similar to: tiny audio drops (blips)

Haven't seen email since the 29th.. just testing. -------------- next part -------------- A non-text attachment was scrubbed... Name: mhess.vcf Type: text/x-vcard Size: 288 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20050801/96555713/mhess.vcf

My cdrs are missing accountcodes for incoming calls from other asterisk servers.. I've seen a few people mentioning this on the list and the solution seems to be setting up a dialplan for incoming calls from a particular sip peer.. in my opinion this does not scale well at all and I am looking for a solution to correct this problem. example sip peer: [asterisk_gw] type=friend

Maybe I've missed it but I'm wondering if there has been any movement towards getting t.38 support into asterisk.. has there been any news? Where is t.38 support at? will it even happen? -------------- next part -------------- A non-text attachment was scrubbed... Name: mhess.vcf Type: text/x-vcard Size: 279 bytes Desc: not available Url :

I'm having a problem with voicemail where the system will allow me to login to the vm box no problem but when it starts tell tell me the number of messages I have it hangs up.. I get "you have" and it dies right there.. I'm running cvs tag v1-0.. what might be causing this? I looked through my mail list archive and didn't notice anything like this.. -------------- next

Hi, I've noticed on asterisk 1.8.18 I'm hearing the blip of '#' DTMF to end the recording on the recording itself. Is there an easy way to truncate the last 200ms of the recording or so to eliminate this? The DTMF is coming in through rfc2833 and not inband. Thanks. -- James -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have this in sip show history for a particular channel marked as dead (should be removed) in sip show channels: 1. TxReqRel INVITE / 102 INVITE 2. Rx SIP/2.0 / 102 INVITE 3. CancelDestroy 4. Rx SIP/2.0 / 102 INVITE 5. CancelDestroy 6. Unhold SIP/2.0 7. Rx SIP/2.0 / 102 INVITE 8. CancelDestroy 9. Unhold SIP/2.0 10. Rx

I just updated one of my stable asterisk systems to head to test it out.. and I'm receiving a interesting log message now in asterisk.. Aug 2 13:20:56 NOTICE[15382]: chan_sip.c:5617 check_auth: stale nonce received from '<sip:3034585725@voip.livewirenet.com;user=phone>' (one line per registration) I'm using an AudioCodes mp108.. it worked fine with the latest stable..

Hi, I tried to use rmnlIndepMetrop (bayesm package) for my MNL model with 4 choice alternatives, 5 independent variables, 69 observations, dim(X) [1] 276 5, nu=6. So I run such code: if(nchar(Sys.getenv("LONG_TEST")) != 0) {R=2000} else {R=10} set.seed(66) df=read.table("X_metrop.dat",header=TRUE) inp=as.matrix(df) y=as.numeric(inp[,1]) n=length(y) p=4

Hi all, When the Dahdi driver detects DTMF, it seems it's not muting the first 5-15 ms and sometimes the last 2-10 ms of the DTMF tone. This shows up in recorded voicemail greetings -- you hear a very short DTMF '#', or sometimes two blips, at the end of the recording. I have a Mitel SX-200 connected to Asterisk 1.6.0.1 by a couple of Digium cards: a TE420 w/Octasic and pri_net

Has anybody compiled app_conference as of late? I've already asked on the app_conference devel list but as I'm rather in a hurry my thinking is somebody here has both run into and found a way to get this compiled and running. Using stable asterisk and the most recent app_conference from it's cvs on sourceforge.. -------------- next part -------------- A non-text attachment was

I'm looking for opinions on g726-32 vs. g711u.. They both have decent audio quality.. and looking at the wiki I get the impression that g726 is like the little brother to g711. Yet, I've run into quite a few sip termination vendors who don't support it. Does anyone on the list actively use g726 for anything and what have those experiences been? The g726 codec for me at least

Strange issue.... when I record a file from a phone to the asterisk system I get a blip in the recording every 30 seconds. It's a very small blip, but it is there. It seems like it's only if I'm recording, not when I'm playing back that the issue happens. My SATA drives, ETH0, and my Sangoma card are all on seperate IRQs. Any thoughts on what might be causing this and how to

This may not be the right place to ask, but I have an annoying issue with a Sipura/SPA1000-2.0.10(e) ATA device connected to an Asterisk box. (The system is remote to me, so I've only been able to observe this by dialling into a VoIP phone on-site, then run commands on the box remotely!) First of all it's all working fine connected to an Asterisk box and the user can make/take calls

Background: I administer a cluster of RedHat EWS 3U4 Linux workstations at a university. I built R 2.1.0 from source: ./configure \ --prefix=/sscc/opt/R-2.1.0 \ --with-blas=no \ 2>&1 \ | tee NUInstall.configure R is now configured for i686-pc-linux-gnu Source directory: . Installation directory: /sscc/opt/R-2.1.0 C compiler:

I've just done some rudimentary testing to see how Vorbis degrades at absurdly low bitrates without downsampling. In summary, don't hope for anything decent below -q 0 for now. I tried oggenc -b <bitrate> -M <bitrate> for the below and a few in between: 24k - spectral energy "floor" captured decently, but many pure-tone blips (think old computer movie sound effects)

Hi, I have an IVR that sounds just fine and dandy over ZAP. However, when I dial in through an 800 number from a provider that I connect to via IAX I get this 'blip' in the sound file. At first I thought it was just packet loss, but it happens at the exact same spot every single time. There are several parts to the IVR menu and several places where this blip happens every single time

Hi! My winamp crashes when i play this file: http://rehlin.hemmet.chalmers.se/~papa/files/test.ogg the wav is in the same place. Does someone else have the same 'problem'? (ok, i know its a silly test sound, but i still think it should decode without crashing) ome numbers: betaversion of oggenc: 4 bitrate used: 350 version of winamp: 2.74 version of vorbis plugin: 1.04 windows: 2000 pro

I'm having an unusual problem at one of my branch offices. Every now and then they will make a call and the person they call is unable to hear them, but they are able to hear the person. The Asterisk server has only one ethernet interface and is on the same physical network as the 2 snom 300 phones and is connected to the PSTN lines with a Rhino R4FXO-EC card. Usually hanging up and

Good morning, On Sat, 2017-07-15 at 20:51 +0000, ScanCaster wrote: > On Fri, 14 Jul 2017 09:16:34 +0000, Philipp Schafft wrote: > > > This posting confirms me in that I should write a separate e-mail to the > > list on how metadata updates work and how they should be implemented on > > the source side. > > And what is different in the way this is updated??? The

First of all, *thanks* to all the developers for their hard work on Ogg Vorbis. It's greatly appreciated. I'm having a problem with encoding in Debian GNU/Linux 3.0. The oggs contain occasional dropouts and "blips" here and there. Example files: Original (~7 MB): http://personal.inet.fi/musiikki/nebularia/test/t2title.flac Ogg Vorbis file (~800 KB):