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...some distortion, echo, chopping sound and all other bad quality things that can happen to a SIP trunk? I have plenty of bandwidth and crisp clear lines so the only thing that I can think of is to limit bandwidth but even that requires quite some scripting work. Is there any easy way to simulate a distorted SIP line temporarily for testing? I am appreciate experienced inputs. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110428/0d2edf56/attachment.htm>

...and since Sept 27th), connected via iax2 with low-utilized ds3 internet, C7960 calls exten on remote system (also C7960), and call goes to VM. No other calls in either system (eg, no load). Both boxes have iax config'ed as: trunk=yes allow=ilbc jitterbuffer=yes Recorded VM messages are very distorted. Changing only jitterbuffer=no (and * restart), recorded VM messages are very clean. With jitterbuffer=yes and trunk=no, messages are very clean. Both boxes config'ed as: trunk=yes allow=gsm jitterbuffer=yes Recorded VM messages are very clean. Conclusion: looks like the combination of tr...

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 goo...

...n my mobile VoIP application to cancel speaker echo. The used version is 1.2rc1 from the website, and I'm compiling with fixed-point. On most occasions, the AEC works very well and cancels most of the echo (combined with the preprocessor). On some devices, where the microphone signal gets a bit distorted (because of sensitivity or some internal AGC pass), the AEC outputs distorted and unacceptable voice, while only augmenting the echo signal instead of cancelling it. I have tried a few things: 1. Compile in floating point - solves it, but irrelevant to some devices which don't have an FPU and...

...rver, 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 the caller, call is intermittent. Before the CVS upgrade, the call quality was fine. We did the CVS upgrade cos, earlier version had probs, when restarting PRI channels. pls help us to solve this prob, this is very mission critical. Herc -------------- next part -------------- An HTM...

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

...hould have a BER less than 1 billionth percent. It explained why the faxes got mangled. The fact that the transmission of faxes went well, turned out to be because those calls got a DIGITAL TC (0x08) from the conventional PBX. Those digital ISDN call, as intended for G4 faxes (e.g.) weren't distorted in any way. Puzzled by the fact that I couldn't find anything on the wiki, mailing lists or mantis about the inability to properly send faxes through an octoBRI, I tried to determine the cause myself. First I installed the latest stable version of bristuff (0.2.0-RC8n). To my surprise this...

Hi, I'm experiencing a few (but meaningful) cases of audio distortion (or bad quality). I can't say yet how often this happens. Please listen to the following sound file: http://213.96.91.201/temp/distorted_audio_1.wav This was recorded by Asterisk while the local SIP caller was dialing out a SIP trunk (so the problem is on my side, definitely, and it doesn't seem to be related to the bandwidth between my peer and the SIP provider). You can hear a scratchy sound during the whole fragment. I can&...

...r >> echo. The used version is 1.2rc1 from the website, and I'm compiling >> with fixed-point. >> On most occasions, the AEC works very well and cancels most of the echo >> (combined with the preprocessor). >> On some devices, where the microphone signal gets a bit distorted >> (because of sensitivity or some internal AGC pass), the AEC outputs >> distorted and unacceptable voice, while only augmenting the echo signal >> instead of cancelling it. >> I have tried a few things: >> >> 1. Compile in floating point - solves it, but irrel...

...es not exactly follow validation (I have no control of this). The stream switches from a prerecorded segment, to a live segment, back to a prerecorded segment. I believe the live segment is missing a B.O.S. packet, which is creating some unintended effects. The audio ends up playing, but it is very distorted until it switches back to a prerecorded segment. I've uploaded a test of the chain here: https://s3.amazonaws.com/teamalphaservices-test/b1-Live-b2.ogg It plays perfectly in Google chrome, but when attempting to use my own decoder, it has strange distortion when playing the live part. Here...

Does anyone have any idea what would cause distorted audio but ONLY for DTMF tones coming in over our analog lines. (The analog interfaces are X100P's). I have carefully adjusted the gains in the zapata.conf using a local test line after trying various settings with no gain or just random gain settings. RelaxDTMF has no effect. I set up a...

...working and I have been able to get * to accept a command for MusicOnHold and for Meetme after loading the ztdummy module. I used the default config for /etc/zaptel.conf since I saw no guidance on this. My problem now is that when I activate MusicOnHold, the sample music file sounds very slow and distorted. My best guess is that it is playing so slow it sounds like Darth Vader breathing. Anybody have any ideas on what is going on?

https://bugs.freedesktop.org/show_bug.cgi?id=97614 Bug ID: 97614 Summary: occasional screen distortions with nouveau under 4.8.0-rc4+ Product: xorg Version: unspecified Hardware: Other OS: All Status: NEW Severity: normal Priority: medium Component: Driver/nouveau

...I checked with nvapeek: > 0000154c: 0000009c > 00001540: b1010001 great - finally somebody who confirms this issue. > uname -a delivers > Linux mobuntu 3.11.0-15-generic #23-Ubuntu SMP Mon Dec 9 18:17:04 UTC 2013 > x86_64 x86_64 x86_64 GNU/Linux > I do not have any issues with distorted graphics during normal usage but my > problem is that resume from suspend mode makes X hang. > I also have these errors in dmesg > > [ 18.985158] nouveau E[ PBUS][0000:01:00.0] MMIO write of 0x00000000 > FAULT at 0x00fd94 > [ 18.986213] nouveau E[ PBUS][0000:01:00.0]...

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

Hi, I have tried using the VorbisEncoder class available with the vorbis-java port of the ogg-vorbis and shout and the resulting ogg file from the encoder is jumbed up and distorted. I have tried compiling the sources and changing the quality in the source with no success. Has anyone managed to use the encoder successfully. if so your response will be of great. -- Solomon Kariri, Software Developer, Cell: +254736 729 450 Skype: solomonkariri -------------- next part -------...

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

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. Tho...

I'm not sure where to check next, so I'm reaching out to those that know this stuff better than I. I've got Asterisk up and running, but I've still got an occasional audio issue. Once in a while (maybe 1 out of every 20-30 calls), the audio becomes heavily distorted, but only on the local side. The party on the other end says the audio is fine. We can hear them, although on these calls the remote party is quieter than usual. They can hear us just fine. But when we speak, our side becomes heavily distorted. It only happens when we speak. As long as we a...

https://bugs.freedesktop.org/show_bug.cgi?id=104897 Bug ID: 104897 Summary: nouveau memory allocation issue causes screen distortions under KDE Product: xorg Version: unspecified Hardware: x86-64 (AMD64) OS: Linux (All) Status: NEW Severity: major Priority: medium