Hi All,
I've been looking for information on how to use asterisk and early media to
allow for a video-preview of the caller at the callee's phone for days...
but I haven't been too successful :(
I found that there seems to be a company 2N Helios IP which claims
(youtube-video) that their SIP
FYI: Joshua Colp already replied to my initial post of this message in
asterisk-app-dev.
he suggested to move it here (asterisk-users)
he so far stated, that early media/Video should theoretically work... but
probably no one tried this in recent times...
looking foreward to receive further
Le 07/05/2014 00:57, Rusty Newton a écrit :
On Thu, May 1, 2014 at 11:08 AM, Administrator TOOTAI ad...@tootai.net wrote:
snip
As explained in one on my previous message, it's a bug, easily reproducible:
take a queues.conf (or sip.conf or iax.conf or voicemail.conf or ...) like
this (what is
Le 07/05/2014 01:28, Steve Edwards a écrit :
On Thu, May 1, 2014 at 11:08 AM, Administrator TOOTAI
ad...@tootai.net wrote:
snip
As explained in one on my previous message, it's a bug, easily
reproducible: take a queues.conf (or sip.conf or iax.conf or
voicemail.conf or ...) like this
Le 07/05/2014 00:57, Rusty Newton a écrit :
[...]
I tried to reproduce using your description here and could not
reproduce the issue.
I tried with both sip.conf and queues.conf.
Making a change in an included .conf file, but NOT the parent .conf
file and then reloading that module from the
Rainer Piper wrote:
perhaps a silly question ...
if a channel switches from simple_bridge to native_bridge ... is the
channel switching to direct_media between the endpoints ?
if so, why doesn't turn direct_media = no and
disable_direct_media_on_nat = yes switching to native_bridge off ?
The
Fronc Hias wrote:
FYI: Joshua Colp already replied to my initial post of this message in
asterisk-app-dev.
he suggested to move it here (asterisk-users)
he so far stated, that early media/Video should theoretically work...
but probably no one tried this in recent times...
looking foreward to
Hi Joshua,
I'll give it a try at ttps://issues.asterisk.org/jira
and I hope my English is good enough to explain the problem. ;-)
wow ... early bird it is 03:36 (PDT) in the morning at your place
Thanks!
Rainer
Am 07.05.2014 12:36, schrieb Joshua Colp:
Rainer Piper wrote:
perhaps a silly
Rainer Piper wrote:
Hi Joshua,
I'll give it a try at ttps://issues.asterisk.org/jira
and I hope my English is good enough to explain the problem. ;-)
wow ... early bird it is 03:36 (PDT) in the morning at your place
The office is in Alabama so it is 6:09AM there. I'm in Atlantic Canada
and I get ready for launch in germany at 13:15 ;-)
Am 07.05.2014 13:09, schrieb Joshua Colp:
Rainer Piper wrote:
Hi Joshua,
I'll give it a try at ttps://issues.asterisk.org/jira
and I hope my English is good enough to explain the problem. ;-)
wow ... early bird it is 03:36 (PDT) in the
upps ... off topic
and typo lunch not launch ;-)
Am 07.05.2014 13:14, schrieb Rainer Piper:
and I get ready for launch in germany at 13:15 ;-)
Am 07.05.2014 13:09, schrieb Joshua Colp:
Rainer Piper wrote:
Hi Joshua,
I'll give it a try at ttps://issues.asterisk.org/jira
and I hope my
Hi,
I tried to turn on Video and get the following cli-WARNING output
-- Executing [8600@outgoing-kamailio:1] Answer(PJSIP/7000-,
) in new stack
0x7f46f41ff2e0 -- Probation passed - setting RTP source address to
192.168.8.203:17200
-- Executing [8600@outgoing-kamailio:2]
On Wed, May 7, 2014 at 2:46 AM, Administrator TOOTAI ad...@tootai.net wrote:
I got it: if the filename is given in totality it's working (as you do it).
It's the #include /path to directorie/*.conf which is not taking in
account (here *.conf description)
That still works for me as well.
