use_avpf=yes; Determines whether res_pjsip will use and enforce usage of
media_encryption=dtls; Determines whether res_pjsip will use and enforce
dtls_verify=no ; Verify that the provided peer certificate is valid 
(default:

dtls_rekey=0   ; Interval at which to renegotiate the TLS session and rekey
dtls_cert_file=/etc/pki/tls/certs/pbx.crt
dtls_private_key=/etc/pki/tls/private/pbx.key
dtls_setup=actpass
ice_support=yes   ;This is specific to clients that support NAT traversal
media_use_received_transport=yes

[auth-userpass](!)
type=auth
auth_type=userpass

[aor-single-reg](!)
type=aor
remove_existing=yes
max_contacts=1


;===DEVICES

[webrtc1](endpoint-basic)
auth=webrtc1
aors=webrtc1

[webrtc1](auth-userpass)
password=secret
username=webrtc1

[webrtc1](aor-single-reg)

relevant part of http.conf
[general]
enabled=yes
bindaddr=0.0.0.0
tlsenable=yes
tlsbindaddr=0.0.0.0:8089
tlscertfile=/etc/pki/tls/certs/pbx.crt
tlsprivatekey=/etc/pki/tls/private/pbx.key

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Re: [asterisk-users] Asterisk 13 WebRTC Status report

2015-09-15 Thread Marek Červenka 

Dne 15.9.2015 v 13:37 Marek Červenka napsal(a):

hi,

i'm fighting with webrtc for 14 days
reporting my experience to minimize number of crazy asterisk users

i have working webrtc with simpl5 + asterisk 13 + pjproject 2.4.5 + 
chan_pjsip + secure websockets + secure audio + audio in both ways


problems
first, i needed run chan_sip for old hard phones and wss with 
chan_pjsip only for webrtc. this is possible with patch from

https://issues.asterisk.org/jira/browse/ASTERISK-24106

chan_sip is not usable for webrtc because of
https://issues.asterisk.org/jira/browse/ASTERISK-24602



this is the blocking issue 
https://issues.asterisk.org/jira/browse/ASTERISK-24146



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Re: [asterisk-users] webrtc no audio

2015-08-28 Thread Marek Červenka 

are you sure you dont have this problem?
https://issues.asterisk.org/jira/browse/ASTERISK-24146

i'm now fighting with
https://issues.asterisk.org/jira/browse/ASTERISK-24602

Dne 27.8.2015 v 20:07 Vinicius Fontes napsal(a):

I have it working now!

*I had to install Asterisk 13 with PJSIP support.That's important, 
even if you won't use PJSIP.* Then I did this configuration, which is 
working fine under NAT:


*sip.conf:*
[6001]
type=friend
secret=REDACTED
host=dynamic
context=interno
disallow=all
;allow=alaw,h263,h264,vp8
allow=g722
dtmf=auto
videosupport=yes
transport=ws,udp
avpf=yes
callerid=WebRTC 6001
encryption=yes
qualify=yes
directmedia=no
nat=force_rport,comedia
icesupport=yes
dtlsenable=yes ; Tell Asterisk to enable DTLS for this peer
dtlsverify=no ; Tell Asterisk to not verify your DTLS certs
dtlscertfile=/etc/asterisk/keys/asterisk.pem ; Tell Asterisk where 
your DTLS cert file is
dtlsprivatekey=/etc/asterisk/keys/asterisk.pem ; Tell Asterisk where 
your DTLS private key is
dtlssetup=actpass ; Tell Asterisk to use actpass SDP parameter when 
setting up DTLS


*rtp.conf:*
icesupport=true
stunaddr=stun.l.google.com:19302 http://stun.l.google.com:19302

*res_stun_monitor.conf:*
stunaddr = stun.l.google.com:19302 http://stun.l.google.com:19302   
 ; Address of the STUN server to query.*

*
stunrefresh = 30

2015-08-12 5:23 GMT-03:00 Marek Červenka cerv...@fpf.slu.cz 
mailto:cerv...@fpf.slu.cz:


Dne 11.8.2015 v 12:18 Joshua Colp napsal(a):

Vinicius Fontes wrote:

I'm having the same issue! The difference in my case is
Asterisk server
has a public IPv4 and the browser is behind a single NAT.

I'm forwarding my configuration below (which I posted
previously on
asterisk-users).

How can we debug ICE negotiation?


You have to do a packet capture, look at the exchange in
Wireshark, and see how the negotiation flows. It requires a
basic understanding of ICE.


it looks like we are facing this problem
https://issues.asterisk.org/jira/browse/ASTERISK-24146 too
if we use [] in sipml5 expert config To disable TURN/STUN to
speedup ICE candidates gathering you can use an empty array. e.g. [].
it works better




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Marek Cervenka
===


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Marek Cervenka
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Re: [asterisk-users] simultaneous use of chan_sip/chan_pjsip

2015-08-27 Thread Marek Červenka 

Dne 13.8.2015 v 21:48 Marek Červenka napsal(a):

Dne 13.8.2015 v 17:20 Rusty Newton napsal(a):
On Thu, Aug 13, 2015 at 3:54 AM, Marek Červenka cerv...@fpf.slu.cz 
mailto:cerv...@fpf.slu.cz wrote:


hello,

is it possible simultaneously use chan_sip and chan_pjsip?

if yes, can you recommend settings

i'm thinking about
- chan_sip - for sip hardphones/softphones  (sip udp 5060)
- chan_pjsip - for webrtc


You can use both.. you will want to make sure your bind addresses and 
ports don't conflict.


