similar to: Standard Sound Files Distortion

I'd like to increase the speed of the Asterisk sound files. Miss Alison talks a bit slow. I can use sox to increase the speed, but then the pitch changes and she starts to sound like a chipmunk. Any audio experts out there know how I can increase the speed a little bit, and change the pitch accordingly so that she sounds ... normal? Thanks Doug.

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

I have the following in extensions.conf: exten => 1000,1,Meetme(|dMic|) According to the 'show application meetme' docs: 'i' - announce user join/leave (new in Asterisk 1.2) Well, when users join the conference, Asterisk records their name, but does not broadcast it into the conference. I have Asterisk version 1.2.4. I know this has worked in the past. This sure as heck

Hi everyone, How can I introduce 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

I'd like to use native moh instead of with mpg123... for some reason the processes never bloody die. For native moh to not spawn an external player, I'd need to convert the default supplied moh sound files in /var/lib/asterisk/mohmp3 to ulaw and g729 format. Anyone know of a free, easy way to convert them? Thanks, Doug.

Anyone know if the MySQL() application has a configurable timeout? If it tries to connect to a bogus IP, it's timeout seems to be a few minutes. I'd like to cut it down to a few seconds. Doug. __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com -------------- next part --------------

Can someone tell me why this is not valid... [start] exten => 1000,1,Answer exten => 1000,2,Wait,1 exten => 1000,3,AgentcallbackLogin(1000||2000@Local) exten => 2000,1,Macro(DialProxy,115551212) exten => 3000,1,Queue(testq||||45) while this is: [start] exten => 1000,1,Answer exten => 1000,2,Wait,1 exten => 1000,3,AgentcallbackLogin(1000||2000@start) exten =>

Crikey. I can't get this to work! Allegedly, you can press 0 while in the voicemail greeting and be dropped to the 'o' extension. For some reason, I can't get it to work. The 'docs' aren't clear about what context the o extension should be in. The voip wiki says "the context for the voicemail box that we're looking for in the dialplan for the jump to the

When I call meetme: exten => 1000,1,Answer exten => 1000,n,Meetme(|||d) Asterisk is complaing with: -- Executing Answer("IAX2/xxx.yyy.142.204:4569-2", "") in new stack -- Executing MeetMe("IAX2/xxx.yyy.142.204:4569-2", "|||d") in new stack -- Playing 'conf-getconfno' (language 'en') Warning, flexible rate not

Is anyone getting '500 Internal Server' errors back from their Polycom phones when Asterisk sends a SIP NOTIFY message to them? I called Polycom tech support, who where utterly useless. Of course Polycom won't officially support it anyway, as they only support Asterisk Business Edition. We're using 1.2.9, but it's been ocurring for quite some time. We have about 35 phones and

Paste your extensions.conf section that is relevant. -Matthew ----- Original Message ----- From: "Greg - Cirelle Enterprises" <gcirino@cirelle.com> To: <asterisk-dev@lists.digium.com> Sent: Monday, December 27, 2004 4:32 PM Subject: [Asterisk-Dev] realtime voicemail > Let me clarify my last message. > > If I put in the wrong password I get polled > again for

I''ve got this: file { ''/opt/sugarsync/tomcat/tomcat-home/current'': ensure => "tomcat-$config[''tomcat_version_server'']"; where $config[''tomcat_version_server''] was set with extlookup (the yaml one), by loading: --- tomcat_config: tomcat_version_server: 6.0.20-1 tomcat_version_libs: 1.0-1

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, 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

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