Hello List
I would very much like to have some feedback on this. Where do I have to
look ? Is it in the Asterisk version (13.38.3) maybe ? Is it for sure in
my config ?!
Kind regards.
Op 28/06/2023 om 16:14 schreef Jonas Kellens:
Hello list
when trying to set up webRTC
Hello list
when trying to set up webRTC communications with sipjs client package
(tried 0.7.0, 0.10.0 and 0.19.0), I see in the asterisk debug log-file
the following :
DEBUG[30891][C-] chan_sip.c: Processing media-level (audio) SDP
c=IN IP4 99.88.77.66... OK.
DEBUG[30891][C-000
On Fri, Feb 11, 2022 at 9:31 AM Jonas Kellens
mailto:jonas.kell...@telenet.be>> wrote:
Hello
I notice a major difference in what Asterisk console is telling me
(which seems correct) and what Asterisk Manager is telling.
A SIP user is called, and the phone does no
Hello
I notice a major difference in what Asterisk console is telling me
(which seems correct) and what Asterisk Manager is telling.
A SIP user is called, and the phone does not ring. This is the situation.
On Asterisk console I see (which seems to be in line with an unreachable
phone) :
Hello Joshua
could it be a bug ?
I am using asterisk-certified-13.21-cert6
Kind regards.
J.
Op 01-07-21 om 20:20 schreef Joshua C. Colp:
On Thu, Jul 1, 2021 at 3:15 PM Jonas Kellens <mailto:jonas.kell...@telenet.be>> wrote:
Hello Joshua
this is the SIP REGISTER at
happens only once ?
And why should there *ever* be an "Expired" if there is a SIP REGISTER
every 180 seconds ?!
Kind regards.
Op 01-07-21 om 17:41 schreef Joshua C. Colp:
On Thu, Jul 1, 2021 at 12:34 PM Jonas Kellens
mailto:jonas.kell...@telenet.be>> wrote:
Hello Josh
[6] =>
)
So there is a re-register at 11:10:45
How do you explain the "Expired" 10 minutes later ??
Op 30-06-21 om 20:32 schreef Joshua C. Colp:
On Wed, Jun 30, 2021 at 3:28 PM Jonas Kellens
mailto:jonas.kell...@telenet.be>> wrote:
Hello
I see the following
Hello
I see the following event from the Asterisk Manager :
2021-06-30 11:20:55
Array
(
[0] => Event: PeerStatus
[1] => Privilege: system,all
[2] => SystemName: tstv7
[3] => ChannelType: SIP
[4] => Peer: SIP/testacc7700921
[5] => PeerStatus: Unregistered
[6] => Cause
Hello
can anyone explain to me why (and HOW) there is a difference in data
between the Asterisk console "sip show peers" and the realtime MySQL
configuration ?
Using : asterisk-certified-13.21-cert6
Asterisk console data :
/usr/sbin/asterisk -rx 'sip show peers' | grep 660091086
66009108
rence, as I experienced on
Centos 7.9.
Kind regards.
Op 12-02-21 om 19:11 schreef Jonas Kellens:
Hello list
when installing latest DAHDI (dahdi-linux-complete-3.1.0+3.1.0) for
usage with asterisk-certified-13.21-cert6 on CentOS 6.10 all works
well when starting dahdi with "/sb
Hello list
when installing latest DAHDI (dahdi-linux-complete-3.1.0+3.1.0) for
usage with asterisk-certified-13.21-cert6 on CentOS 6.10 all works well
when starting dahdi with "/sbin/service dahdi start".
But when installing the same DAHDI version in CentOS 7.9 I get the error
:*/usr/sbin/
G.Jacobsen:
Why do you want such minimal registration time?
On Tuesday, 8 October 2019, 17:23:03 EEST, Jonas Kellens
wrote:
Hello
is it possible to determine the SIP.conf parameters 'defaultexpirty'
and 'maxexpiry' on a peer basis ?