Why not use chan_pjsip for all SIP connectivity?


because it's BIG change for production environment
we have own web gui for config generation and we need move to 
chan_pjsip safely


for the record

it looks like the simultaneous use is not possible

with this configuration

sip.conf
[general]
transport=udp
...

pjsip.conf
[global]

[transport-wss]
type=transport
protocol=wss
bind=0.0.0.0
...

module res_pjsip_transport_websocket.so is not loaded and load fails

*CLI module load res_pjsip_transport_websocket.so
[Aug 27 12:31:23] DEBUG[13977]: res_pjsip.c:1918 register_service_noref: 
Registered SIP service WebSocket Transport Module (0xb51353e0)
[Aug 27 12:31:23] DEBUG[13977]: res_pjsip.c:1950 
unregister_service_noref: Unregistered SIP service WebSocket Transport 
Module

Unable to load module res_pjsip_transport_websocket.so
Command 'module load res_pjsip_transport_websocket.so' failed.


*CLI module show like websoc
Module Description  Use Count  Status 
Support Level
res_http_websocket.so  HTTP WebSocket Support   
2  Running  extended
res_pjsip_transport_websocket.so PJSIP WebSocket Transport 
Support0  Not Running  core



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Marek Cervenka
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Re: [asterisk-users] simultaneous use of chan_sip/chan_pjsip

2015-08-27 Thread Marek Červenka 

Dne 27.8.2015 v 12:37 Joshua Colp napsal(a):

On 15-08-27 07:33 AM, Marek Červenka wrote:

Dne 13.8.2015 v 21:48 Marek Červenka napsal(a):

Dne 13.8.2015 v 17:20 Rusty Newton napsal(a):

On Thu, Aug 13, 2015 at 3:54 AM, Marek Červenka
mailto:cerv...@fpf.slu.czcerv...@fpf.slu.cz wrote:

hello,

is it possible simultaneously use chan_sip and chan_pjsip?

if yes, can you recommend settings

i'm thinking about
- chan_sip - for sip hardphones/softphones  (sip udp 5060)
- chan_pjsip - for webrtc


You can use both.. you will want to make sure your bind addresses and
ports don't conflict.

Why not use chan_pjsip for all SIP connectivity?


because it's BIG change for production environment
we have own web gui for config generation and we need move to
chan_pjsip safely


for the record

it looks like the simultaneous use is not possible


Simultaneous use of everything but the websocket support is possible. 
There is an issue open[1] to make that configurable but noone has done 
it as of this time.


[1] https://issues.asterisk.org/jira/browse/ASTERISK-24106



with patch from ticket(disable ws in chan_sip) it works ok
thanks!

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[asterisk-users] simultaneous use of chan_sip/chan_pjsip

2015-08-13 Thread Marek Červenka 

hello,

is it possible simultaneously use chan_sip and chan_pjsip?

if yes, can you recommend settings

i'm thinking about
- chan_sip - for sip hardphones/softphones  (sip udp 5060)
- chan_pjsip - for webrtc

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Marek Cervenka
===


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Re: [asterisk-users] simultaneous use of chan_sip/chan_pjsip

2015-08-13 Thread Marek Červenka 

Dne 13.8.2015 v 17:20 Rusty Newton napsal(a):
On Thu, Aug 13, 2015 at 3:54 AM, Marek Červenka cerv...@fpf.slu.cz 
mailto:cerv...@fpf.slu.cz wrote:


hello,

is it possible simultaneously use chan_sip and chan_pjsip?

if yes, can you recommend settings

i'm thinking about
- chan_sip - for sip hardphones/softphones  (sip udp 5060)
- chan_pjsip - for webrtc


You can use both.. you will want to make sure your bind addresses and 
ports don't conflict.


Why not use chan_pjsip for all SIP connectivity?


because it's BIG change for production environment
we have own web gui for config generation and we need move to chan_pjsip 
safely



--
---
Marek Cervenka
===

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Re: [asterisk-users] webrtc no audio

2015-08-12 Thread Marek Červenka 

Dne 11.8.2015 v 12:18 Joshua Colp napsal(a):

Vinicius Fontes wrote:

I'm having the same issue! The difference in my case is Asterisk server
has a public IPv4 and the browser is behind a single NAT.

I'm forwarding my configuration below (which I posted previously on
asterisk-users).

How can we debug ICE negotiation?


You have to do a packet capture, look at the exchange in Wireshark, 
and see how the negotiation flows. It requires a basic understanding 
of ICE.




it looks like we are facing this problem 
https://issues.asterisk.org/jira/browse/ASTERISK-24146 too
if we use [] in sipml5 expert config To disable TURN/STUN to speedup 
ICE candidates gathering you can use an empty array. e.g. [].

it works better




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