search for: 2.4kbps

when i issue asterisk -vvv command i get this error please help regards Barbra [app_softhangup.so] => (Hangs up the requested channel) == Registered application 'SoftHangup' [codec_lpc10.so] => (LPC10 2.4kbps (signed linear) Voice Coder) == Registered translator 'lpc10tolin' from format 7 to 6, cost 50 == Registered translator 'lintolpc10' from format 6 to 7,

Thank you. You're right ... my error ... I meant to say 12 bytes (including the 2 bytes for VAD). And it is 10ms/frame. No matter ... thank you for the SPEEX specs. In terms of quality, what SPEEX bit rate compares with G.729 at 8kbps data rate please? Is there some reason why you chose the 20ms frame rate? Do you keep that same frame rate for the different bit rates? The faster frame

I caught a thread the other day concerning Astricon and users embedding Asterisk on a Linksys or Netgear broadband router. I lost track of the email thread, if anyone is presently working with this scenario please shoot me an email. Thanks Cory Andrews ++++++++++++++++++ VoIPSupply.com PBXSelect.com ++++++++++++++++++ 454 Sonwil Drive Buffalo, NY 14225 voice direct - 716.250.3402 fax -

Thank you Jean-Marc. My understanding is that G.729 is a telephone codec, so there must have been some reason why its developers went to 10ms/frame. Do you know why that might be? From a recent post on this list I saw somebody talking about your decoded sample rate being 8KHZ/sec. and then he mentioned that being 160 bytes at 20ms/frame. That said, I take it that your decoded samples are

Hello list! I?m having an issue when reloading Asterisk, I?ve had this problem in Asterisk 1.6.1.6 so I upgrade to 1.6.2.1 version, but I still have the same error. For example, I send a "reload" in Asterisk CLI and this is the output: isb152*CLI> reload == Parsing '/etc/asterisk/extconfig.conf': == Found == Parsing '/etc/asterisk/manager.conf': == Found

B. Mitchell Loebel a ?crit : > Thank you. You're right ... my error ... I meant to say 12 bytes > (including the 2 bytes for VAD). And it is 10ms/frame. No matter ... > thank you for the SPEEX specs. In terms of quality, what SPEEX bit rate > compares with G.729 at 8kbps data rate please? Haven't done formal testing and it depends on whether it's G.729 or G.729A. I'd

Hi, I write to this list, because I didn't find anything on the net. I installed asterisk using bristuff-0.0.2 without any errors, but when I start asterisk with "asterisk -vvvc" I get following error: [codec_ilbc.so] => (iLBC/PCM16 (signed linear) Codec Translator) == Registered translator 'ilbctolin' from format ILBC to SLINR, cost 127 Segmentation fault Removing

2003-06-28 Bandwidth Study - John Todd (jtodd @loligo.com) Purpose: ------------- To obtain a better chart of actual bandwidth usage per codec as seen "on-the-wire" when using IAX2 trunking between two Asterisk telephony servers. Discussion: ------------- Past threads on the asterisk-dev and asterisk-users lists have indicated that the optimal way to save bandwidth on

Hi, I have licensed the digium G.729A codec. But for some reason incoming and outgoing calls will ALWAYS use G.711a. When I force my phone to only accept G.729 then an incoming call from ZAP goes straight to my voicemailbox as the phone doesn't accept the codec Asterisk wants, even if I force it in sip.conf. Is there anywhere else I need to look to tell ZAP to use G.729 preferrably ? The

Hello Jean-Marc: How many bits do you have in a frame please and what is your frame rate? For example, G.729 (ACELP) has 12 bits/frame including VAD and the frame rate is 100/second ... I'm looking for the comparable figures for SPEEX. Thank you. --- B. Mitchell Loebel, CEO, VP Engineering 408 425-9920 InstaFlash International Corporation (formerly

Hi, I am trying to replace my Cisco 5300 gateway with my new Zap TE405P card. Incoming calls and outgoing calls between my cisco and my SIP phone works fine on G.729. Recording messages in the asterisk voice-mailbox also works fine from both my SIP phone as well as PSTN -> Cisco -> Asterisk. I have licensed the digium G.729A codec. When I connect my ISDN PRI to my Zap card and I call

Dear all, I'm trying to use ulaw and alaw with Diax and Asterisk but I'm not able to, I got the following error message: Mar 20 11:47:59 NOTICE[7099]: chan_iax2.c:6350 socket_read: Rejected connect attempt from 192.168.0.55, requested/capability 0x8/0xc incompatible with our capability 0xfe02. I do not understand why because my Asterisk box load these codecs properly! Does somebody

Hi, I've been having a little problem with my asterisk servers, I have 4 identical asterisk servers setup (same hardware, same OS, same config). Once in a while (once or twice a day) one of the server crashes on the cron job reload. But I realized this only happens on 3 of the 4 servers. Tried to spot the difference between that one server that wasn't crashing. The difference I found was

Well, it's an ARM7TDMI core, so basically one register operation per clock, with memory accesses taking longer. Having the memory on-chip should make memory access much less of an impact. I was afraid that you would answer the way you did: I thought about my question after I sent it, and the "LP" in CELP is what makes it a sequential process; it can't do linear prediction on a

I am interested in using Speex in an embedded system built around an ARM microcontroller. I have seen other posts indicating that Speex can run in real-time on some iPAQ PDA's, but these are using a StrongARM 166MHz processor. I'm looking more at the chips from Atmel (SAM7), Philips (LPC2xxx), and TI (TMS 470), which are ARM7TDMI with on-chip SRAM and flash, running at speeds of 33 to

Hi everyone, I am running asterisk on red hat linux 9 box. The sound card is Intel 82801db AC' 97 audio and the module is i810_audio. It runs well with other applications like xmms and the standard tests deliver a sound . I have also tried to record voice and that works well too. 1-)Now when i run asterisk and i dial out an extension to play any sound there is none. The same thing

Hello! Asterisk "CVS-HEAD-08/06/04-14:55:13" won't start on two of three different Gentoo machines. This is the output of gdb: ultra asterisk # gdb /usr/sbin/asterisk GNU gdb 6.0 Copyright 2003 Free Software Foundation, Inc. GDB is free software, covered by the GNU General Public License, and you are welcome to change it and/or distribute copies of it under certain conditions.

Hi, Ive been struggling with asterisk for a few days now. I understand pretty much how it works and how to tie things together (SIP -> SIP internally works fine etc), but just cannot get asterisk to work with an X100P clone (its a Ambient MD3200, if that means anything to you guys). I have tried (initially) asterisk 1.07 with zaptel 1.07, and now Asterisk CVS-HEAD with zaptel cvs. Both give

Hi List, I am facing the reverse problem as stated here.I am using ATA 186 to make and recieve call to * through OH323 driver. When I use G711 codec in the ATA to make call then then as soon as i dial an extension the * crashes with 'segmentation fault'. But the same scenerio works fine when i use 723 codec in the ATA .I can dial the number and extension very well/(I have 723 support in

Hi John, Try adding username=5702 and username=5703 to each of the configs in sip.conf. I recall I had this problem with the Grandstreams. -----Original Message----- From: John Coll [mailto:john.coll@csoft.co.uk] Sent: Saturday, January 03, 2004 11:56 AM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Newbie - getting two local phones tocommunicate would be a good start :)