My default value is 300 seconds, but
Hello
is it possible to determine the SIP.conf parameters 'defaultexpirty' and
'maxexpiry' on a peer basis ?
My default value is 300 seconds, but some specific SIP-clients can only
send a SIP REGISTER every 3600 seconds. In current configuration these
SIP peers now become "Unreachable" after
Hello
I see on the CLI :
tst*CLI> core show hints
-= Registered Asterisk Dial Plan Hints =-
50@blf : SIP/testacc7
State:Idle Watchers 3
6001@blf : Custom:q-6001 State:Idle
Watchers 1
5@blf
Hello
I notice that BLF-buttons on my IP-phone are greyed out and again active
after some time. This goes on and on...
When looking at Asterisk CLI I see in the SIP NOTIFY :
Subscription-State: terminated;reason=timeout
The BLF-buttons turn on again after a new SIP SUBSCRIBE from my
IP-ph
Hello
is this mailing list still active ?
Op 10-05-19 om 14:10 schreef Jonas Kellens:
Hello
I am trying to set up webRTC video calls from my Chrome webbrowser
(Fedora) to my Chrome webbrowser (Windows 10).
There is local video input (I can see myself), but never video on the
receiving
Hello
is this mailing list still active ?
Op 10-05-19 om 14:10 schreef Jonas Kellens:
Hello
I am trying to set up webRTC video calls from my Chrome webbrowser
(Fedora) to my Chrome webbrowser (Windows 10).
There is local video input (I can see myself), but never video on the
receiving
Hello
I am trying to set up webRTC video calls from my Chrome webbrowser
(Fedora) to my Chrome webbrowser (Windows 10).
There is local video input (I can see myself), but never video on the
receiving side.
This is the case in both directions (so it makes no difference which
peer is calling
plan etc?
Have you considered a strictly hardware issue? Memory? HD? MB??
The crystal ball is very cloudy on this one!
John Novack
Jonas Kellens wrote:
Hello
thank you for your answer.
If I read your (and others) reaction correctly I can conclude that
this is an Asterisk problem and n
to version 11, which
did require some syntax changes to the dialplan.
Given that even version 11 is EOL, you really need to put your efforts
into doing the migration rather than tracking this one down
JMO
John Novack
Jonas Kellens wrote:
Hello
using Asterisk 1.8.32.
I notice th
Hello
using Asterisk 1.8.32.
I notice that there is a spontaneous reboot of the Asterisk system from
time to time.
When I look in the logs (verbose file) I noticed that every time this
occurs it's at a moment that there is a MySQL action, be it a lookup or
an insert/update/delete.
I must
Hello list
is there a way to limit the number of dns lookup's for 1 and the same host ?
I see on Asterisk CLI a flood of :
[May 31 15:45:37] > doing dnsmgr_lookup for 'proxy1.sip.x2reg.be'
[May 31 15:45:37] > doing dnsmgr_lookup for 'proxy1.sip.x2reg.be'
[May 31 15:45:37] >
For those never getting a decend answer by the community on this
mailinglist, I share my solution to my video problem :
preferred_codec_only=no
(I had this on 'yes')
Op 26-06-17 om 14:43 schreef Jonas Kellens:
Hello
this is the debug output of a test video call. You
Hello
concerning this question of aug 2012, I am now using 1.8.32.2 and it
seems that the code of app_queue.c has changed.
The function ast_devstate_changed() is no longer used. Can anyone tell
me what it is replaced with ?
Kind regards
Op 18-08-12 om 12:42 schreef Alec Davis:
ts.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas
Kellens
*Sent:* Friday, April 21, 2017 10:18 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Asterisk 1.8.32.3 : no video (h.264)
Hello
you mean while placin
Hello James
I am running asterisk as root, just to 'disable' all issues related to
file rights. So this should not be the problem.
Kind regards.
