Displaying 20 results from an estimated 10000 matches similar to: "#-all.gsm"
2010 Mar 04
9
30 mins GSM file
I need to create 30 mins of GSM file for Asterisk .
Silent / Blank file.
Whats the best way to create it ?
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2009 Feb 24
8
HDD FULLL
I have 320 GB SATA HDD.
When I checked my phpsysinfo, it shows 95% HDD is filled.
[root at vicidialnow ~]# df
Filesystem 1K-blocks Used Available Use% Mounted on
/dev/sda2 301924504 285002780 1337472 100% /
/dev/sda1 101086 11062 84805 12% /boot
tmpfs 1553832 0 1553832 0% /dev/shm
[root at vicidialnow ~]# du
16896 .
You have new mail in /var/spool/mail/root
[root at vicidialnow ~]# df -i
2009 Jan 23
2
Cant Find
I cant find /usr/local/apache2/conf/httpd.conf .
I have CentOS.
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2009 Apr 14
5
.GSM -> .WAV (or ,MP3) Conversion
Hey there,
I'm trying to convert some call recordings from asterisk we have in .gsm
format to something I can pipe through ffmpeg - wav would be good, mp3
would be amazing!
I've been trying playing with sox but I don't seem to be getting too far
with
1239101491.30.gsm -ql -r 64000 -t wav 1239101491.30.conv.wav resample
as ffmpeg borks at it:
tim at freee-meee:~/dmc/call
2014 Dec 30
2
forcing GSM on certain extensions
I'm trying to force GSM when I call on certain extension but I'm getting connected with "ulaw"
Which is not suitable when bandwidth is low and slow.
my phone is iax-322
in iax.conf
[iaxy-322]
...
disallow=all
allow=gsm
allow=ulaw
allow=alaw
[zoiper_kathy_old_phone]
...
disallow=all
allow=gsm
allow=ilbc
allow=ulaw
allow=alaw
allow=speex
I've define "allow=gsm"
2010 Apr 08
2
IVR menu sound processing for AMR and GSM + live test available
Hi!
We are in process of setting up an audio guide that will cover notable places of
our capital Riga, Latvia.
The target audience are tourists that dials a free phone number from a mobile
handset to listen to a 3 minute introduction to historic place.
All audio, 10+ languages are recorder in studio at 44KHz. The audio is stored on
server in A-law 8KHz because we'll be pushing it through E1
2009 Feb 23
3
GSM codec is a good choice ???
Dear all, I have Asterisk 1.4 with SIP. I have a voicemail implemented
with GSM sound files.
The problem is I have IP phones Utopix HyperPhone 202 which support
only G.729a/u and G.723.1 high/low, but not GSM.
If I choose G.729A the "pass-throu" calls among users are OK, but
Asterisk can't transcode GSM to G.729A in voicemail calls.
My company doesn'y want to pay for a G.729
2009 Feb 27
9
call file concurrency
Is there a convenient way to limit the number of call files (outgoing
directory) that are processed concurrently?
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2009 Aug 20
12
IPKall and FWD
We all know the FWD is NO more available.
How to set up IPKALL so that my Inbound number rings on my eyebeam or xlite
?
Any alternative for FWD ?
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2005 Aug 23
2
SIP powercycle not hanging up
I have Sipura 841's talking to a CVS-HEAD of August 4.
If I disconnect the power to the Sipura, Asterisk does not hang up the
channel.
My sip.conf for this phone looks like:
;
[super1] ; Sipura 841
disallow = all
allow = ulaw
callerid = "super1"
2012 Jan 06
1
Why write your dialplan using Lua?
Hello,
Reading through the Wiki:
"Asterisk supports the ability to write dialplan instructions in the Lua
programming language. This method can be used as an alternative to or in
combination with extensions.conf and/or AEL. PBX lua allows users to use
the full power of lua to develop telephony applications using Asterisk"
My question is, what is the benefit of using Lua? I recently
2009 May 19
9
Hang at 5:34 pm EST
Some at 5:34 pm EST DAILY, all my call get disconnect.
