similar to: How "expensive" are the different codecs? (Regarding CPU time)

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

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

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 <<

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

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

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]

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

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

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: --

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

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

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

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

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

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

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

I have configured my GS HT-486 for "send dtmf" in audio, and on the asterisk box, sip.conf has dtmfmode set to inband. Everything seems to be working fine, however, I see my console get flooded with the following warning: "dsp.c:1467 ast_dsp_process: Unable to process inband DTMF on 256 frames" Should I be cautious about them or just ignore? Better still, what should I do

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

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 >

I'm trying to play a wav-file on a channel. This is what I see in the asterisk debug console AGI Rx << STREAM FILE "test.wav" "12345" [Aug 20 16:10:19] WARNING[25219]: file.c:602 ast_openstream_full: File test.wav does not exist in any format So it doesn't find the file, or it's in a wrong format? I can listen to it with windows media player... it's a