similar to: Which is best provider for G.729

[ Optimised G.729A 'Howlet' for Asterisk & FreSWITCH ] Howler Technologies are proud to announce today the launch of their fully indemnified and highly optimised G.729A solution for Asterisk, including a unique floating license model. This is the first in a series of products dubbed 'Howlets' that add highly performant transcoding and signal processing modules to open-source

I have problemin g729 codec compatibility,I get the g729 module from http://asterisk.hosting.lv/ and I have Asterisk 1.4.22-3 RPM What g729 module should I download ? I already downloaded codec_g723-ast14-icc-glibc-pentium4.so [trixbox1.localdomain asterisk]# cat /proc/cpuinfo processor : 0 vendor_id : GenuineIntel cpu family : 15 model : 4 model

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Hello I have a confusion relating to G.729 codec. I know how to install where to get license but i really don't know why we need it? Why people use G.729 codec with asterisk? look all functionality can be done with out it ie calling from sip to iax protocol and sip/ iax to E1, then why we need this?? regards Adam -------------- next part -------------- An HTML attachment was

Not sure if anybody noticed, but it seems like Skype For Asterisk is out. $66 per channels, pretty pricey http://store.digium.com/productview.php?product_code=1SFA0001 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090817/cd8c6546/attachment.htm

I have an interesting problem. I am attempting to install a payphone utilizing a Cisco ATA-188. The payphone actually works, but there are some timing issues. What happens is you pick up the payphone and the ATA grabs a line and goes offhook. While you monkey with putting money in and dialing the number, you are eating up the time before you get the offhook reorder tones (or howler tones

I know many of you have been waiting for this for a while, so I'll keep this short: The Skype for Asterisk Public Beta is now available on the Digium store. We are pleased to announce the open beta of Skype For Asterisk is ready to begin and we look forward to you participation. To obtain your copy of the software, please visit Digium?s web store and purchase (for zero dollars) the

I am wondering if it is possible for an extension that is served by a zaptel device to revert to dial tone once a call disconnects. For instance, if I make a call to another extension, talk with them, and THEY hang up, can I then be presented with a new dial tone rather than a congestion tone? Further, can an extension be set up so that, once the call goes back to dial tone, if the user does NOT

Hello Every one! I am little bit new to asterisk. I am doing research on different telecom options as well. I have question for you professionals In order to get E1 line working with Asterisk. What E1 line parameters need to be specified in Asterisk(configuration files). They vary from country to country. What is difference in one countries E1 and others E1. the most important If we want to

I need a refresher course on how many licenses I need to buy. I have an Asterisk server that receives calls by SIP (G729) and then sends them to the PSTN via 32 Zap interfaces on an Astribank. I cannot remember if the license is per channel or per call so I do not know if I need 32 or 64 licenses for this application. Could anyone please remind me? -- Telecomunicaciones Abiertas de M?xico

hello , all i want free g729 codec for asterisk i tried so many moduless on asterisk.hosting.lv but cant find any related codec to my machine i cant understand where to start my asterisk version is 1.6.0.5 and following are output of cpu cat /proc/cpuinfo processor : 0 vendor_id : GenuineIntel cpu family : 6 model : 23 model name : Intel(R) Core(TM)2 Duo CPU

I have a SPA-3000 that is failing to hanging up pretty often; almost every day now. The weird thing is that an almost identically configured (same FW, different HW rev) second unit right next to it isn't having the same problem. Swap the lines and the problem stays with the unit. I've been going round and round with Sipura and their latest message told me to use a procedure detailed in

Hi, Does anybody else have problems like this. I'm in the UK with a 1mb ADSL service from Eclipse. I have a Draytek Vigour 2600 ADSL router. My * box is configured with a public IP address which is presented on one of the switch ports on the rear of the router. When there is some SIP activity, incoming mainly, towards my * box, the router will lockup after a short period?! I've tried

I''m trying without any succes to create application in which I have one keyed table. The table structure is: CREATE TABLE employees ( pin INTEGER PRIMARY KEY, first_name VARCHAR(30) NOT NULL, last_name VARCHAR(30) NOT NULL ); Whenever I enter new record, I need the the filed pin is also filled by user. Whenever a user edit the table, there should be option to edit or not edit the

Parametric Technology Customer Support is unable to process your email due to the subject of the mail. Mail messages sent to cs_ptc@ptc.com must contain a valid Customer Support call number or the string "copen" as the subject of the mail. For existing calls, specify the call number as the subject of the mail. Call numbers must be in the form C###### (where # is a valid digit [0-9]).

Parametric Technology Customer Support is unable to process your email due to the subject of the mail. Mail messages sent to cs_ptc@ptc.com must contain a valid Customer Support call number or the string "copen" as the subject of the mail. For existing calls, specify the call number as the subject of the mail. Call numbers must be in the form C###### (where # is a valid digit [0-9]).

Parametric Technology Customer Support is unable to process your email due to the subject of the mail. Mail messages sent to cs_ptc@ptc.com must contain a valid Customer Support call number or the string "copen" as the subject of the mail. For existing calls, specify the call number as the subject of the mail. Call numbers must be in the form C###### (where # is a valid digit [0-9]).

Hi, We had a last-minute cancellation from Vivox for today's conference. It happens that someone suggested a guest idea, Howler Technologies CTO Jay Fenton, who agreed to join the call from the road. Anything you want to know about transcoding to and from g729 is out topic for the first hour. My pal David Duffet knows this technology well and has kindly signed in to help guide us through this

On Fri, 2019-02-15 at 15:57 +1100, Darren Tucker wrote: > That was the original intent (and it's mentioned in RFC4419) however > each moduli file we ship (70-80 instances of 6 sizes) takes about 1 > cpu-month to generate on a lowish-power x86-64 machine. Most of it > is > parallelizable, but even then it'd likely take a few hours to > generate > one of each size. I

Hi all, I have done a major upgrade, first I have bought a Digium TDM400P (TDM02B) 2-Port FXO to replace x100p cards, than moved from an old version Asterisk to a new centos 4 box, compiled asterisk from Asterisk CVS-HEAD built on 2005-06-11. Seems that my iaxy is now "incompatible" with the TDM400P. When I receive a call from the TDM02B, the caller can't hear my voice (I am using