Op 03-06-17 om 08:09 schreef James Cloos:
"JK" == Jonas Kellens writes:
JK> [Jun 2 14:29:28] ERROR[27360][C-0a
Hello
I get the following error when using our Let's Encrypt ssl certificate
for webRTC calls :
[Jun 2 14:29:28] == DTLS ECDH initialized (secp256r1), faster PFS enabled
[Jun 2 14:29:28] ERROR[27360][C-0ae5]: res_rtp_asterisk.c:1441
ast_rtp_dtls_set_configuration: Specified certificat
ng to keep track of calls through multiple transfers.
On 29 May 2017 at 08:17, Jonas Kellens wrote:
Hello
using Asterisk 1.8.32.3.
What is the best way of knowing a call is being transfered (attended and
unattended) ? And also knowing whereto (sip user) the call is being
transfered and who i
Hello
using Asterisk 1.8.32.3.
What is the best way of knowing a call is being transfered (attended and
unattended) ? And also knowing whereto (sip user) the call is being
transfered and who is the transferer ?
So I can log this information.
Kind regards.
J.
--
,
Marcelo H. Terres
IM: mhter...@jabber.mundoopensource.com.br
https://www.mundoopensource.com.br
https://twitter.com/mhterres
https://linkedin.com/in/marceloterres
On 20 April 2017 at 12:42, Jonas Kellens wrote:
Hello
in sip.conf I have ;
videosupport=yes
Kind regards.
J.
On 20-04-17 13:09
://twitter.com/mhterres
https://linkedin.com/in/marceloterres
On 19 April 2017 at 13:18, Jonas Kellens wrote:
Hello
using asterisk 1.8.32.3
I am not able to make a call with video support. I do not know what I am
missing to make this video call.
Codec h264 should be supported.
sip*CLI> core s
Hello
using asterisk 1.8.32.3
I am not able to make a call with video support. I do not know what I am
missing to make this video call.
Codec h264 should be supported.
sip*CLI> core show codecs
Disclaimer: this command is for informational purposes only.
It does not indicate anything ab
Hello
function sip_header is read-only.
Kind regards.
J.
On 14-04-17 11:28, registrator wrote:
In this case you will help function SIP_HEADER(from)
Sent from: Lenovo P70-A
On Apr 14, 2017 12:04 PM, Jonas Kellens wrote:
Hello
this does not set user field in From-header.
I get
04-17 10:46, registrator wrote:
Hello!
May be you help CALLERID(name) function?
exten => _X.,1,Set(CALLERID(name)=$name)
Then you well see INVITE
SIP : FROM "$name" .
Sent from: Lenovo P70-A
On Apr 14, 2017 10:54 AM, Jonas Kellens wrote:
Hello
any input on this ? How to s
Hello
any input on this ? How to set user-field in From-header with the
Dial()-command in dialplan ?
Kind regards
J.
On 03-04-17 10:25, Jonas Kellens wrote:
Hello
how can I set the fromuser field of the SIP INVITE when using the
Dial()-command in the dialplan ?
None of the below
Hello
in what way does this set the 'fromuser' field in the SIP INVITE ?
Kind regards.
J.
On 05-04-17 22:05, Pete Mundy wrote:
Hi Jonas
Does the information at this link help?
http://the-asterisk-book.com/1.6/funktionen-callerid.html
Pete
On 5/04/2017, at 8:11 pm, Jon
Hello
anyone have some useful input on this ?
Thanks.
On 03-04-17 10:25, Jonas Kellens wrote:
Hello
how can I set the fromuser field of the SIP INVITE when using the
Dial()-command in the dialplan ?
None of the below Dial() command give the correct result :
exten => _XX.,n,Dial(
Hello
how can I set the fromuser field of the SIP INVITE when using the
Dial()-command in the dialplan ?
None of the below Dial() command give the correct result :
exten => _XX.,n,Dial(SIP/${EXTEN}:passwdk5j6::user...@myprovider.biz)
exten =>
_XX.,n,Dial(SIP/${EXTEN}:passwdk5j6::user...@mypr
Hello
I can confirm that touch-ing /etc/asterisk/musiconhold.conf (just open
with vi and close again) and then issuing a 'module reload
res_musiconhold.so' on the Asterisk CLI makes the new files load into
Asterisk.
