similar to: Help with IAX , voice Distortion or Breakage

Hello all, We have the following problem: When calling via iax, the sound is off after a while - most often after about 5 minutes (sometimes later or earlier) - at one end or at both ends. While the channel is up, and packages are still being transmitted, you just can't hear anything. Sometimes you can hear something just a little, but with the voice greatly distorted, sounding like a

On Samstag, 14. Januar 2006 1:47 tim panton wrote: > That is weird, you would expect IAX to do better than SIP (bandwidth > wise) My point exactly. > 1) are you sure IAX trunking is actually happening ? It shows (T) in iax2 show so I am pretty sure. Timestamps are enabled as well. > 2) what codecs are you using. Are the codecs the same for IAX as > for sip? G.711 alaw and

Hi, this is the scenario: One * is placed in a central location with more than enough up/down bandwidth. One * is placed behind a DSL 3000/384. Both * are linked via IAX trunking. Everything is fine until the upload channel of the remote site is filled with a download, then heavy voice distortion starts. Well of course this is expected. So I fooled around with HFSC QoS scheduling on the remote

hi All, Our configuration is, ISDN PRI lines connected to Asterisk Server, SIP users connected to Asterisk We route calls from ISDN PRI to SIP users, We did a CVS upgrade few days ago, now the sip users(CISCO phones) are experiencing, distortion of voice while they are engaged in a call. ie, SIP users can here the call clearly, but the outside caller heres distorted voice. ie, to

I'm new to * and I have my * box connected to another * box located at my ISP via iax2. Using ilbc everyting works fine but with any other codec I've tried (gsm, ulaw, alaw) there is severe distortion on the sound going out of my box as well as a second or two of delay. The incoming sound quality is fine. This happens even if the outgoing sound is coming from the voice mail prompts on

Hi All, At present calls over IAX2 (ilbc) are good but they suffer from occasional distortion. The strange thing is that the distortion can only be heard by the calling party and not the called party in 95% of cases. IAX2 is being used with trunking enabled, using the ztdummy module as a timing source. Bandwidth shouldn't be an issue as there is more than sufficient plus we use QoS

Hello, I have been facing an issue that voice is getting distorted sometimes in MOH (MusicOnHold) application. I have checked and confirmed that lame is properly installed, even tried native formats (ulaw, alaw, gsm), but the randomly seen distortion in MOH can't be eliminated. I came to know about requirement of timing device for MOH and MeetMe and a very good illustration by Andrew

Hi all, I routinesly rips my CDs to WAV and then convert to ogg vorbis format for use in my car and portable player. I don't usually notice anything amiss, but on the last track of Mike Oldfield's "Music of the Spheres" album ("Musica Universalis", at the very end crescendo), the converted .ogg file exhibits terrible distortion (sounds like digital clipping). This

Ok maybe it's just my imagination, but it looks like the one-line "duh" fix I committed from Andrej Vakrcka <ander@cauldron.sk> this week DRAMATICALLY reduces distortion in encoded files. Take a look and compair. --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this list, send a message to

I've been noticing lately what seems to be some distortian in the standard asterisk sound files, used for voicemail. These files are stored on the local Asterisk system. When Asterisk plays them, I can hear some cracles and pops. I'd never noticed these until recently. I did a little test. This sounds fine... exten => 1000,1,Answer exten => 1000,n,Wait,1 exten =>

Hi All, I've run into a small hiccup in encoding my audios with Speex. When I encode audience laughter and applause with 'speexenc' (version 1.0.1), the result is quite acceptable... until I enable VBR. Then it distorts horribly. My understanding of VBR is that it frees the encoder to vary the number of bits emitted to better maintain the quality requested, and so I would have

Hi listers! Have a problem with distortion in some analog lines. When some call comes in from PSTN the sound is really distorte, nothing can be understanded, but Internal calls work ok. Funny thing is that when I start/stop asterisk,dahdi, and wanrouter services eveything goes fine again. This is happening every week or so. I'm using asterisk 1.4.36, dahdi linux 2.2.0.2 and wanpipe 3.4.9

For whatever reason, the music on hold is extremely distorted and loud. It didn't used to be this way and I haven't changed anything, yet it persists. This is on all the channels we use (SIP, IAX2, Zap, ALSA). Can anyone help with this, or has anyone seen this? The mp3s play fine on any computer and haven't changed since they did work. Those wishing to hear for themselves, feel

Whilst in a call using the mec3 echo canceller today I had period of about 20 seconds of speech distortion. It's hard to describe but to me the call sounded as though we were having the conversation in a bathroom with some extra noise bursts and echo thrown in. I could carry on the call, with difficulty, and my correspondent didn't complain of any noise at all. After that 20 seconds

Hi all, I'm having a playback issue I can't eliminate. I'm encoding 16khz mono WAV files (voiceover recordings) to speex using a quality setting of 4. In any standalone speex player e.g. winamp or using speex frontend etc. the files play fine, but in a jspeex environment there's a subtle high pitched beeping type distortion, mostly on the transients. The sound is similar to

Hi, I set "alaw = 1-7" in /etc/zaptel.conf hoping to make my zap channels the same as the PSTN. This caused the levels to be about 20dB too high, as well as being distorted. I adjusted the txgain and rxgain settings in /etc/asterisk/zapata.conf to sane levels, but now I still have rather distorted zap extension -> zap extension sound quality. What am I missing here ? zap 1-4 are

On 12/06/18 13:50, Nadav Amit wrote: > GCC considers the number of statements in inlined assembly blocks, > according to new-lines and semicolons, as an indication to the cost of > the block in time and space. This data is distorted by the kernel code, > which puts information in alternative sections. As a result, the > compiler may perform incorrect inlining and branch

On Sun, Jun 10, 2018 at 07:19:08AM -0700, Nadav Amit wrote: > +/* > + * This generates an indirect call based on the operation type number. > + * The type number, computed in PARAVIRT_PATCH, is derived from the > + * offset into the paravirt_patch_template structure, and can therefore be > + * freely converted back into a structure offset. > + */ > +.macro PARAVIRT_ALT

On Tue, Jun 19, 2018 at 12:48:51PM -0700, Nadav Amit wrote: > +#define paravirt_alt \ > + "PARAVIRT_CALL type=\"%c[paravirt_typenum]\"" \ > + " clobber=\"%c[paravirt_clobber]\"" \ > + " pv_opptr=\"%c[paravirt_opptr]\"" That wants to be: + " pv_opptr=\"%c[paravirt_opptr]\";" And other than

Hallo all, i ready send one emeil with screenshots but it seems to take some time till it will be accepted, so i send one more time with links. > Hallo all, > there is sintel online premiere up today http://www.sintel.org/ . But > oggtheora video is every thing but not gut. I downloaded 720p version > but IMHO some times it looks for me like scaled 480p. In my > experience theora