Hi,
I'm using the Record dialplan Application in an Context. My goal is to get
a single screenshot of the h264 media stream per call.
same => n,Record(/tmp/test.wav,0,10,qk)
I nicely get a File test.h264. Is there a way to Playback this h264 video
file on my computer or convert it somehow? VLC
2018-03-26 23:01 GMT+02:00 Antony Stone <
antony.st...@asterisk.open.source.it>:
> That sounds like a pretty big challenge for Asterisk.
>
As far as it at least claim to record some sort of h264 it doesn't sound as
a to big challenge for me. My plan is to use e.g. ffmpeg to convert the
h264 media
Hello,
I noticed that Earlymedia isn't possible as soon as the Media Stream has to
pass through Asterisk. In my Scenario I need Local Channels and Earlymedia
at the same time. So the Media Stream needs to flow through Asterisk.
Is there a Workaround to get Earlymedia with Local Channels working?
Hello,
I have an Asterisk 15 with PJSIP behind NAT (Amazon EC2).
Now I would like to get Early Media Video working between clients in
different NATed networks. The 183 signalling goes trough perfectly, but
asterisk doesn't forward the Early Media RTP stream from the caller to the
recipent.
I hav
Yes, media is flowing through Asterisk because both client's are behind
different NAT's.
Do I need to do something special in the Call Flow? Or anything additional
to the pjsip.conf?
2018-04-09 16:50 GMT+02:00 Joshua Colp :
> On Mon, Apr 9, 2018, at 11:42 AM, Benjamin Marty wro
e the
call.
2018-04-09 17:02 GMT+02:00 Joshua Colp :
> On Mon, Apr 9, 2018, at 11:53 AM, Benjamin Marty wrote:
> > Yes, media is flowing through Asterisk because both client's are behind
> > different NAT's.
>
> This doesn't answer the question of what is ACTUALLY
12:04 PM, Benjamin Marty wrote:
> > My understanding based on Wireshark analysis is that the signaling works
> > (also the recipent phone is displaying the video frame before accepting
> the
> > call), also the calling phone send video (i see that also via Wireshark)
> > but
/${EXTEN})
2018-04-09 18:14 GMT+02:00 Joshua Colp :
> On Mon, Apr 9, 2018, at 1:05 PM, Benjamin Marty wrote:
> > wohoo, so if I unterstand it correctly with that patch early media video
> > works over the Asterisk server? In other words the Asterisk server get's
> > able to
e Nachricht-
> Von: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] Im Auftrag von Joshua Colp
> Gesendet: Montag, 9. April 2018 18:15
> An: asterisk-users@lists.digium.com
> Betreff: Re: [asterisk-users] Asterisk behind NAT Early Media Vi
I just noticed, the calling device isn't even sending the early media video
stream. It just sends an early media audio stream. Is there propably a
change in the signaling needed?
(On another P2P SIP Server the early media video works.)
2018-04-10 12:29 GMT+02:00 Benjamin Marty :
> Hi
, gsm, ilbc, h264
aors = 7004
auth = auth7004
[7004]
type = aor
max_contacts = 2
[auth7004]
type=auth
auth_type=userpass
password=1234
username=7004
extensions.conf:
[internal]
exten => _700X,1,Dial(PJSIP/${EXTEN})
2018-04-10 16:43 GMT+02:00 Benjamin Marty :
> I just noticed, the cal
& DevOps
>
>
> *COMMEND INTERNATIONAL GMBH *A-5020 Salzburg, Saalachstraße 51
> Tel: +43-662-85 62 25
> Fax: +43-662-85 62 26
> http://www.commend.com
>
>
>
> *Security and Communication by Commend *FN 178618z | LG Salzburg
>
>
>
> *Von:* asterisk-use
that direction? Or can you guide me to the code part where Asterisk is
doing the Port change when a NAT is detected and the Client itself is
sending "fake" RTP Early media traffic to get a NAT Binding for incoming
RTP Early media traffic?
Benjamin
2018-04-11 11:50 GMT+02:00 Joshua Colp :
I'm currently trying to setup an Asterisk Box with a Let's Encrypt
certificate.
I merged privatekey, cert and chain to one file:
cat /etc/letsencrypt/live/domain/privkey.pem >
/etc/asterisk/tls/a-keycert.pem
cat /etc/letsencrypt/live/domain/fullchain.pem >>
/etc/asterisk/tls/a-keycert.pem
My sip
I'm calling a script which needs to wait a certain time and also hold the
call for this time. But the script dialplan application seems to work non
blocking. Is there a way to hold the call/dialplan till the shell script is
finished?
same => n,Set(PUSHRESULT=${SHELL(sendpush.sh)})
--
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Well, my issue. Script wasn't correctly called. The shell thingie works
perfectly in a blocking manner :)
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Hello
I'm currently using Asterisk 13 with the chan_sip sip driver. The
extensions are offloaded via realtime module to a MySQL database (via
ODBC). So basically I have a MySQL Table with the SIP users + SIP passwords
and the other stuff from the standard Asterisk database schema.
Now I want to
Hello,
I have an Asterisk 15.6.0 installation with PJSIP SIP Driver and Sorcery
for Realtime. My Goal is to enforce endpoints to UDP, TCP or TLS. For that
I set the 'transport' column in the endpoint to the corresponding transport
in pjsip.conf. But if I e.g. set the transport to my 'transport-tls
Hello,
I have an Asterisk 16.0.1 installation with PJSIP SIP Driver. I like to get
the useragent in the Dialplan in the form of an Variable to check if it is
allowed to place a Call. Is there anything available to achieve that in
Asterisk? With the old chan_sip driver this was possible with
CHANNE
Found a way to solve it with the following Snippet:
`same =>
n,NoOp(${PJSIP_CONTACT(${PJSIP_AOR(${EXTEN},contact)},user_agent)})`
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