Very strange !!
I would not know how to automate this through script...
TOOTAI wrote:
Le 23/03/2017 à 20:17, Jonas Kellens a écrit :
Hello
is there any more information on how to reload/read musiconhold files ?
CLI> module reload res_musiconhold
--
Daniel
On 07-03-17 10:46, Jonas Kellens wrote:
Hello
I did not mention it but of course the MOH directory is
Hello
is there any more information on how to reload/read musiconhold files ?
Kind regards.
On 07-03-17 10:46, Jonas Kellens wrote:
Hello
I did not mention it but of course the MOH directory is listed in
/etc/asterisk/musiconhold.conf :
[default]
mode=files
directory=/var/lib/asterisk
ther move the file into the MOH directory or define a new class in
musiconhold.conf that is for your directory.
On Fri, Mar 3, 2017 at 7:19 AM, Jonas Kellens
mailto:jonas.kell...@telenet.be>> wrote:
Hello
using Asterisk 1.8.32.3
Current music on hold :
myserver*CLI&g
Hello
using Asterisk 1.8.32.3
Current music on hold :
myserver*CLI> moh show files
Class: default
File: /var/lib/asterisk/moh/macroform-robot_dity
File: /var/lib/asterisk/moh/macroform-cold_day
File: /var/lib/asterisk/moh/reno_project-system
File: /var/lib/asterisk/moh/manolo_ca
On 21-11-16 17:20, Matthew Jordan wrote:
On Mon, Nov 21, 2016 at 10:05 AM, Jonas Kellens
mailto:jonas.kell...@telenet.be>> wrote:
On 21-11-16 15:17, Matthew Jordan wrote:
On Mon, Nov 21, 2016 at 7:05 AM, Jonas Kellens
mailto:jonas.kell...@telenet.be>> wrote:
On 21-11-16 19:14, Jonas Kellens wrote:
On 21-11-16 17:20, Matthew Jordan wrote:
On Mon, Nov 21, 2016 at 10:05 AM, Jonas Kellens
mailto:jonas.kell...@telenet.be>> wrote:
On 21-11-16 15:17, Matthew Jordan wrote:
On Mon, Nov 21, 2016 at 7:05 AM, Jonas Kellens
mailto:jona
On 21-11-16 17:20, Matthew Jordan wrote:
On Mon, Nov 21, 2016 at 10:05 AM, Jonas Kellens
mailto:jonas.kell...@telenet.be>> wrote:
On 21-11-16 15:17, Matthew Jordan wrote:
On Mon, Nov 21, 2016 at 7:05 AM, Jonas Kellens
mailto:jonas.kell...@telenet.be>> wrote:
On 21-11-16 15:17, Matthew Jordan wrote:
On Mon, Nov 21, 2016 at 7:05 AM, Jonas Kellens
mailto:jonas.kell...@telenet.be>> wrote:
Hello
when using Asterisk version 13.12.2 I notice that it takes up to
30 seconds (sometimes even longer) for a call queue to call its
m
Hello
when using Asterisk version 13.12.2 I notice that it takes up to 30
seconds (sometimes even longer) for a call queue to call its members.
Example 1 :
[Nov 21 08:17:57] pbx.c: Executing [queue@pbx-routing:15]
Queue("SIP/incoming-0246", "myqueue1300,,,") in new stack
[Nov 21 08:1
On 11-10-16 14:44, Joshua Colp wrote:
Jonas Kellens wrote:
Hello
I am experiencing a freeze of the Asterisk proces when issuing a 'sip
reload'.
I have this issue every time on asterisk versions : 13.11.2, 13.11.1,
13.10.0 and certified-13.8-cert3.
