similar to: Log file warnings from chan_sip in build_reply_digest

I've written an Ogg Vorbis (only !) player, which some may like to play with. It works for me (tm) and I like it. If you don't, well, you know what you can do with it ;) I leave it running 24/7 and it plays my music without problems and without annoying me. That's all it's for, really. You can get it from http://www.geoid.clara.net/rik/arch/squelch.tar.gz The README follows ...

Jean-Marc Valin wrote: > Andy Ross wrote: > > Uh, production applications almost always require squelch, no? > > Some do, some don't. In general, distinguishing between a keyboard > and a speech transient is next to impossible based only on a few ms > of speech. That is true for distinguishing it by waveform, but not by amplitude. As I mentioned, these transients are

Hi, I released Squelch 1.0beta9. It's a multi-platform Ogg Vorbis player, if you haven't heard of it. [1] Find it here: http://www.geoid.clara.net/rik/squelch.html Differences from beta8: * Vorbis comment editor ! [2] * More intelligent re-initialisation of output driver. * Stupid bugs in auto-update of master track list resolved. * Some bugs fixed, some more introduced ;) In theory,

> That is true for distinguishing it by waveform, but not by amplitude. > As I mentioned, these transients are objectively tiny. *Your* transients may be "tiny" and in any case, it doesn't help if you don't know the level you're recording at. I guess I'd be > curious as to which voice codec applications require no squelch (other > than trivial examples

I've made squelch-1.0beta5 and uploaded it - you can find it linked from http://www.geoid.clara.net/rik/squelch.html This version has a config dialog. It works out which output drivers libao has available and gives you a choice. You can also tell it where your 'audio dir' is - though that works a bit strangely at the moment - when you change it, all the files you have in your playlist

Jean-Marc Valin wrote: > Andy Ross wrote: > > Not knowing how VAD works, I can't say for sure. > > There are many ways to implement a VAD. I meant "not knowing how speex's VAD works", of course, not VAD in general. If you would stop interpreting everything I say in the least charitable manner, this might be going more smoothly than it is. (Tom was right, by the

Jean-Marc Valin wrote: > The noise suppressor will only attempt to remove stationary noise, > such as thermal noise, fans, ... The AGC can indeed do strange > things in these cases, but it's been improved in svn (compared to > 1.2beta1). OK, then the problem is that I misunderstood the feature. I assumed that dynamic squelch was part of it, but it's really something more

> OK, then the problem is that I misunderstood the feature. I assumed > that dynamic squelch was part of it, but it's really something more > along the lines of active noise cancellation. That's fine, I'll work > on improving my own squelch code. No. Active noise cancellation is yet another thing, where you cancel the noise in the "acoustic world" by

We have an Asterisk v1.2.16 box registering with an ITSP using SIP. The registration succeeds, and is confirmed with SIP SHOW REGISTER. However, we frequently (every few minutes) see this on our console: REGISTER attempt 1 to 999@pbx.itsp.com REGISTER attempt 2 to 999@pbx.itsp.com Any ideas what is going on? In particular 1. What causes the two register attempt messages above? 2. Why

Hi All, When I want use Asterisk as a PBX to cooperate SIP ITSP, I can not set the caller ID, so SIP ITSP do not accept the call. In Asterisk, I set a account in sip.conf to register on ITSP SIP Server: register => 6292@218.1.121.237/6292 And I added a user 6292 in Asterisk just like the account on ITSP SIP Server: [6291] type=friend username=6291 callerid=6291 host=dynamic

Asterisk 1.2.7 Redhat 9 I have DiDs from two different ITSP both set up as IAX2. Each one works when it's the only one in my iax.conf, but when I have them both defined in iax.conf at the same time, only one will work. My iax.conf is provided below. Any ideas how to fix? I'd like to use both DiDs! Thanks, H My iax.conf is below. When I dial the DiD provided by ITSP_B, the other

Hello list, my CLI is spammed with : [Sep 13 08:31:38] > doing dnsmgr_lookup for 'ssw6.itsp.tld' [Sep 13 08:31:38] > doing dnsmgr_lookup for 'ssw6.itsp.tld' [Sep 13 08:31:47] > doing dnsmgr_lookup for 'ssw4.itsp.tld' [Sep 13 08:31:48] > doing dnsmgr_lookup for 'ssw4.itsp.tld' [Sep 13 08:31:49] > doing dnsmgr_lookup for 'ssw4.itsp.tld' [Sep

SetUp: - Asterisk behind a NAT, - Red Hat 9.0 - Asterisk 1.2.14 My Asterisk box is behind a NAT and I have a DiD from an ITSP. I have my dial plan set up so that when outside callers dial the DiD, the call is answered by my auto-attendant. The caller can then select who they'd like to speak to and the call is transferred to the external line associated with that person (usually a mobile

I also added a .desktop file for Gnome/KDE so that you can click on squelch from your panel etc. No icon right now - tmake doesn't support installing, so it's quite difficult to get that to work. Also I put a tiny script called 'squelch_wrapper' in the dist, which sets your LD_LIBRARY_PATH before running squelch - this way, you get the right Qt library loaded. Rik --- >8

Asterisk 1.2.7 RedHat 9.0 I frequently have the need to redirect calls that come in on a DiD provisioned by my ITSP, back to the ITSP so that they can terminate the call on the PSTN. For example when an external call comes in, I often have to send it to a cell phone. I believe that this is referred to as "hairpinning" the call. I do this by answering the incoming call and then I use

I'm using Asterisk (1.2.14, RedHat 9) but I've been having trouble with SIP re-invites. I have a DiD from an ITSP and when someone calls in, Asterisk plays a menu recording and transfers the call to the external line the caller selects. Since both sides of the call are external, I want to use re-invite to avoid the rtp packets from going through my server after the call is bridged. I

Great ! I'm gonna it try ASAP ! Is there another way (ie not using different ports) to get several trunks to a given ITSP ? Let me explain this a bit further. My setup is: ITSP <---- SIP----> Asterisk <----> Phones For various reasons, I want my Asterisk box to have several trunks/SIP account with my ITSP. First method, is to configure a specific port for each trunk: ITSP will

Hi, I still have the same problem trying to configure ITSP failover in extensions.conf for a connected PRI. Any comments thoughts or direction would be greatly appreciated. I sympathize with wanting inbound DID failover. If we have a client with multiple DIDs we will spread them across two or three ITSPs so that all inbound connectivity will not be lost if one of them has an issue. I

Asterisk 1.2.7 RedHat 9.0 I frequently have the need to redirect calls that come in on a DiD provisioned by my ITSP, back to the ITSP so that they can terminate the call on the PSTN. For example when an external call comes in, I often have to send it to a cell phone. I believe that this is referred to as "hairpinning" the call. I do this by answering the incoming call and then I use