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
--
---
Marek Cervenka
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
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
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
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
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
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
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
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()
*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
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
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
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
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
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
--
---
Marek Cervenka
===
--
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-
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
, 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 (ce
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
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
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 2
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 upd
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
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
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,
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
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
hi,
can someone confirm that queue_log data are the same if are received via
AMI as if they are saved via ODBC
thanks
--
---
Marek Cervenka
===
--
_
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
her 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,w
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
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
hi,
is there way to get SQL schema for Asterisk 13.7.0 without alembic?
thanks
--
---
Marek Cervenka
===
--
_
-- Bandwidth and Colocation Provided by
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 ?
--
---
Marek Cervenka
===
--
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 <c
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
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?
--
---
Marek Cervenka
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
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)
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
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.sl
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 c
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:
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
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
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
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
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
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