I do not have this on versions
On 27-10-16 15:53, Jonas Kellens wrote:
Hello
I'm a bit confused on how to group agents (give agents a group number)
when using realtime queues.
I read on the wiki :
* If you include groups in your queue definition the calls get
routed in the order of the group regardless o
On 26-10-16 23:24, Stefan Tichy wrote:
On Wed, Oct 26, 2016 at 04:57:15PM +0200, Jonas Kellens wrote:
if it is indeed manager.conf that I need to edit then the problem is
that I see no param : tlsdontverifyserver=yes
A comment copied from sip.conf.sample:
"If set to yes, don't
Hello
I'm a bit confused on how to group agents (give agents a group number)
when using realtime queues.
I read on the wiki :
* If you include groups in your queue definition the calls get routed
in the order of the group regardless of the specified strategy. So I
just have a member= l
On 26-10-16 15:03, Dan Jenkins wrote:
On Wed, Oct 26, 2016 at 1:46 PM, Jonas Kellens
mailto:jonas.kell...@telenet.be>> wrote:
Hello
I keep getting the following error when trying to connect to the
Asterisk server using AMI :
$socket = fsockopen("tls://
Hello
I keep getting the following error when trying to connect to the
Asterisk server using AMI :
$socket = fsockopen("tls://11.22.33.44","5039", $errno, $errstr, 5);
Erorr on CLI :
[Oct 26 14:38:19] ERROR[2992]: tcptls.c:609 handle_tcptls_connection:
Problem setting up ssl connection: er
ilson Amaral wrote:
Hi
This happens to me when one peer (provider) is bad !
Try to remove all peers from your sip.conf and gradually add them back!
*From:* Jonas Kellens
*To:* Asterisk Users Mailing List - Non-Comme
Hello
I am experiencing a freeze of the Asterisk proces when issuing a 'sip
reload'.
I have this issue every time on asterisk versions : 13.11.2, 13.11.1,
13.10.0 and certified-13.8-cert3.
I do not have this on versions certified-13.8-cert2,
certified-13.8-cert1 and asterisk 1.8.32.3.
Th
On 02-09-16 11:51, Administrator TOOTAI wrote:
Le 02/09/2016 à 11:26, Jonas Kellens a écrit :
Hello
when setting a local forward (in this case to extension 23) on a SIP
phone, I see the following on the Asterisk CLI :
[Aug 31 14:59:34] -- Got SIP response 302 "Moved Temporarily&
n an issue ?
Have a nice week.
--
Ludovic Gasc (GMLudo)
http://www.gmludo.eu/
2016-09-17 11:47 GMT+02:00 Jonas Kellens <mailto:jonas.kell...@telenet.be>>:
Hello
a call goes out and is answered :
[Sep 17 11:29:52] VERBOSE[23420][C-0051] app_dial.c:
SIP/myprovider
Hello
a call goes out and is answered :
[Sep 17 11:29:52] VERBOSE[23420][C-0051] app_dial.c:
SIP/myprovider-010b is making progress passing it to
SIP/mysippeer-0108
[Sep 17 11:30:05] VERBOSE[23420][C-0051] app_dial.c:
SIP/myprovider-010b answered SIP/mysippeer-0108
[Se
On 10-09-16 09:42, Jonas Kellens wrote:
On 10-09-16 00:50, Richard Mudgett wrote:
On Fri, Sep 9, 2016 at 5:37 PM, Jonas Kellens
mailto:jonas.kell...@telenet.be>> wrote:
Hello
when I type on the Asterisk CLi 'queue show', I first get a list
of my queues and th
On 10-09-16 00:50, Richard Mudgett wrote:
On Fri, Sep 9, 2016 at 5:37 PM, Jonas Kellens
mailto:jonas.kell...@telenet.be>> wrote:
Hello
when I type on the Asterisk CLi 'queue show', I first get a list
of my queues and then the following :
failed to extend
Hello
when I type on the Asterisk CLi 'queue show', I first get a list of my
queues and then the following :
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 323
failed to extend from 240 to 327
failed to extend fr
Hello
when setting a local forward (in this case to extension 23) on a SIP
phone, I see the following on the Asterisk CLI :
[Aug 31 14:59:34] -- Got SIP response 302 "Moved Temporarily" back
from 11.22.33.44:40670
[Aug 31 14:59:34] -- Now forwarding
Local/myaccount184@CallFromQueue-
On 17-08-16 23:24, George Joseph wrote:
On Wed, Aug 17, 2016 at 1:40 PM, Jonas Kellens
mailto:jonas.kell...@telenet.be>> wrote:
On 16-08-16 17:45, George Joseph wrote:
On Tue, Aug 16, 2016 at 3:21 AM, Jonas Kellens
mailto:jonas.kell...@telenet.be>> wrote:
On 17-08-16 23:17, Jonathan H wrote:
On 17 August 2016 at 20:40, Jonas Kellens wrote:
When I compile "--without-pjproject" I loose all webrtc functionality. I get errors about
the lack of "ice-frag and ice-pwd in the SDP-body".
