You could use RTMP protocol from asterisk to rtmpsrc in gstreamer. I used
chan_rtmp long time back but it only supported audio back then.
From: asterisk-users On Behalf Of
Jerry Geis
Sent: Friday, March 27, 2020 8:29 AM
To: Development discussion of video media support in Asterisk
Subject:
there.
Shishir Pokharel
Sr. Software Engineer
[Description: Description: Description: on24_logo]
201 3rd Street, Suite 300
San Francisco, CA 94103
shishir.pokha...@on24.com<mailto:shishir.pokha...@on24.com>
415.369.8354
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-user
You could initiate a local channel – one leg enters to the conf. room and
another to application playback to play the file you wanted.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis
Sent: Tuesday, July 14, 2015 1:08 PM
To:
Can you share us your extensions.conf or the dialplan logic for this call?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chandrakant
Solanki
Sent: Monday, October 20, 2014 11:19 PM
To: Asterisk Users Mailing List - Non-Commercial
enable/disable.
On Tue, Oct 21, 2014 at 1:56 PM, Shishir Pokharel
shishir.pokha...@on24.commailto:shishir.pokha...@on24.com wrote:
Can you share us your extensions.conf or the dialplan logic for this call?
From:
asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com
Start from
http://www.voip-info.org/
or
Asterisk : The Future of Telephony Book
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Horace Miles
Sent: Friday, September 05, 2014 12:19 AM
To: asterisk-users@lists.digium.com
Subject:
Asterisk is not started. Start asterisk or look at the logs if there is any
issues .
Try asterisk -vvvgc and debug
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anthony Azzopardi
Sent: Wednesday, September 03, 2014 11:57 AM
You might want to check if videosupport=yes in sip.conf
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
Sent: Tuesday, September 02, 2014 1:52 PM
To: Asterisk Users Mailing List - Non-Commercial
Almost all of the phones has this feature in build (Polycom,CISCO
SPA,Digium etc..) Try going through this link
https://wiki.asterisk.org/wiki/display/AST/Presence+State and setting up the
right subscribe settings on the phone buttons;
-Original Message-
From:
You might be able to do it by asterisk AMD, but personally I haven't used it on
fax detection..
http://www.voip-info.org/wiki/view/Asterisk+cmd+AMD
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tech Support
Sent: Friday, August 08,
how to set up in the sip
proxy server? actually, i'm beginner on asterisk.
thank you.
Hi Jaya,
it would be nice for me if i can assist you, but i don't know to much about
asterisk. i'm sorry
On Fri, Aug 8, 2014 at 3:05 AM, Shishir Pokharel
shishir.pokha...@on24.commailto:shishir.pokha...@on24
You can use sip proxy servers on top of asterisk server to have a
authentication from freeradius, at this point I don’t think asterisk supports
what you are looking for.
Try this
http://www.opensips.org/Documentation/Tutorials-Radius
From: asterisk-users-boun...@lists.digium.com
Uncommenting features.conf is not sufficient, You got to have some logic on
the dialplan to support what you are asking for. If I were you, I would
probably use some dial plan logic with asterisk internal DB .
From: asterisk-users-boun...@lists.digium.com
: [asterisk-users] enable features
may i have an example of what you are describing?
On 7/8/2014 23:13, Shishir Pokharel wrote:
Uncommenting features.conf is not sufficient, You got to have some logic on
the dialplan to support what you are asking for. If I were you, I would
probably use some dial plan
Hi guys
How can I record the confbridge only when after a marked user is logged in to
conference ? Is there any option on the confbridge to start recording when
marked user is logged in instead of when the first user logs in ? I tried
setting up same =
I was in the same place as you are now and following links helped me, thanks to
Matthew Jordan;
*
https://wiki.asterisk.org/wiki/display/AST/New+in+1.8#Newin1.8-AsteriskManagerInterface
*
https://wiki.asterisk.org/wiki/display/AST/New+in+10#Newin10-AsteriskManagerInterface
*
Specification
On Wed, Oct 23, 2013 at 6:17 PM, Shishir Pokharel
shishir.pokha...@on24.commailto:shishir.pokha...@on24.com wrote:
Thanks Rusty.
Do you happen to know is there any changes on data of response Key for Manager
Actions which exists in both AMI 1.0 and AMI 1.3 ?
There are most likely many
Newton
Sent: Wednesday, October 23, 2013 9:42 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk AMI 1.3 Specification
On Tue, Oct 22, 2013 at 2:03 PM, Shishir Pokharel shishir.pokha...@on24.com
wrote:
Hi folks,
We are upgrading from AMI 1.0
Hi folks,
We are upgrading from AMI 1.0 to AMI 1.3 and looking for any documents or AMI
1.3 Specifications. I found AMI 1.4 Specification in wiki.asterisk.org but not
for AMI 1.3. Can someone provide me the link for AMI 1.3 specification ?
Thanks in advance
Shishir
--
Can you post sip debug and the console log for this call?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Markus
Sent: Thursday, October 17, 2013 3:22 PM
To: Asterisk Users Mailing List - Non-Commercial
.
But if the final response is 480 doesn't it mean that the call was placed but
there was no reply?
On Aug 13, 2013 10:30 PM, Shishir Pokharel
shishir.pokha...@on24.commailto:shishir.pokha...@on24.com wrote:
21.1.5http://tools.ietf.org/html/rfc3261#section-21.1.5 183 Session Progress
The 183 (Session
Thanks
On Aug 12, 2013 8:05 PM, Shishir Pokharel
shishir.pokha...@on24.commailto:shishir.pokha...@on24.com wrote:
Which version of asterisk are you using ?
From:
asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun
Which version of asterisk are you using ?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mordechay Kaganer
Sent: Sunday, August 11, 2013 8:59 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] SIP
Read queue configuration esp. QEUUESTRATEGY and agent TIMEOUT.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jarek Jarzebowski
Sent: Tuesday, July 23, 2013 3:04 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
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