Displaying 20 results from an estimated 5000 matches similar to: "How "expensive" are the different codecs? (Regarding CPU time)"
2010 Feb 08
3
High codec translation times on x64
Hi Users,
I was wondering if someone of you have the same thing on CentOS 64x?
asterisk01*CLI> core show translation
Translation times between formats (in microseconds) for one
second of data
Source Format (Rows) Destination Format (Columns)
g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729
speex ilbc g726 g722 siren7 siren14 slin16
g723
2007 Sep 26
2
My G729 problem re-visited
Ok, I built a test system to duplicate my problem and provide myself
a platform that I can mess around with to try and break any features.
My problem is G729 pass-through from a gateway to a phone. I think
I even have transcoding working, which makes me more confused on
what's wrong with my pass-through. It must be a configuration issue.
The basics...
*CLI> core show version
Asterisk
2006 Jan 23
1
Installing the none commercial intel g729 codecs into Asterisk@Home 2.2?
Yep I did the same.
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Francesco Peeters (Asterisk)
Sent: Saturday, 21 January 2006 5:34 PM
To: fbraeuer@gmail.com; Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users]
2010 May 05
1
SIP - SIP over PBX no audio when canreinvite=no
Hello list,
I am trying to solve a problem and after unsucessfully chasing forums
and google for some hours, I turn to you in hope of a solution. I feel
it's just a configuration issue but I just can't get my head wrapped
around it.
The situation is basically this: I have an Asterisk connected to an
Alcatel OmniPCX via SIP. Asterisk only ever does SIP and has no
dedicated hardware
2005 Jan 18
1
No compatible codecs
Original Post
----------------
I have an Asterisk related problem with mutualphone.
I can connect to any number with any softphone that I am using (iaxcomm,
SJPhone, and a few others).
Also I have a Grandstream BT 101. But I cannot call (via Asterisk) to
mutualphone destinations. Other destinations go fine.
A working phone call (e.g. from iaxcomm) gives the following on the
console:
--
2011 Sep 30
1
Core show translation > 4000ms
Hi list,
we have 2 asterisk boxes in VM (kvm) on 2 different Dell servers, one is
Lenny kernel 2.6.26 asterisk 1.6.2.20, the second CentOS 2.6.18 asterisk
1.4.36 (Elastix). Both 64bits, no hardware involved, dahdi on both
machines for meetme timing.
Doing core show translation give on the Lenny server
Translation times between formats (in microseconds) for one
second of data
2009 Oct 13
3
strange transcoding values
Hello guys,
i have a question about a voip gateway we use.
I saw those values typing in cli:
core show translation
g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 slin16
g723 - - - - - - - - - - - - - -
gsm - - 2001 2001 6000 2001 2000 16000 - 34002 - 6000
2007 May 04
2
Asterisk Codec Translation Table
Hello list,
I have always though codec translation table is dircetly connected to system speed, utill i came across this:
in my lab, i have 2 boxes,
First box is an Intel Celeron 1.7 GHZ with 256M RAM:
show translation
Translation times between formats (in milliseconds) for one second of data
Source Format (Rows) Destination Format (Columns)
g723 gsm ulaw alaw
2005 Sep 28
1
Asterisk sound files, audio bandwidth, and sound quality
Hello, everybody:
I'm developing an application using Asterisk and a TDM-400 card.
I understand the concept of the difference between GSM and WAV files
when using Asterisk, but I'm not happy with the sound quality with the
GSM compression. It's merely *acceptable* for a telephone call, but for
anything else, it leaves something to be desired.
Case in point -- if you compare the
2009 Dec 30
2
Skype for Asterisk
Hi Sir,
We have integrated Skype with Asterisk (skype user id:-
rexesbposolutions). Each call which is coming to skype account is
getting transfered to Asterisk Queue. It has following two cases:
case 1: When we call from normal skype account to skype account
(rexesbposolutions), everything is working fine.
case 2: This skype account (rexesbposolutions) has been assigned with a
online virtual
2004 Nov 24
1
Which modem is known to work with asterisk?
Hi!
I want to connect my asterisk system to the PSTN. Now I'm thinking about
using an analog modem as FXO device.
Which modems do run? How is the voice quality? (I looked into the wiki,
but on
<http://www.voip-info.org/tiki-index.php?page=Asterisk%20hardware> no
regular modems were mentioned.