So I guess I DO need pjproject. But I do n
On 16-08-16 17:45, George Joseph wrote:
On Tue, Aug 16, 2016 at 3:21 AM, Jonas Kellens
mailto:jonas.kell...@telenet.be>> wrote:
On 16-08-16 04:38, George Joseph wrote:
On Mon, Aug 15, 2016 at 1:24 PM, Jonas Kellens
mailto:jonas.kell...@telenet.be>> wrote:
Remove yourself !
Don't hijack my thread !
On 17-08-16 14:53, Dario Estupinan wrote:
REMOVE ME please.
2016-08-15 15:16 GMT-05:00 Jonas Kellens <mailto:jonas.kell...@telenet.be>>:
Hello
after I have upgraded from Asterisk 12 to
asterisk-certified-13.8-ce
On 15-08-16 23:00, Carlos Chavez wrote:
I highly recommend that you use alembic to set up your database as
this will make sure you are always using the correct database schema.
You should be able to find the "official" structure in the
contrib/realtime/mysql directory of the Asterisk s
On 16-08-16 04:38, George Joseph wrote:
On Mon, Aug 15, 2016 at 1:24 PM, Jonas Kellens
mailto:jonas.kell...@telenet.be>> wrote:
Hello
using pjproject 2.5.5
using asterisk-certified-13.8-cert1
IIRC there were API changes in pjproject 2.5 that aren't accounted for
Hello
after I have upgraded from Asterisk 12 to asterisk-certified-13.8-cert1
none of my realtime SIP peers (saved in MySQL DB) register anymore.
[Aug 15 22:03:43] NOTICE[30098]: chan_sip.c:28451
handle_request_register: Registration from
'' failed for '78.119.140.190:5076' - Wrong
passwor
Hello
using pjproject 2.5.5
using asterisk-certified-13.8-cert1
Compiled pjproject 2.5.5 with :
./configure CFLAGS="-DNDEBUG -DPJ_HAS_IPV6=1" --prefix=/usr
--libdir=/usr/lib64 --enable-shared --disable-video --disable-sound
--disable-opencore-amr
Compiled Asterisk 13 with
./configure --libdi
me on : CentOS release 6.8 (Final)
Kind regards.
On 12-08-16 17:22, Jonas Kellens wrote:
Hello
running into several problems when installing
asterisk-certified-13.8-cert1 (more then I ever had in Asterisk 11 and
12).
I compile : ./configure --libdir=/usr/lib64 --with-pjproject-bundled
Firs
present on the system
Second, I am not able to start Asterisk with following error :
"/usr/sbin/asterisk: error while loading shared libraries: libpj.so.2:
cannot open shared object file: No such file or directory"
Help appreciated.
Kind regards.
On 12-08-16 16:58, Jonas Kellens w
On 12-08-16 16:38, Joshua Colp wrote:
Jonas Kellens wrote:
Question : I noticed I received an error when installing pjproject
--with-external-srtp
I do not seems to have the srtp capability.