I
Le 07/05/2014 16:50, Rusty Newton a écrit :
On Wed, May 7, 2014 at 2:46 AM, Administrator TOOTAI ad...@tootai.net wrote:
I got it: if the filename is given in totality it's working (as you do it).
It's the #include /path to directorie/*.conf which is not taking in
account (here *.conf
Administrator TOOTAI wrote:
Le 07/05/2014 16:50, Rusty Newton a écrit :
On Wed, May 7, 2014 at 2:46 AM, Administrator TOOTAI
ad...@tootai.net wrote:
I got it: if the filename is given in totality it's working (as you
do it).
It's the #include /path to directorie/*.conf which is not taking in
Le 07/05/2014 17:22, Joshua Colp a écrit :
Administrator TOOTAI wrote:
Le 07/05/2014 16:50, Rusty Newton a écrit :
On Wed, May 7, 2014 at 2:46 AM, Administrator TOOTAI
ad...@tootai.net wrote:
I got it: if the filename is given in totality it's working (as you
do it).
It's the #include /path
On Wed, May 7, 2014 at 10:14 AM, Administrator TOOTAI ad...@tootai.net wrote:
Le 07/05/2014 16:50, Rusty Newton a écrit :
I contructed a basic sip.conf, and added this line to the end:
#include /etc/asterisk/sip_includes/*.conf
Here is the point. Modify it the way explained in previous
Le 07/05/2014 18:53, Rusty Newton a écrit :
On Wed, May 7, 2014 at 10:14 AM, Administrator TOOTAI ad...@tootai.net wrote:
Le 07/05/2014 16:50, Rusty Newton a écrit :
I contructed a basic sip.conf, and added this line to the end:
#include /etc/asterisk/sip_includes/*.conf
Here is the point.
On Wed, May 7, 2014 at 12:59 PM, Administrator TOOTAI ad...@tootai.net wrote:
Please try the includes *exactly* as I have them in sip.conf (same
directories name and subdirectories) knowing that local is in /etc/asterisk
I used your identical config to narrow it down. I re-opened
Hi all,
I have a user with an old Mitel PBX connected to a couple of SPA112's. The
user is reporting that their phones ring several times a day and when they
answer the call, all they hear is dial tone or busy signal.
Their PBX guy says that the SPA112's aren't providing line supervision and
salut,
je suis entrain de developper une application jEE avec asterisk qui
consiste à ajouter, supprimer,modifier des clients en utilisant Asterisk au
lieu de base de donnée, donc je suis débutante dans ce domaine,j'ai fait
pour l'instant une connection via asterisk avec API manager , mais je
Most FXS ATAs do not support supervision so they don't work well when plugged
into a PBX's analog FXO (aka CO) ports.
If the Mitel can provide supervision on analog phone ports (i.e. FSX) then you
could use an ATA with FXO ports. If the Mitel does not support supervision on
analog phone
On Tue, May 6, 2014 at 1:01 PM, Richard Kenner ken...@gnat.com wrote:
That is definitely a leak and the fix looks good.
Thanks.
That leak is most likely the one biting you.
It definitely is.
Committed the fix for this leak on Asterisk v12 branch in -r413454.
There is another leak
If I login as a user and run asterisk it connects to pulse and runs.
if I login as root and run the command
su user -c asterisk -vc
it does not connect to pulse and run.
I thought su ran as that user - am I missing something to get the su
command to run
correctly and connect to pulseaudio as
On Wed, 7 May 2014, Meriem Abid wrote:
salut,
je suis entrain de developper une application...
You will have better luck if you can post in English.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com
Committed the fix for this leak on Asterisk v12 branch in -r413452.
This leak also applied to Asterisk v11.
Thanks.
Is this for both the one in the talking callback or the one in
handle_cli_confbridge_kick or both (the fix is similar in both)?
--
On Wed, May 7, 2014 at 4:43 PM, Richard Kenner ken...@gnat.com wrote:
Committed the fix for this leak on Asterisk v12 branch in -r413452.
This leak also applied to Asterisk v11.
Thanks.
Is this for both the one in the talking callback or the one in
handle_cli_confbridge_kick or both (the
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