I tried RE-INSTALLATION, I tried Reinstallation on a virgin HDD, but its
same.
I tried changing VOIP provider I tried changing Internet Provider..But no
help..
What could be the reason ?
Here are my enties of crontab :
### recording mixing/compressing/ftping scripts
0,3,6,9,12,15,18,21,24,27,30,33,36,39,42,45,48,51,54,57 * * * *
2007 Oct 03
2
No audio on Zap (T1/PRI) channels
I have 12 T1's going into 3 servers, 4 in each into "Digium, Inc. Wildcard
TE410P Quad-Span togglable E1/T1/J1 card 3.3v (rev 02)" cards.
Each "group" of T1's have the primary D on 24 and the secondary D on 96.
The first server (ts20) and the last server (ts22) can playback
"demo-congrats" fine. The "middle" server (ts21) cannot -- just dead air.
2009 Jan 25
10
CentOS and BAT File
In windows, we use BAT file to execute few series of command , which help us
in not writing each command manually everytime we want to execute those
commands.
In CentOS, I want to do the same thing.
Any Advice ?
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2011 Sep 13
3
sox: Failed reading obd-demo.mp3: Do not understand format type: mp3
Hi,
Can someone please comment about the below issue
[root at host0040 kaushal]# file obd-demo.mp3
obd-demo.mp3: MPEG ADTS, layer III, v1, 256 kBits, 44.1 kHz, Monaural
[root at host0040 kaushal]# sox obd-demo.mp3 -e stat
sox: Failed reading obd-demo.mp3: Do not understand format type: mp3
[root at host0040 kaushal]# sox -V obd-demo.mp3 -r 8000 -c 1 -t ul -w vm-intro.ulaw
sox: Failed reading
2009 Aug 19
2
Newbie: How to copy track from CD for MOH without getting "Junk at beginning of frame ..."
I was copying tracks from CD into mp3 files so that I could use it in
Asterisk 1.4.21.2 MOH. (BTW, I have already secured proper license to
play MOH to callers.)
I used MS Media Player version 11 and rip it at 128kbps (smallest) but
whenever I listen to MOH, I saw the following message on the Asterisk
console.
WARNING[20829]: mp3/interface.c:215 decodeMP3: Junk at the beginning of
frame 49443303
2010 Feb 26
3
: PSTN calls
Hi All,
I have installed astriesk 6 and am able to make calls using sip x-lite.
Its working as I expected.
Now I want to make call from sipx-lite to PSTN using asterisk.
can any please suggest me which Hardware card that I can buy? and use ( pl give me all ths list of cards.... which are good.).
2) what is that I need to do after buying the card to make it talk to the real world PSTN network?
2015 Jun 07
4
Connecting two Asterisk
Hi again!
I always try to get my mobile phone work with my Asterisk.
I tried to install Asterisk on my PC (with public IP), but it has problems,
too...
I think, my UMTS-Provider doesn't want to connect to dynamic IP or my DSL-Provider
does not want it, too, since I have no problem to connect and get a very good
audio quality if I connect to other SIP-Provider or to an Asterisk (SAME
2011 Jul 13
2
TDM400p susceptible to EMI?
I have a TDM400p with 3 fxs and 1 fxo daughter cards.
It's in a mini-itx case with a 'right-angle' PCI riser card so the TDM400p
is 'sandwiched' between the Atom D525 CPU and the 2.5" hard drive.
I'm getting a bunch of clicks and pops on all ports.
Has anybody had a similar experience? Did you find a solution?
--
Thanks in advance,
2010 Mar 24
6
Restarting Asterisk using a script - Thanks to all -
Hi All,
I do have asterisk installed for a call center and I would like to know if
it is possible to create a scipt and execute it from a PC connected to the
Network without accessing the server. This script should restart asterisk
and another service related to aheeva.
The problem now is that each time I have to access using PUTY to the server
to start and run services manually.
Service