(However I can easily install with "yum install libsrtp-devel")
Can this have anything to d
d regards.
On 12-08-16 15:02, Jonas Kellens wrote:
Hello
setting "nat=no" or omitting "nat=" in peer definition does not help
either. Still no audio.
Why do you think this is a NAT issue ? IP and port information in
SDP-body is correct.
Kind regards.
On 12-08-
000wrtc settings ice should do the same
On Aug 11, 2016 10:00 PM, "Jonas Kellens" <mailto:jonas.kell...@telenet.be>> wrote:
On 11-08-16 18:03, Matt Fredrickson wrote:
On Thu, Aug 11, 2016 at 9:40 AM, Jonas Kellens
mailto:jonas.kell...@telenet.be>>
On 11-08-16 18:03, Matt Fredrickson wrote:
On Thu, Aug 11, 2016 at 9:40 AM, Jonas Kellens wrote:
My main reason not to upgrade to Ast 13 is because I'm afraid of losing
functionality as there are certain functions deprecated/replaced. This can
also cause headache :-)
I will do so if the
nd debug against Asterisk 13 first, and
then you can try backing down versions, rather than reverse. WebRTC
is a rapidly moving target, and has required ongoing changes that may
not have made it into older and feature frozen versions of Asterisk.
Matthew Fredrickson
On Wed, Aug 10, 2016 at 3:01 PM
etsEncrypt certs work fine for this, so
no need to spend out on one.
Switch to Asterisk 13.10 and save yourself a whole lotta headache.
On 11 August 2016 at 15:09, Jonas Kellens <mailto:jonas.kell...@telenet.be>> wrote:
Hello
Using Asterisk 12.8.2.
On 10-08-16 22:03, Ma
ticularly
with regards to interoperating with a modern browser version.
Hope that helps,
Matthew Fredrickson
On Wed, Aug 10, 2016 at 5:02 AM, Jonas Kellens wrote:
On 10-08-16 08:52, Ludovic Gasc wrote:
For WebRTC, I recommend you to use Asterisk 13+.
Have a nice day.
Ludovic Gasc (GMLudo)
http:/
On 10-08-16 08:52, Ludovic Gasc wrote:
For WebRTC, I recommend you to use Asterisk 13+.
Have a nice day.
Ludovic Gasc (GMLudo)
http://www.gmludo.eu/
Hello
then why is there an option in sip.conf and rtp.conf " icesupport=yes" ??
This is no answer to my question.
So again : what am I m
Hello
I'm trying for several days now to get ICE support for my Asterisk 11.23
on CentOS 6.
My call setup : sipml5_webRTC (nat) --> public Asterisk on 178.18.90.230
--> softphone Zoiper
(problem : no audio)
Reverse does not work either.
(problem : failed get local SDP)
I followed this guid
Hello
nobody who can help me with this realm issue ??
On 21-06-16 16:36, Jonas Kellens wrote:
Hello
no matter what I set in sip.conf for the param "realm=blablabla" , I
notice in a wireshark trace file that the realm is completely ignored.
I see that realm value is still
Hello
no matter what I set in sip.conf for the param "realm=blablabla" , I
notice in a wireshark trace file that the realm is completely ignored. I
see that realm value is still 'asterisk', being the default. Why is this ?
(I would like to add a printscreen of the wiresharl trace but then th
Hello
I am trying to use the functions SHARED and IMPORT to share variables
across SIP-channels.
During my use I encounter 2 problems/questions.
Question 1. only 1 shared variable per channel ??
When I set 2 shared variables on a channel, and I read them a bit futher
in the dialplan, there
Hello
so I got this working with Google Calendar and meanwhile also with MS
Exchange.
Does anyone have a working example with Horde Calendar (kronolith)? This
one seems very tough !
Kind regards
Jonas.