Or should I use a special FXO-card? I don't want to spend 99$ for such a
thing but
2006 Dec 18
2
ZAP problem
when placing calls to the system through SIP, I got these messages,
Dec 19 00:26:55 WARNING[5570]: channel.c:2571 ast_request: No translator
path exists for channel type Zap (native 68) to 256
Dec 19 00:26:55 NOTICE[5570]: app_dial.c:1056 dial_exec_full: Unable to
create channel of type 'Zap' (cause 0 - Unknown)
any explanation for this?
Thanks,
-------------- next part --------------
An
2012 Jun 15
1
Does Asterisk support AMR and AMR-WB
Hi all, I have a project for the 3G related, AMR and AMR-WB support.
I'm using the client develop suite from the PortSIP(http://www.portsip.com),
as their said
support the AMR, AMR-WB with RFC4867.
Now I have to setup a SIP server/SIP PBX in our Lab for test, does the
Asterisk
support these codecs and RFC4867 ? If no, there has any plugin to support
this ?
Also, any other Server/PBX which
2003 Sep 25
3
SIP codecs Errors
Hi all:
I recently update a system from CVS (Asterisk CVS-09/25/03-15:58:42), and I receiving the following message:
*CLI> WARNING[1187305408]: File chan_sip.c, Line 1864 (process_sdp): No compatible codecs!
The "show codecs" command shows:
*CLI> show codecs
1 (1 << 0) G.723.1
2 (1 << 1) GSM
4 (1 << 2) G.711 u-law
8 (1 << 3) G.711 A-law
16 (1 <<
2005 Sep 12
1
optimizing for via C3
Hi
I'm trying to build an Asterisk packages for a C3 system (256MB memory,
cpuinfo below).
/proc/cpuinfo:
processor : 0
vendor_id : CentaurHauls
cpu family : 6
model : 9
model name : VIA Nehemiah
stepping : 8
cpu MHz : 1000.736
cache size : 64 KB
fdiv_bug : no
hlt_bug : no
f00f_bug : no
coma_bug : no
fpu
2017 Apr 19
2
Asterisk 1.8.32.3 : no video (h.264)
Hello
using asterisk 1.8.32.3
I am not able to make a call with video support. I do not know what I am
missing to make this video call.
Codec h264 should be supported.
sip*CLI> core show codecs
Disclaimer: this command is for informational purposes only.
It does not indicate anything about your configuration.
INT BINARY HEX TYPE NAME
2003 Nov 06
3
which channel format number is right?
Hi all,
if i enter a "show codecs" at cli * response with:
1 (1 << 0) G.723.1
2 (1 << 1) GSM
4 (1 << 2) G.711 u-law
8 (1 << 3) G.711 A-law
16 (1 << 4) MPEG-2 layer 3
32 (1 << 5) ADPCM
64 (1 << 6) 16 bit Signed Linear PCM
128 (1 << 7) LPC10
2005 Feb 01
1
chan_sip.c:7296 handle_request: Unable to create/find channel
Hi,
I have installed chan_sip on asterisk-1.0.3 / 5 (tried
both, same result).
My sip phone registers fine.
But when dialing a number, I get:
Feb 2 09:44:45 NOTICE[20380]: chan_sip.c:7296 handle_request: Unable to
create/find channel
...
Feb 2 09:44:52 WARNING[20380]: chan_sip.c:686 retrans_pkt: Maximum
retries exceeded on call 384534305@192.168.1.20 for seqno 219
(Non-critical Response)
2017 Apr 20
2
Asterisk 1.8.32.3 : no video (h.264)
Hello
in sip.conf I have ;
videosupport=yes
Kind regards.
J.
On 20-04-17 13:09, Marcelo Terres wrote:
> I suppose that you enable the video support on sip.conf, right?
>
> Regards,
> Marcelo H. Terres <mhterres at gmail.com>
> IM: mhterres at jabber.mundoopensource.com.br
> https://www.mundoopensource.com.br
> https://twitter.com/mhterres
>
2004 Aug 31
2
Asterisk codecs and packet size
Does anybody knows if it's posible or if there is some develoment in
course to be able to use longer transmit packet sizes (as long as I know
this is fixed in 20ms now) with the compressed voip codecs in asterisk
(g729, g726, gsm, etc).
I need to use asterisk to connect remote sip clients with 24kb bandwidth
lines and I'm using a licences g729 codec but because I can't increase