Jerry Geis wrote:
I just switched from 1.4.30 to 1.6.2
I initiated a call file - same way in 1.4.30 and nothing happened.
I was not aware of changes in the call file to 1.6.2?
I was watching the cli and no error showed or anything.
In the manager.conf I have things setup.
[MyDial]
nik600 wrote:
I was trying to record a call usng Mixmonitor and then convert it
using ffmpeg but the recording file is continuosly growing and ffmpeg
ends the conversion before of the call completion.
Here's my quick and easy eagi script:
#!/bin/sh
cat /dev/fd/3 | sox -t raw -r 8000 -w -s -c
Philipp von Klitzing wrote:
I would like to know if any one have experience with live audio
streaming like 1. Streaming from an online resource
Look at app_ices and icecast.
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Ices
If that doesn't work for some reason (In my case, I
Jonathan Addleman wrote:
However, I can't find any way to interact with an existing confbridge
conference. Surely there's some equivalent to meetme's 'meetme list'
command? Anything else I can use through the cli or manager API? I just
need to list conferences and members. Thanks!
Replying
I've just started switching my project to use confbridge instead of
meetme and app_conference (because of audio glitches that kept appearing
in those applications).
However, I can't find any way to interact with an existing confbridge
conference. Surely there's some equivalent to meetme's
Jeff Brower wrote:
Jonathan-
How did you measure the gaps? Using signal or speech analysis
software to display the recording? If you measure number of
samples between the gaps, does it correspond to multiples of RTP
packet payload length (for example, for 8 kHz G711 multiples of
80
marco.mo...@gmail.com wrote:
It looks to me that u are having clock synchronism problems due to
the fact you are using Virtual Machine so u don't have an ISDN card
generating clock. Are u using what was called ztdummie as clock
source? Can't precise the name of it in chan_dahdi but u have it.
Jeff Brower wrote:
How did you measure the gaps? Using signal or speech analysis
software to display the recording? If you measure number of samples
between the gaps, does it correspond to multiples of RTP packet
payload length (for example, for 8 kHz G711 multiples of 80 samples
between
I'm having a problem with conferences both meetme and app_conference,
though I've done most of the testing with meetme.
Essentially, I get little gaps in the audio - usually fewer than a dozen
or so samples, though it does vary. They seem to occur at random, but I
usually get one ever few
David Backeberg wrote:
Timers are built on the premise that they have access to either a real
timing device, or unobstructed access to a processor which clicks
through a proc cycle at a pre-determined rate. Once you break those
rules, don't be surprised when the timers stop working, and 'bad
Ian Murray wrote:
Forgive the possibly stupid question, but do these problems you describe
apply equally to the dom0 as to any domU's in a xen system? I used to
think not, but now I'm starting to realize that I'm probably mistaken...
Dom0 is still a virtual machine, so I would say so.
I'm still unable to do much with my new 1.6 installation. I just tried
reinstalling, and using the standard debian configuration files, with
just the necessary modifications, in case I had some legacy stuff in
there from earlier versions that was interfering. I'm testing in a xen
domU with
Hello,
I made a post to the forums
(http://forums.digium.com/viewtopic.php?f=1t=72901sid=3d5c2717ca5ab7ad676957ae436d4b51)
but haven't received any replies, so thought I'd try here.
On my debian machine running asterisk 1:1.4.21.2~dfsg-3, I've been
noticing that there's a problem with
In an attempt to fix problems with EAGI delays in 1.4 (see my other
message for more on that), I've tried upgrading to 1.6, in case it's a
bug that's fixed in the newer version.
Unfortunately, I'm having all kinds of trouble with this new install. My
system relies on conferences, and whenever
Tzafrir Cohen wrote:
It actually is (maintained, and a recent version of it is in
stable/testing.
Hmm.. I think several years ago it wasn't... I guess I'm just living in
the past. Sorry about that!
--
Jon-o Addleman - http://www.redowl.ca
___
Jon-o Addleman wrote:
I'm using the ices command to stream a conference to an icecast server.
This is working nicely, for the most part, but the volume is very low.
The streamed ogg vorbis audio is much quieter than what I hear in a
SIP
client, for example (on the same machine with the
pedro noticioso wrote:
hi there guys!
how can I eliminate this message?
[May 11 11:00:46] WARNING[7039]: res_musiconhold.c:506
monmp3thread: Unable to spawn mp3player
[May 11 11:09:06] WARNING[7039]: res_musiconhold.c:424
spawn_mp3: Found no files in
'/var/lib/asterisk/mohmp3'
I'm no
I forgot to add that the built-in support for playing mp3s which
replaced, for some people, the mp123 program, requires asterisk-addons,
which also isn't packaged for debian! There are other possibilities
though. I think you could use mp321 plus sox to convert to the proper
sound format, for
Peter Fern wrote:
Probably because the Local proxy channel drops out once the two sides
have been bridged. If you want the Local chan to stay up, use the /n
parameter and the local channel won't perform the native transfer. This
does have it's own problems, but should do what you want.
Wai Wu wrote:
I notice those options. However, I was looking to start the recording
through a third party control program. I know I can do this via
chanspy, but is there better way?
Not that I know of... I was looking for something kind of similar, and
ended up actually using a conference, and
Jeff Hoppe wrote:
When you say that you tweaked the volumes, is that modifying the
Asterisk code to call sox and soxmix or are you mixing outside of
Asterisk. Also, I used Sox to increase the volume and then Soxmix to
mix the two audio files. Is there a way to just use soxmix to
increase the
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