On 27-10-15 14:52, Jonas Kellens wrote:
Mark
thank you for your input.
I am
omain.tld/private-6e3543acbc76853414124a/basic.ics
user = i...@domain.tld
secret = mysecretpasswd
refresh = 15
timeframe = 60
So the "Private iCal url" of Google Calendar is the one to go !
Jonas.
On 27-10-15 14:04, Mark Wiater wrote:
On 10/27/2015 8:56 AM, Jonas Kellens wrote:
I ha
ists.digium.com] *On Behalf Of *Jonas
Kellens
*Sent:* Tuesday, October 27, 2015 1:33 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Calendar integration : Could not
authenticate to server: rejected Basic challenge
Hello
I have changed type
r-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 2
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_c
Hello
I notice that priority of queue members is not being respected.
Using mysql realtime.
These are the queue members (in table queue_members) :
Local/queuemem0@ExternalCallFromQueue
Local/queuemem1@ExternalCallFromQueue
Local/queuemem2@ExternalCallFromQueue
Local/queuemem3@ExternalCallFrom
On 12-08-15 16:31, A J Stiles wrote:
On Wednesday 12 Aug 2015, Jonas Kellens wrote:
Hello
I was wondering of it is possible to have Queue Agents with the same
priority (penalty) but with a certain order ?
So I have 20 Agents.
Agent 1 till Agent 10 has penalty 1.
Agent 11 till Agent 15 has
Hello
I was wondering of it is possible to have Queue Agents with the same
priority (penalty) but with a certain order ?
So I have 20 Agents.
Agent 1 till Agent 10 has penalty 1.
Agent 11 till Agent 15 has penalty 2.
(only contacted if 1 -> 10 are busy)
Agent 16 till Agent 20 has penalty 3.
On 07-08-15 13:23, Ethy H. Brito wrote:
On Fri, 07 Aug 2015 12:47:40 +0200
Jonas Kellens wrote:
Hello
I have 2 strange errors when using the Background()-application and
DTMF-input that is received.
First of all, my first 2 lines are not being executed. The first line
being executed is the
Hello
I have 2 strange errors when using the Background()-application and
DTMF-input that is received.
First of all, my first 2 lines are not being executed. The first line
being executed is the Set() application, thus line 3.
Secondly, the received digits (911) is not the same as the EXTE
Hello
I have already several Asterisk servers running with similar
configuration, but now I stumble into a problem.
I have mysql configuration res_config_mysql.conf :
[MyAsteriskDB]
dbhost = 127.0.0.1
dbname = MyAsteriskDB
dbuser = astadmin
dbpass = mysecret
dbport = 3306
dbsock = /var/lib/my
Hello
i have the following field (text string) in a MySQL database :
"${KNUMMER} ${phone_number_to} ${phone_number_from} ${CHANNEL:4}"
I read this string form the database and want to have the dialplan
variables to be replaced with the correct content.
How can I do this ?
Currently this
Hello,
I have the following in my dialplan :
exten => callpark,n,Set(PARKINGDYNPOS=200-210)
exten => callpark,n,Set(PARKINGDYNCONTEXT=parked_001)
exten => callpark,n,Park(2s,parkinglot_001)
I see on the CLI :
[Nov 25 15:08:47] -- Executing [callpark@pbx-routing:10]
Set("SIP/SipT01
On 04-11-14 11:52, Jonas Kellens wrote:
On 04-11-14 11:50, Ishfaq Malik wrote:
On 4 November 2014 10:40, Jonas Kellens <mailto:jonas.kell...@telenet.be>> wrote:
Hello,
I have 5 Asterisk servers all using mysql realtime to store queue
log information.
There is 1
On 04-11-14 11:50, Ishfaq Malik wrote:
On 4 November 2014 10:40, Jonas Kellens <mailto:jonas.kell...@telenet.be>> wrote:
Hello,
I have 5 Asterisk servers all using mysql realtime to store queue
log information.
There is 1 out of 5 servers which stores the data in
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